diff options
Diffstat (limited to 'tinyDAV/src/audio')
25 files changed, 7336 insertions, 7368 deletions
diff --git a/tinyDAV/src/audio/alsa/tdav_common_alsa.c b/tinyDAV/src/audio/alsa/tdav_common_alsa.c index d1deec8..becc310 100755 --- a/tinyDAV/src/audio/alsa/tdav_common_alsa.c +++ b/tinyDAV/src/audio/alsa/tdav_common_alsa.c @@ -1,17 +1,17 @@ /* Copyright (C) 2014 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -28,247 +28,247 @@ int tdav_common_alsa_init(tdav_common_alsa_t* p_self) { - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if (p_self->b_initialized) { - ALSA_DEBUG_WARN("Already initialized"); - return 0; - } - tsk_safeobj_init(p_self); - p_self->b_initialized = tsk_true; - return 0; + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if (p_self->b_initialized) { + ALSA_DEBUG_WARN("Already initialized"); + return 0; + } + tsk_safeobj_init(p_self); + p_self->b_initialized = tsk_true; + return 0; } int tdav_common_alsa_lock(tdav_common_alsa_t* p_self) { - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - return tsk_safeobj_lock(p_self); + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + return tsk_safeobj_lock(p_self); } int tdav_common_alsa_unlock(tdav_common_alsa_t* p_self) { - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - return tsk_safeobj_unlock(p_self); + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + return tsk_safeobj_unlock(p_self); } int tdav_common_alsa_prepare(tdav_common_alsa_t* p_self, tsk_bool_t is_capture, int ptime, int channels, int sample_rate) { - int err = 0, val; - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(p_self); - - if (p_self->b_prepared) { - ALSA_DEBUG_WARN("Already prepared"); - goto bail; - } - if (!p_self->p_device_name) { - p_self->p_device_name = strdup("default"); - } - p_self->b_capture = is_capture; - - if ((err = snd_pcm_open(&p_self->p_handle, p_self->p_device_name, is_capture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, /*SND_PCM_NONBLOCK | SND_PCM_ASYNC*/0)) != 0) { - ALSA_DEBUG_ERROR("Failed to open audio device %s (%s)", p_self->p_device_name, snd_strerror(err)); - goto bail; - } - ALSA_DEBUG_INFO("device('%s') opened", p_self->p_device_name); - - if ((err = snd_pcm_hw_params_malloc(&p_self->p_params)) != 0) { - ALSA_DEBUG_ERROR("Failed to allocate hardware parameter structure(%s)", snd_strerror(err)); - goto bail; - } - - if ((err = snd_pcm_hw_params_any(p_self->p_handle, p_self->p_params)) < 0) { - ALSA_DEBUG_ERROR("Failed to initialize hardware parameter structure (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - if ((err = snd_pcm_hw_params_set_access(p_self->p_handle, p_self->p_params, SND_PCM_ACCESS_RW_INTERLEAVED)) != 0) { - ALSA_DEBUG_ERROR("Failed to set access type (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - if ((err = snd_pcm_hw_params_set_format(p_self->p_handle, p_self->p_params, SND_PCM_FORMAT_S16_LE)) != 0) { - ALSA_DEBUG_ERROR("Failed to set sample format (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - val = sample_rate; - if ((err = snd_pcm_hw_params_set_rate_near(p_self->p_handle, p_self->p_params, &val, 0)) != 0) { - ALSA_DEBUG_ERROR("Failed to set sample rate (rate=%d, device=%s, err=%s)", p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - ALSA_DEBUG_INFO("sample_rate: req=%d, resp=%d", sample_rate, val); - p_self->sample_rate = val; - - val = channels; - if ((err = snd_pcm_hw_params_set_channels_near(p_self->p_handle, p_self->p_params, &val)) != 0) { - ALSA_DEBUG_ERROR("Failed to set channels (channels=%d, device=%s, err=%s)", p_self->channels, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - ALSA_DEBUG_INFO("channels: req=%d, resp=%d", channels, val); - p_self->channels = val; - - if (!is_capture) { - unsigned int periods = ALSA_PLAYBACK_PERIODS; - snd_pcm_uframes_t periodSize = (ptime * p_self->sample_rate * p_self->channels) / 1000; - if ((err = snd_pcm_hw_params_set_periods_near(p_self->p_handle, p_self->p_params, &periods, 0)) != 0) { - ALSA_DEBUG_ERROR ("Failed to set periods (val=%u, device=%s, err=%s)", periods, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - snd_pcm_uframes_t bufferSize = (periodSize * periods); - if ((err = snd_pcm_hw_params_set_buffer_size(p_self->p_handle, p_self->p_params, bufferSize)) != 0) { - ALSA_DEBUG_ERROR ("Failed to set buffer size (val=%lu, device=%s, err=%s)", bufferSize, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - ALSA_DEBUG_INFO("periods=%u, buffersize=%lu", periods, bufferSize); - } - - if ((err = snd_pcm_hw_params (p_self->p_handle, p_self->p_params)) != 0) { - ALSA_DEBUG_ERROR ("Failed to set parameters (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - if ((err = snd_pcm_prepare(p_self->p_handle)) != 0) { - ALSA_DEBUG_ERROR ("Failed to prepare device (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - /*if (is_capture)*/ { - p_self->n_buff_size_in_bytes = (ptime * p_self->sample_rate * (2/*SND_PCM_FORMAT_S16_LE*/ * p_self->channels)) / 1000; - if (!(p_self->p_buff_ptr = tsk_realloc(p_self->p_buff_ptr, p_self->n_buff_size_in_bytes))) { - ALSA_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_self->n_buff_size_in_bytes); - err = -4; - goto bail; - } - p_self->n_buff_size_in_samples = (p_self->n_buff_size_in_bytes >> 1/*SND_PCM_FORMAT_S16_LE*/); - ALSA_DEBUG_INFO("n_buff_size_in_bytes=%u", p_self->n_buff_size_in_bytes); - } - - ALSA_DEBUG_INFO("device('%s') prepared", p_self->p_device_name); - - // everything is OK - p_self->b_prepared = tsk_true; + int err = 0, val; + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(p_self); + + if (p_self->b_prepared) { + ALSA_DEBUG_WARN("Already prepared"); + goto bail; + } + if (!p_self->p_device_name) { + p_self->p_device_name = strdup("default"); + } + p_self->b_capture = is_capture; + + if ((err = snd_pcm_open(&p_self->p_handle, p_self->p_device_name, is_capture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, /*SND_PCM_NONBLOCK | SND_PCM_ASYNC*/0)) != 0) { + ALSA_DEBUG_ERROR("Failed to open audio device %s (%s)", p_self->p_device_name, snd_strerror(err)); + goto bail; + } + ALSA_DEBUG_INFO("device('%s') opened", p_self->p_device_name); + + if ((err = snd_pcm_hw_params_malloc(&p_self->p_params)) != 0) { + ALSA_DEBUG_ERROR("Failed to allocate hardware parameter structure(%s)", snd_strerror(err)); + goto bail; + } + + if ((err = snd_pcm_hw_params_any(p_self->p_handle, p_self->p_params)) < 0) { + ALSA_DEBUG_ERROR("Failed to initialize hardware parameter structure (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + if ((err = snd_pcm_hw_params_set_access(p_self->p_handle, p_self->p_params, SND_PCM_ACCESS_RW_INTERLEAVED)) != 0) { + ALSA_DEBUG_ERROR("Failed to set access type (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + if ((err = snd_pcm_hw_params_set_format(p_self->p_handle, p_self->p_params, SND_PCM_FORMAT_S16_LE)) != 0) { + ALSA_DEBUG_ERROR("Failed to set sample format (device=%s, err=%s)", p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + val = sample_rate; + if ((err = snd_pcm_hw_params_set_rate_near(p_self->p_handle, p_self->p_params, &val, 0)) != 0) { + ALSA_DEBUG_ERROR("Failed to set sample rate (rate=%d, device=%s, err=%s)", p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + ALSA_DEBUG_INFO("sample_rate: req=%d, resp=%d", sample_rate, val); + p_self->sample_rate = val; + + val = channels; + if ((err = snd_pcm_hw_params_set_channels_near(p_self->p_handle, p_self->p_params, &val)) != 0) { + ALSA_DEBUG_ERROR("Failed to set channels (channels=%d, device=%s, err=%s)", p_self->channels, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + ALSA_DEBUG_INFO("channels: req=%d, resp=%d", channels, val); + p_self->channels = val; + + if (!is_capture) { + unsigned int periods = ALSA_PLAYBACK_PERIODS; + snd_pcm_uframes_t periodSize = (ptime * p_self->sample_rate * p_self->channels) / 1000; + if ((err = snd_pcm_hw_params_set_periods_near(p_self->p_handle, p_self->p_params, &periods, 0)) != 0) { + ALSA_DEBUG_ERROR ("Failed to set periods (val=%u, device=%s, err=%s)", periods, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + snd_pcm_uframes_t bufferSize = (periodSize * periods); + if ((err = snd_pcm_hw_params_set_buffer_size(p_self->p_handle, p_self->p_params, bufferSize)) != 0) { + ALSA_DEBUG_ERROR ("Failed to set buffer size (val=%lu, device=%s, err=%s)", bufferSize, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + ALSA_DEBUG_INFO("periods=%u, buffersize=%lu", periods, bufferSize); + } + + if ((err = snd_pcm_hw_params (p_self->p_handle, p_self->p_params)) != 0) { + ALSA_DEBUG_ERROR ("Failed to set parameters (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + if ((err = snd_pcm_prepare(p_self->p_handle)) != 0) { + ALSA_DEBUG_ERROR ("Failed to prepare device (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + /*if (is_capture)*/ { + p_self->n_buff_size_in_bytes = (ptime * p_self->sample_rate * (2/*SND_PCM_FORMAT_S16_LE*/ * p_self->channels)) / 1000; + if (!(p_self->p_buff_ptr = tsk_realloc(p_self->p_buff_ptr, p_self->n_buff_size_in_bytes))) { + ALSA_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_self->n_buff_size_in_bytes); + err = -4; + goto bail; + } + p_self->n_buff_size_in_samples = (p_self->n_buff_size_in_bytes >> 1/*SND_PCM_FORMAT_S16_LE*/); + ALSA_DEBUG_INFO("n_buff_size_in_bytes=%u", p_self->n_buff_size_in_bytes); + } + + ALSA_DEBUG_INFO("device('%s') prepared", p_self->p_device_name); + + // everything is OK + p_self->b_prepared = tsk_true; bail: - if (err) { - tdav_common_alsa_unprepare(p_self); - } - tdav_common_alsa_unlock(p_self); - return err; + if (err) { + tdav_common_alsa_unprepare(p_self); + } + tdav_common_alsa_unlock(p_self); + return err; } int tdav_common_alsa_unprepare(tdav_common_alsa_t* p_self) { - int err = 0; - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(p_self); - - if (p_self->b_started) { - ALSA_DEBUG_ERROR("Must stop the capture device before unpreparing"); - err = -2; - goto bail; - } - - if (p_self->p_params) { - snd_pcm_hw_params_free(p_self->p_params); - p_self->p_params = tsk_null; - } - if (p_self->p_handle) { - snd_pcm_close(p_self->p_handle); - p_self->p_handle = tsk_null; - } - p_self->b_prepared = tsk_false; - - ALSA_DEBUG_INFO("device('%s') unprepared", p_self->p_device_name); + int err = 0; + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(p_self); + + if (p_self->b_started) { + ALSA_DEBUG_ERROR("Must stop the capture device before unpreparing"); + err = -2; + goto bail; + } + + if (p_self->p_params) { + snd_pcm_hw_params_free(p_self->p_params); + p_self->p_params = tsk_null; + } + if (p_self->p_handle) { + snd_pcm_close(p_self->p_handle); + p_self->p_handle = tsk_null; + } + p_self->b_prepared = tsk_false; + + ALSA_DEBUG_INFO("device('%s') unprepared", p_self->p_device_name); bail: - tdav_common_alsa_unlock(p_self); - return err; + tdav_common_alsa_unlock(p_self); + return err; } int tdav_common_alsa_start(tdav_common_alsa_t* p_self) { - int err = 0; - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(p_self); - - if (p_self->b_started) { - ALSA_DEBUG_WARN("Already started"); - err = - 3; - goto bail; - } - if (!p_self->b_prepared) { - ALSA_DEBUG_ERROR("Not prepared"); - err = -2; - goto bail; - } - - if ((err = snd_pcm_start(p_self->p_handle)) != 0) { - ALSA_DEBUG_ERROR ("Failed to start device (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); - goto bail; - } - - p_self->b_started = tsk_true; - ALSA_DEBUG_INFO("device('%s') started", p_self->p_device_name); + int err = 0; + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(p_self); + + if (p_self->b_started) { + ALSA_DEBUG_WARN("Already started"); + err = - 3; + goto bail; + } + if (!p_self->b_prepared) { + ALSA_DEBUG_ERROR("Not prepared"); + err = -2; + goto bail; + } + + if ((err = snd_pcm_start(p_self->p_handle)) != 0) { + ALSA_DEBUG_ERROR ("Failed to start device (channels=%d, rate=%d, device=%s, err=%s)", p_self->channels, p_self->sample_rate, p_self->p_device_name, snd_strerror(err)); + goto bail; + } + + p_self->b_started = tsk_true; + ALSA_DEBUG_INFO("device('%s') started", p_self->p_device_name); bail: - tdav_common_alsa_unlock(p_self); - return err; + tdav_common_alsa_unlock(p_self); + return err; } int tdav_common_alsa_stop(tdav_common_alsa_t* p_self) { - int err = 0; - if (!p_self) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(p_self); - - if (p_self->b_started) { - p_self->b_started = tsk_false; - //err = snd_pcm_drain(p_self->p_handle); - ALSA_DEBUG_INFO("device('%s') stopped", p_self->p_device_name); - } - if (p_self->b_prepared) { - tdav_common_alsa_unprepare(p_self); - } + int err = 0; + if (!p_self) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(p_self); + + if (p_self->b_started) { + p_self->b_started = tsk_false; + //err = snd_pcm_drain(p_self->p_handle); + ALSA_DEBUG_INFO("device('%s') stopped", p_self->p_device_name); + } + if (p_self->b_prepared) { + tdav_common_alsa_unprepare(p_self); + } bail: - tdav_common_alsa_unlock(p_self); - return err; + tdav_common_alsa_unlock(p_self); + return err; } int tdav_common_alsa_deinit(tdav_common_alsa_t* p_self) { - if (p_self && p_self->b_initialized) { - tdav_common_alsa_stop(p_self); - tdav_common_alsa_unprepare(p_self); - TSK_FREE(p_self->p_device_name); - TSK_FREE(p_self->p_buff_ptr); - tsk_safeobj_deinit(p_self); - p_self->b_initialized = tsk_false; - } - return 0; + if (p_self && p_self->b_initialized) { + tdav_common_alsa_stop(p_self); + tdav_common_alsa_unprepare(p_self); + TSK_FREE(p_self->p_device_name); + TSK_FREE(p_self->p_buff_ptr); + tsk_safeobj_deinit(p_self); + p_self->b_initialized = tsk_false; + } + return 0; } #endif /* HAVE_ALSA_ASOUNDLIB_H */ diff --git a/tinyDAV/src/audio/alsa/tdav_consumer_alsa.c b/tinyDAV/src/audio/alsa/tdav_consumer_alsa.c index 65bfcd8..273862d 100755 --- a/tinyDAV/src/audio/alsa/tdav_consumer_alsa.c +++ b/tinyDAV/src/audio/alsa/tdav_consumer_alsa.c @@ -1,17 +1,17 @@ /* Copyright (C) 2014 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -26,201 +26,200 @@ #define ALSA_DEBUG_ERROR(FMT, ...) TSK_DEBUG_ERROR("[ALSA Consumer] " FMT, ##__VA_ARGS__) #define ALSA_DEBUG_FATAL(FMT, ...) TSK_DEBUG_FATAL("[ALSA Consumer] " FMT, ##__VA_ARGS__) -typedef struct tdav_consumer_alsa_s -{ - TDAV_DECLARE_CONSUMER_AUDIO; +typedef struct tdav_consumer_alsa_s { + TDAV_DECLARE_CONSUMER_AUDIO; - tsk_bool_t b_muted; - tsk_bool_t b_started; - tsk_bool_t b_paused; + tsk_bool_t b_muted; + tsk_bool_t b_started; + tsk_bool_t b_paused; - tsk_thread_handle_t* tid[1]; - - struct tdav_common_alsa_s alsa_common; + tsk_thread_handle_t* tid[1]; + + struct tdav_common_alsa_s alsa_common; } tdav_consumer_alsa_t; static void* TSK_STDCALL _tdav_producer_alsa_playback_thread(void *param) { - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)param; - int err; - - ALSA_DEBUG_INFO("__playback_thread -- START"); - - tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); - - while (p_alsa->b_started) { - tdav_common_alsa_lock(&p_alsa->alsa_common); - //snd_pcm_wait(p_alsa->alsa_common.p_handle, 20); - //ALSA_DEBUG_INFO ("get (%d)", p_alsa->alsa_common.n_buff_size_in_bytes); - err = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(p_alsa), p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_bytes); // requires 16bits, thread-safe - //ALSA_DEBUG_INFO ("get returned %d", err); - if (err < p_alsa->alsa_common.n_buff_size_in_bytes) { - memset(((uint8_t*)p_alsa->alsa_common.p_buff_ptr) + err, 0, (p_alsa->alsa_common.n_buff_size_in_bytes - err)); - - } - if ((err = snd_pcm_writei(p_alsa->alsa_common.p_handle, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_samples)) != p_alsa->alsa_common.n_buff_size_in_samples) { - if (err == -EPIPE) { // pipe broken - err = snd_pcm_recover(p_alsa->alsa_common.p_handle, err, 0); - if (err == 0) { - ALSA_DEBUG_INFO ("recovered"); - goto next; - } - } - ALSA_DEBUG_ERROR ("Failed to read data from audio interface failed (%d->%s)", err, snd_strerror(err)); - tdav_common_alsa_unlock(&p_alsa->alsa_common); - goto bail; - } - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(p_alsa)); + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)param; + int err; + + ALSA_DEBUG_INFO("__playback_thread -- START"); + + tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); + + while (p_alsa->b_started) { + tdav_common_alsa_lock(&p_alsa->alsa_common); + //snd_pcm_wait(p_alsa->alsa_common.p_handle, 20); + //ALSA_DEBUG_INFO ("get (%d)", p_alsa->alsa_common.n_buff_size_in_bytes); + err = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(p_alsa), p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_bytes); // requires 16bits, thread-safe + //ALSA_DEBUG_INFO ("get returned %d", err); + if (err < p_alsa->alsa_common.n_buff_size_in_bytes) { + memset(((uint8_t*)p_alsa->alsa_common.p_buff_ptr) + err, 0, (p_alsa->alsa_common.n_buff_size_in_bytes - err)); + + } + if ((err = snd_pcm_writei(p_alsa->alsa_common.p_handle, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_samples)) != p_alsa->alsa_common.n_buff_size_in_samples) { + if (err == -EPIPE) { // pipe broken + err = snd_pcm_recover(p_alsa->alsa_common.p_handle, err, 0); + if (err == 0) { + ALSA_DEBUG_INFO ("recovered"); + goto next; + } + } + ALSA_DEBUG_ERROR ("Failed to read data from audio interface failed (%d->%s)", err, snd_strerror(err)); + tdav_common_alsa_unlock(&p_alsa->alsa_common); + goto bail; + } + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(p_alsa)); next: - tdav_common_alsa_unlock(&p_alsa->alsa_common); - } + tdav_common_alsa_unlock(&p_alsa->alsa_common); + } bail: - ALSA_DEBUG_INFO("__playback_thread -- STOP"); - return tsk_null; + ALSA_DEBUG_INFO("__playback_thread -- STOP"); + return tsk_null; } /* ============ Media Consumer Interface ================= */ static int tdav_consumer_alsa_set(tmedia_consumer_t* self, const tmedia_param_t* param) { - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; - int ret = 0; + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; + int ret = 0; - ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); + ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); - return ret; + return ret; } static int tdav_consumer_alsa_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; - int err = 0; - ALSA_DEBUG_INFO("******* tdav_consumer_alsa_prepare ******"); - - if (! p_alsa || !codec && codec->plugin) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(&p_alsa->alsa_common); - - // Set using requested - TMEDIA_CONSUMER(p_alsa)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(p_alsa)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(p_alsa)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - - // Prepare - err = tdav_common_alsa_prepare(&p_alsa->alsa_common, tsk_false/*is_record*/, TMEDIA_CONSUMER( p_alsa)->audio.ptime, TMEDIA_CONSUMER( p_alsa)->audio.in.channels, TMEDIA_CONSUMER( p_alsa)->audio.in.rate); - if (err) { - goto bail; - } - - ALSA_DEBUG_INFO("prepared: req_channels=%d; req_rate=%d, resp_channels=%d; resp_rate=%d", - TMEDIA_CONSUMER(p_alsa)->audio.in.channels, TMEDIA_CONSUMER(p_alsa)->audio.in.rate, - p_alsa->alsa_common.channels, p_alsa->alsa_common.sample_rate); - - // Set using supported (up to the resampler to convert to requested) - TMEDIA_CONSUMER(p_alsa)->audio.out.channels = p_alsa->alsa_common.channels; - TMEDIA_CONSUMER(p_alsa)->audio.out.rate = p_alsa->alsa_common.sample_rate; - + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; + int err = 0; + ALSA_DEBUG_INFO("******* tdav_consumer_alsa_prepare ******"); + + if (! p_alsa || !codec && codec->plugin) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(&p_alsa->alsa_common); + + // Set using requested + TMEDIA_CONSUMER(p_alsa)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(p_alsa)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(p_alsa)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); + + // Prepare + err = tdav_common_alsa_prepare(&p_alsa->alsa_common, tsk_false/*is_record*/, TMEDIA_CONSUMER( p_alsa)->audio.ptime, TMEDIA_CONSUMER( p_alsa)->audio.in.channels, TMEDIA_CONSUMER( p_alsa)->audio.in.rate); + if (err) { + goto bail; + } + + ALSA_DEBUG_INFO("prepared: req_channels=%d; req_rate=%d, resp_channels=%d; resp_rate=%d", + TMEDIA_CONSUMER(p_alsa)->audio.in.channels, TMEDIA_CONSUMER(p_alsa)->audio.in.rate, + p_alsa->alsa_common.channels, p_alsa->alsa_common.sample_rate); + + // Set using supported (up to the resampler to convert to requested) + TMEDIA_CONSUMER(p_alsa)->audio.out.channels = p_alsa->alsa_common.channels; + TMEDIA_CONSUMER(p_alsa)->audio.out.rate = p_alsa->alsa_common.sample_rate; + bail: - tdav_common_alsa_unlock(&p_alsa->alsa_common); - return err; + tdav_common_alsa_unlock(&p_alsa->alsa_common); + return err; } static int tdav_consumer_alsa_start(tmedia_consumer_t* self) { - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; - int err = 0; + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; + int err = 0; + + ALSA_DEBUG_INFO("******* tdav_consumer_alsa_start ******"); - ALSA_DEBUG_INFO("******* tdav_consumer_alsa_start ******"); + if (!p_alsa) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (!p_alsa) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(&p_alsa->alsa_common); + tdav_common_alsa_lock(&p_alsa->alsa_common); - if (p_alsa->b_started) { - ALSA_DEBUG_WARN("Already started"); - goto bail; - } + if (p_alsa->b_started) { + ALSA_DEBUG_WARN("Already started"); + goto bail; + } - /* start device */ - err = tdav_common_alsa_start(&p_alsa->alsa_common); - if (err) { - goto bail; - } + /* start device */ + err = tdav_common_alsa_start(&p_alsa->alsa_common); + if (err) { + goto bail; + } - /* start thread */ - p_alsa->b_started = tsk_true; - tsk_thread_create(&p_alsa->tid[0], _tdav_producer_alsa_playback_thread, p_alsa); + /* start thread */ + p_alsa->b_started = tsk_true; + tsk_thread_create(&p_alsa->tid[0], _tdav_producer_alsa_playback_thread, p_alsa); - ALSA_DEBUG_INFO("started"); + ALSA_DEBUG_INFO("started"); bail: - tdav_common_alsa_unlock(&p_alsa->alsa_common); - return err; + tdav_common_alsa_unlock(&p_alsa->alsa_common); + return err; } static int tdav_consumer_alsa_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - int err = 0; - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; - - if (!p_alsa || !buffer || !size) { - ALSA_DEBUG_ERROR("Invalid paramter"); - return -1; - } - - //tdav_common_alsa_lock(&p_alsa->alsa_common); - - if (!p_alsa->b_started) { - ALSA_DEBUG_WARN("Not started"); - err = -2; - goto bail; - } - - if ((err = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(p_alsa), buffer, size, proto_hdr))) {//thread-safe - ALSA_DEBUG_WARN("Failed to put audio data to the jitter buffer"); - goto bail; - } - + int err = 0; + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; + + if (!p_alsa || !buffer || !size) { + ALSA_DEBUG_ERROR("Invalid paramter"); + return -1; + } + + //tdav_common_alsa_lock(&p_alsa->alsa_common); + + if (!p_alsa->b_started) { + ALSA_DEBUG_WARN("Not started"); + err = -2; + goto bail; + } + + if ((err = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(p_alsa), buffer, size, proto_hdr))) {//thread-safe + ALSA_DEBUG_WARN("Failed to put audio data to the jitter buffer"); + goto bail; + } + bail: - //tdav_common_alsa_unlock(&p_alsa->alsa_common); - return err; + //tdav_common_alsa_unlock(&p_alsa->alsa_common); + return err; } static int tdav_consumer_alsa_pause(tmedia_consumer_t* self) { - return 0; + return 0; } static int tdav_consumer_alsa_stop(tmedia_consumer_t* self) { - tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; - int err; - - if (!p_alsa) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - /* should be done here */ - p_alsa->b_started = tsk_false; - - err = tdav_common_alsa_stop(&p_alsa->alsa_common); - - /* stop thread */ - if (p_alsa->tid[0]) { - tsk_thread_join(&(p_alsa->tid[0])); - } - - ALSA_DEBUG_INFO("stopped"); - - return 0; + tdav_consumer_alsa_t* p_alsa = (tdav_consumer_alsa_t*)self; + int err; + + if (!p_alsa) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + /* should be done here */ + p_alsa->b_started = tsk_false; + + err = tdav_common_alsa_stop(&p_alsa->alsa_common); + + /* stop thread */ + if (p_alsa->tid[0]) { + tsk_thread_join(&(p_alsa->tid[0])); + } + + ALSA_DEBUG_INFO("stopped"); + + return 0; } @@ -230,58 +229,56 @@ static int tdav_consumer_alsa_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_alsa_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_alsa_t *p_alsa = self; - if (p_alsa) { - ALSA_DEBUG_INFO("create"); - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(p_alsa)); - /* init self */ - tdav_common_alsa_init(&p_alsa->alsa_common); - } - return self; + tdav_consumer_alsa_t *p_alsa = self; + if (p_alsa) { + ALSA_DEBUG_INFO("create"); + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(p_alsa)); + /* init self */ + tdav_common_alsa_init(&p_alsa->alsa_common); + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_alsa_dtor(tsk_object_t * self) -{ - tdav_consumer_alsa_t *p_alsa = self; - if (p_alsa) { - /* stop */ - if (p_alsa->b_started) { - tdav_consumer_alsa_stop((tmedia_consumer_t*)p_alsa); - } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(p_alsa)); - /* deinit self */ - tdav_common_alsa_deinit(&p_alsa->alsa_common); - - ALSA_DEBUG_INFO("*** destroyed ***"); - } - - return self; +{ + tdav_consumer_alsa_t *p_alsa = self; + if (p_alsa) { + /* stop */ + if (p_alsa->b_started) { + tdav_consumer_alsa_stop((tmedia_consumer_t*)p_alsa); + } + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(p_alsa)); + /* deinit self */ + tdav_common_alsa_deinit(&p_alsa->alsa_common); + + ALSA_DEBUG_INFO("*** destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_alsa_def_s = -{ - sizeof(tdav_consumer_alsa_t), - tdav_consumer_alsa_ctor, - tdav_consumer_alsa_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_alsa_def_s = { + sizeof(tdav_consumer_alsa_t), + tdav_consumer_alsa_ctor, + tdav_consumer_alsa_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_alsa_plugin_def_s = -{ - &tdav_consumer_alsa_def_s, - - tmedia_audio, - "Linux ALSA consumer", - - tdav_consumer_alsa_set, - tdav_consumer_alsa_prepare, - tdav_consumer_alsa_start, - tdav_consumer_alsa_consume, - tdav_consumer_alsa_pause, - tdav_consumer_alsa_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_alsa_plugin_def_s = { + &tdav_consumer_alsa_def_s, + + tmedia_audio, + "Linux ALSA consumer", + + tdav_consumer_alsa_set, + tdav_consumer_alsa_prepare, + tdav_consumer_alsa_start, + tdav_consumer_alsa_consume, + tdav_consumer_alsa_pause, + tdav_consumer_alsa_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_alsa_plugin_def_t = &tdav_consumer_alsa_plugin_def_s; diff --git a/tinyDAV/src/audio/alsa/tdav_producer_alsa.c b/tinyDAV/src/audio/alsa/tdav_producer_alsa.c index d5c4021..fc8d4e2 100755 --- a/tinyDAV/src/audio/alsa/tdav_producer_alsa.c +++ b/tinyDAV/src/audio/alsa/tdav_producer_alsa.c @@ -1,17 +1,17 @@ /* Copyright (C) 2014 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -26,175 +26,174 @@ #define ALSA_DEBUG_ERROR(FMT, ...) TSK_DEBUG_ERROR("[ALSA Producer] " FMT, ##__VA_ARGS__) #define ALSA_DEBUG_FATAL(FMT, ...) TSK_DEBUG_FATAL("[ALSA Producer] " FMT, ##__VA_ARGS__) -typedef struct tdav_producer_alsa_s -{ - TDAV_DECLARE_PRODUCER_AUDIO; - - tsk_bool_t b_muted; - tsk_bool_t b_started; - tsk_bool_t b_paused; - - tsk_thread_handle_t* tid[1]; - - struct tdav_common_alsa_s alsa_common; +typedef struct tdav_producer_alsa_s { + TDAV_DECLARE_PRODUCER_AUDIO; + + tsk_bool_t b_muted; + tsk_bool_t b_started; + tsk_bool_t b_paused; + + tsk_thread_handle_t* tid[1]; + + struct tdav_common_alsa_s alsa_common; } tdav_producer_alsa_t; static void* TSK_STDCALL _tdav_producer_alsa_record_thread(void *param) { - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)param; - int err; - - ALSA_DEBUG_INFO("__record_thread -- START"); - - tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); - - while (p_alsa->b_started) { - tdav_common_alsa_lock(&p_alsa->alsa_common); - if ((err = snd_pcm_readi(p_alsa->alsa_common.p_handle, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_samples)) != p_alsa->alsa_common.n_buff_size_in_samples) { - ALSA_DEBUG_ERROR ("Failed to read data from audio interface failed (%d->%s)", err, snd_strerror(err)); - tdav_common_alsa_unlock(&p_alsa->alsa_common); - goto bail; - } - if (!p_alsa->b_muted && TMEDIA_PRODUCER(p_alsa)->enc_cb.callback) { - TMEDIA_PRODUCER(p_alsa)->enc_cb.callback(TMEDIA_PRODUCER(p_alsa)->enc_cb.callback_data, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_bytes); - } - tdav_common_alsa_unlock(&p_alsa->alsa_common); - } + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)param; + int err; + + ALSA_DEBUG_INFO("__record_thread -- START"); + + tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); + + while (p_alsa->b_started) { + tdav_common_alsa_lock(&p_alsa->alsa_common); + if ((err = snd_pcm_readi(p_alsa->alsa_common.p_handle, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_samples)) != p_alsa->alsa_common.n_buff_size_in_samples) { + ALSA_DEBUG_ERROR ("Failed to read data from audio interface failed (%d->%s)", err, snd_strerror(err)); + tdav_common_alsa_unlock(&p_alsa->alsa_common); + goto bail; + } + if (!p_alsa->b_muted && TMEDIA_PRODUCER(p_alsa)->enc_cb.callback) { + TMEDIA_PRODUCER(p_alsa)->enc_cb.callback(TMEDIA_PRODUCER(p_alsa)->enc_cb.callback_data, p_alsa->alsa_common.p_buff_ptr, p_alsa->alsa_common.n_buff_size_in_bytes); + } + tdav_common_alsa_unlock(&p_alsa->alsa_common); + } bail: - ALSA_DEBUG_INFO("__record_thread -- STOP"); - return tsk_null; + ALSA_DEBUG_INFO("__record_thread -- STOP"); + return tsk_null; } /* ============ Media Producer Interface ================= */ static int tdav_producer_alsa_set(tmedia_producer_t* self, const tmedia_param_t* param) -{ - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; - if (param->plugin_type == tmedia_ppt_producer) { - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "volume")) { - return 0; - } - else if(tsk_striequals(param->key, "mute")){ - p_alsa->b_muted = (TSK_TO_INT32((uint8_t*)param->value) != 0); - return 0; - } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); +{ + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; + if (param->plugin_type == tmedia_ppt_producer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "volume")) { + return 0; + } + else if(tsk_striequals(param->key, "mute")) { + p_alsa->b_muted = (TSK_TO_INT32((uint8_t*)param->value) != 0); + return 0; + } + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int tdav_producer_alsa_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; - int err = 0; - ALSA_DEBUG_INFO("******* tdav_producer_alsa_prepare ******"); - - if (! p_alsa || !codec && codec->plugin) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(&p_alsa->alsa_common); - - // Set using requested - TMEDIA_PRODUCER( p_alsa)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER( p_alsa)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); - TMEDIA_PRODUCER( p_alsa)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); - - // Prepare - err = tdav_common_alsa_prepare(&p_alsa->alsa_common, tsk_true/*is_capture*/, TMEDIA_PRODUCER( p_alsa)->audio.ptime, TMEDIA_PRODUCER( p_alsa)->audio.channels, TMEDIA_PRODUCER( p_alsa)->audio.rate); - if (err) { - goto bail; - } - - ALSA_DEBUG_INFO("prepared: req_channels=%d; req_rate=%d, resp_channels=%d; resp_rate=%d", - TMEDIA_PRODUCER(p_alsa)->audio.channels, TMEDIA_PRODUCER(p_alsa)->audio.rate, - p_alsa->alsa_common.channels, p_alsa->alsa_common.sample_rate); - - // Set using supported (up to the resampler to convert to requested) - TMEDIA_PRODUCER(p_alsa)->audio.channels = p_alsa->alsa_common.channels; - TMEDIA_PRODUCER(p_alsa)->audio.rate = p_alsa->alsa_common.sample_rate; - + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; + int err = 0; + ALSA_DEBUG_INFO("******* tdav_producer_alsa_prepare ******"); + + if (! p_alsa || !codec && codec->plugin) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tdav_common_alsa_lock(&p_alsa->alsa_common); + + // Set using requested + TMEDIA_PRODUCER( p_alsa)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER( p_alsa)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); + TMEDIA_PRODUCER( p_alsa)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + + // Prepare + err = tdav_common_alsa_prepare(&p_alsa->alsa_common, tsk_true/*is_capture*/, TMEDIA_PRODUCER( p_alsa)->audio.ptime, TMEDIA_PRODUCER( p_alsa)->audio.channels, TMEDIA_PRODUCER( p_alsa)->audio.rate); + if (err) { + goto bail; + } + + ALSA_DEBUG_INFO("prepared: req_channels=%d; req_rate=%d, resp_channels=%d; resp_rate=%d", + TMEDIA_PRODUCER(p_alsa)->audio.channels, TMEDIA_PRODUCER(p_alsa)->audio.rate, + p_alsa->alsa_common.channels, p_alsa->alsa_common.sample_rate); + + // Set using supported (up to the resampler to convert to requested) + TMEDIA_PRODUCER(p_alsa)->audio.channels = p_alsa->alsa_common.channels; + TMEDIA_PRODUCER(p_alsa)->audio.rate = p_alsa->alsa_common.sample_rate; + bail: - tdav_common_alsa_unlock(&p_alsa->alsa_common); - return err; + tdav_common_alsa_unlock(&p_alsa->alsa_common); + return err; } static int tdav_producer_alsa_start(tmedia_producer_t* self) { - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; - int err = 0; + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; + int err = 0; + + ALSA_DEBUG_INFO("******* tdav_producer_alsa_start ******"); - ALSA_DEBUG_INFO("******* tdav_producer_alsa_start ******"); + if (!p_alsa) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (!p_alsa) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tdav_common_alsa_lock(&p_alsa->alsa_common); + tdav_common_alsa_lock(&p_alsa->alsa_common); - if (p_alsa->b_started) { - ALSA_DEBUG_WARN("Already started"); - goto bail; - } + if (p_alsa->b_started) { + ALSA_DEBUG_WARN("Already started"); + goto bail; + } - /* start device */ - err = tdav_common_alsa_start(&p_alsa->alsa_common); - if (err) { - goto bail; - } + /* start device */ + err = tdav_common_alsa_start(&p_alsa->alsa_common); + if (err) { + goto bail; + } - /* start thread */ - p_alsa->b_started = tsk_true; - tsk_thread_create(&p_alsa->tid[0], _tdav_producer_alsa_record_thread, p_alsa); + /* start thread */ + p_alsa->b_started = tsk_true; + tsk_thread_create(&p_alsa->tid[0], _tdav_producer_alsa_record_thread, p_alsa); - ALSA_DEBUG_INFO("started"); + ALSA_DEBUG_INFO("started"); bail: - tdav_common_alsa_unlock(&p_alsa->alsa_common); - return err; + tdav_common_alsa_unlock(&p_alsa->alsa_common); + return err; } static int tdav_producer_alsa_pause(tmedia_producer_t* self) { - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; - if (!p_alsa) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - ALSA_DEBUG_INFO("paused"); + if (!p_alsa) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return 0; + ALSA_DEBUG_INFO("paused"); + + return 0; } static int tdav_producer_alsa_stop(tmedia_producer_t* self) { - tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; - int err; - - if (!p_alsa) { - ALSA_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - /* should be done here */ - p_alsa->b_started = tsk_false; - - err = tdav_common_alsa_stop(&p_alsa->alsa_common); - - /* stop thread */ - if (p_alsa->tid[0]) { - tsk_thread_join(&(p_alsa->tid[0])); - } - - ALSA_DEBUG_INFO("stopped"); - - return 0; + tdav_producer_alsa_t* p_alsa = (tdav_producer_alsa_t*)self; + int err; + + if (!p_alsa) { + ALSA_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + /* should be done here */ + p_alsa->b_started = tsk_false; + + err = tdav_common_alsa_stop(&p_alsa->alsa_common); + + /* stop thread */ + if (p_alsa->tid[0]) { + tsk_thread_join(&(p_alsa->tid[0])); + } + + ALSA_DEBUG_INFO("stopped"); + + return 0; } @@ -204,57 +203,55 @@ static int tdav_producer_alsa_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_alsa_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_alsa_t *p_alsa = (tdav_producer_alsa_t*)self; - if (p_alsa) { - ALSA_DEBUG_INFO("create"); - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(p_alsa)); - /* init self */ - tdav_common_alsa_init(&p_alsa->alsa_common); - } - return self; + tdav_producer_alsa_t *p_alsa = (tdav_producer_alsa_t*)self; + if (p_alsa) { + ALSA_DEBUG_INFO("create"); + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(p_alsa)); + /* init self */ + tdav_common_alsa_init(&p_alsa->alsa_common); + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_alsa_dtor(tsk_object_t * self) -{ - tdav_producer_alsa_t *p_alsa = (tdav_producer_alsa_t *)self; - if (p_alsa) { - /* stop */ - if (p_alsa->b_started) { - tdav_producer_alsa_stop((tmedia_producer_t*)p_alsa); - } - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(p_alsa)); - /* deinit self */ - tdav_common_alsa_deinit(&p_alsa->alsa_common); - - ALSA_DEBUG_INFO("*** destroyed ***"); - } - - return self; +{ + tdav_producer_alsa_t *p_alsa = (tdav_producer_alsa_t *)self; + if (p_alsa) { + /* stop */ + if (p_alsa->b_started) { + tdav_producer_alsa_stop((tmedia_producer_t*)p_alsa); + } + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(p_alsa)); + /* deinit self */ + tdav_common_alsa_deinit(&p_alsa->alsa_common); + + ALSA_DEBUG_INFO("*** destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_alsa_def_s = -{ - sizeof(tdav_producer_alsa_t), - tdav_producer_alsa_ctor, - tdav_producer_alsa_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_alsa_def_s = { + sizeof(tdav_producer_alsa_t), + tdav_producer_alsa_ctor, + tdav_producer_alsa_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_alsa_plugin_def_s = -{ - &tdav_producer_alsa_def_s, - - tmedia_audio, - "Linux ALSA producer", - - tdav_producer_alsa_set, - tdav_producer_alsa_prepare, - tdav_producer_alsa_start, - tdav_producer_alsa_pause, - tdav_producer_alsa_stop +static const tmedia_producer_plugin_def_t tdav_producer_alsa_plugin_def_s = { + &tdav_producer_alsa_def_s, + + tmedia_audio, + "Linux ALSA producer", + + tdav_producer_alsa_set, + tdav_producer_alsa_prepare, + tdav_producer_alsa_start, + tdav_producer_alsa_pause, + tdav_producer_alsa_stop }; const tmedia_producer_plugin_def_t *tdav_producer_alsa_plugin_def_t = &tdav_producer_alsa_plugin_def_s; diff --git a/tinyDAV/src/audio/coreaudio/tdav_audiounit.c b/tinyDAV/src/audio/coreaudio/tdav_audiounit.c index dc11f10..d00f8ee 100755 --- a/tinyDAV/src/audio/coreaudio/tdav_audiounit.c +++ b/tinyDAV/src/audio/coreaudio/tdav_audiounit.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> - * + * * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. - * + * * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -36,15 +36,15 @@ static UInt32 kZero = 0; #endif /* TARGET_OS_IPHONE */ #if TARGET_OS_IPHONE - #if TARGET_IPHONE_SIMULATOR // VoiceProcessingIO will give unexpected result on the simulator when using iOS 5 - #define kDoubangoAudioUnitSubType kAudioUnitSubType_RemoteIO - #else // Echo cancellation, AGC, ... - #define kDoubangoAudioUnitSubType kAudioUnitSubType_VoiceProcessingIO - #endif +#if TARGET_IPHONE_SIMULATOR // VoiceProcessingIO will give unexpected result on the simulator when using iOS 5 +#define kDoubangoAudioUnitSubType kAudioUnitSubType_RemoteIO +#else // Echo cancellation, AGC, ... +#define kDoubangoAudioUnitSubType kAudioUnitSubType_VoiceProcessingIO +#endif #elif TARGET_OS_MAC - #define kDoubangoAudioUnitSubType kAudioUnitSubType_HALOutput +#define kDoubangoAudioUnitSubType kAudioUnitSubType_HALOutput #else - #error "Unknown target" +#error "Unknown target" #endif #undef kInputBus @@ -52,21 +52,20 @@ static UInt32 kZero = 0; #undef kOutputBus #define kOutputBus 0 -typedef struct tdav_audiounit_instance_s -{ - TSK_DECLARE_OBJECT; - uint64_t session_id; - uint32_t frame_duration; - AudioComponentInstance audioUnit; - struct{ - unsigned consumer:1; - unsigned producer:1; - } prepared; - unsigned started:1; +typedef struct tdav_audiounit_instance_s { + TSK_DECLARE_OBJECT; + uint64_t session_id; + uint32_t frame_duration; + AudioComponentInstance audioUnit; + struct { + unsigned consumer:1; + unsigned producer:1; + } prepared; + unsigned started:1; unsigned interrupted:1; - - TSK_DECLARE_SAFEOBJ; - + + TSK_DECLARE_SAFEOBJ; + } tdav_audiounit_instance_t; TINYDAV_GEXTERN const tsk_object_def_t *tdav_audiounit_instance_def_t; @@ -78,133 +77,133 @@ static tdav_audiounit_instances_L_t* __audioUnitInstances = tsk_null; static int _tdav_audiounit_handle_signal_xxx_prepared(tdav_audiounit_handle_t* self, tsk_bool_t consumer) { - tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - if(!inst || !inst->audioUnit){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(inst); - - if(consumer){ - inst->prepared.consumer = tsk_true; - } - else { - inst->prepared.producer = tsk_true; - } - - OSStatus status; - - // For iOS we are using full-duplex AudioUnit and we wait for both consumer and producer to be prepared + tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; + if(!inst || !inst->audioUnit) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(inst); + + if(consumer) { + inst->prepared.consumer = tsk_true; + } + else { + inst->prepared.producer = tsk_true; + } + + OSStatus status; + + // For iOS we are using full-duplex AudioUnit and we wait for both consumer and producer to be prepared #if TARGET_OS_IPHONE - if(inst->prepared.consumer && inst->prepared.producer) + if(inst->prepared.consumer && inst->prepared.producer) #endif - { - status = AudioUnitInitialize(inst->audioUnit); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitInitialize failed with status =%ld", (signed long)status); - tsk_safeobj_unlock(inst); - return -2; - } - } - - tsk_safeobj_unlock(inst); - return 0; + { + status = AudioUnitInitialize(inst->audioUnit); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitInitialize failed with status =%ld", (signed long)status); + tsk_safeobj_unlock(inst); + return -2; + } + } + + tsk_safeobj_unlock(inst); + return 0; } tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id) { - tdav_audiounit_instance_t* inst = tsk_null; - - // create audio unit component - if(!__audioSystem){ - AudioComponentDescription audioDescription; - audioDescription.componentType = kAudioUnitType_Output; - audioDescription.componentSubType = kDoubangoAudioUnitSubType; - audioDescription.componentManufacturer = kAudioUnitManufacturer_Apple; - audioDescription.componentFlags = 0; - audioDescription.componentFlagsMask = 0; - if((__audioSystem = AudioComponentFindNext(NULL, &audioDescription))){ - // leave blank - } - else { - TSK_DEBUG_ERROR("Failed to find new audio component"); - goto done; - } - - } - // create list used to hold instances - if(!__audioUnitInstances && !(__audioUnitInstances = tsk_list_create())){ - TSK_DEBUG_ERROR("Failed to create new list"); - goto done; - } - - //= lock the list - tsk_list_lock(__audioUnitInstances); - - // For iOS we are using full-duplex AudioUnit and to keep it unique for both - // the consumer and producer we use the session id. + tdav_audiounit_instance_t* inst = tsk_null; + + // create audio unit component + if(!__audioSystem) { + AudioComponentDescription audioDescription; + audioDescription.componentType = kAudioUnitType_Output; + audioDescription.componentSubType = kDoubangoAudioUnitSubType; + audioDescription.componentManufacturer = kAudioUnitManufacturer_Apple; + audioDescription.componentFlags = 0; + audioDescription.componentFlagsMask = 0; + if((__audioSystem = AudioComponentFindNext(NULL, &audioDescription))) { + // leave blank + } + else { + TSK_DEBUG_ERROR("Failed to find new audio component"); + goto done; + } + + } + // create list used to hold instances + if(!__audioUnitInstances && !(__audioUnitInstances = tsk_list_create())) { + TSK_DEBUG_ERROR("Failed to create new list"); + goto done; + } + + //= lock the list + tsk_list_lock(__audioUnitInstances); + + // For iOS we are using full-duplex AudioUnit and to keep it unique for both + // the consumer and producer we use the session id. #if TARGET_OS_IPHONE - // find the instance from the list - const tsk_list_item_t* item; - tsk_list_foreach(item,__audioUnitInstances){ - if(((tdav_audiounit_instance_t*)item->data)->session_id == session_id){ - inst = tsk_object_ref(item->data); - goto done; - } - } + // find the instance from the list + const tsk_list_item_t* item; + tsk_list_foreach(item,__audioUnitInstances) { + if(((tdav_audiounit_instance_t*)item->data)->session_id == session_id) { + inst = tsk_object_ref(item->data); + goto done; + } + } #endif - - // create instance object and put it into the list - if((inst = tsk_object_new(tdav_audiounit_instance_def_t))){ - OSStatus status = noErr; - tdav_audiounit_instance_t* _inst; - - // create new instance - if((status= AudioComponentInstanceNew(__audioSystem, &inst->audioUnit)) != noErr){ - TSK_DEBUG_ERROR("AudioComponentInstanceNew() failed with status=%ld", (signed long)status); - TSK_OBJECT_SAFE_FREE(inst); - goto done; - } - _inst = inst, _inst->session_id = session_id; - tsk_list_push_back_data(__audioUnitInstances, (void**)&_inst); - } - + + // create instance object and put it into the list + if((inst = tsk_object_new(tdav_audiounit_instance_def_t))) { + OSStatus status = noErr; + tdav_audiounit_instance_t* _inst; + + // create new instance + if((status= AudioComponentInstanceNew(__audioSystem, &inst->audioUnit)) != noErr) { + TSK_DEBUG_ERROR("AudioComponentInstanceNew() failed with status=%ld", (signed long)status); + TSK_OBJECT_SAFE_FREE(inst); + goto done; + } + _inst = inst, _inst->session_id = session_id; + tsk_list_push_back_data(__audioUnitInstances, (void**)&_inst); + } + done: - //= unlock the list - tsk_list_unlock(__audioUnitInstances); - return (tdav_audiounit_handle_t*)inst; + //= unlock the list + tsk_list_unlock(__audioUnitInstances); + return (tdav_audiounit_handle_t*)inst; } AudioComponentInstance tdav_audiounit_handle_get_instance(tdav_audiounit_handle_t* self) { - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return tsk_null; - } - return ((tdav_audiounit_instance_t*)self)->audioUnit; + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return tsk_null; + } + return ((tdav_audiounit_instance_t*)self)->audioUnit; } int tdav_audiounit_handle_signal_consumer_prepared(tdav_audiounit_handle_t* self) { - return _tdav_audiounit_handle_signal_xxx_prepared(self, tsk_true); + return _tdav_audiounit_handle_signal_xxx_prepared(self, tsk_true); } int tdav_audiounit_handle_signal_producer_prepared(tdav_audiounit_handle_t* self) { - return _tdav_audiounit_handle_signal_xxx_prepared(self, tsk_false); + return _tdav_audiounit_handle_signal_xxx_prepared(self, tsk_false); } int tdav_audiounit_handle_start(tdav_audiounit_handle_t* self) { - tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - OSStatus status = noErr; - if(!inst || !inst->audioUnit){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(inst); + tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; + OSStatus status = noErr; + if(!inst || !inst->audioUnit) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(inst); status = (OSStatus)tdav_apple_enable_audio(); if (status == noErr) { if ((!inst->started || inst->interrupted) && (status = AudioOutputUnitStart(inst->audioUnit))) { @@ -215,106 +214,108 @@ int tdav_audiounit_handle_start(tdav_audiounit_handle_t* self) TSK_DEBUG_ERROR("tdav_apple_enable_audio() failed with status=%ld", (signed long)status); } inst->started = (status == noErr) ? tsk_true : tsk_false; - if (inst->started) inst->interrupted = 0; - tsk_safeobj_unlock(inst); - return status ? -2 : 0; + if (inst->started) { + inst->interrupted = 0; + } + tsk_safeobj_unlock(inst); + return status ? -2 : 0; } uint32_t tdav_audiounit_handle_get_frame_duration(tdav_audiounit_handle_t* self) { - if(self){ - return ((tdav_audiounit_instance_t*)self)->frame_duration; - } - return 0; + if(self) { + return ((tdav_audiounit_instance_t*)self)->frame_duration; + } + return 0; } int tdav_audiounit_handle_configure(tdav_audiounit_handle_t* self, tsk_bool_t consumer, uint32_t ptime, AudioStreamBasicDescription* audioFormat) { - OSStatus status = noErr; - tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - - if(!inst || !audioFormat){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + OSStatus status = noErr; + tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; + + if(!inst || !audioFormat) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } #if TARGET_OS_IPHONE - // set preferred buffer size - Float32 preferredBufferSize = ((Float32)ptime / 1000.f); // in seconds - UInt32 size = sizeof(preferredBufferSize); - status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(preferredBufferSize), &preferredBufferSize); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration) failed with status=%d", (int)status); - TSK_OBJECT_SAFE_FREE(inst); - goto done; - } - status = AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, &size, &preferredBufferSize); - if(status == noErr){ - inst->frame_duration = (preferredBufferSize * 1000); - TSK_DEBUG_INFO("Frame duration=%d", inst->frame_duration); - } - else { - TSK_DEBUG_ERROR("AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, %f) failed", preferredBufferSize); - } - - - UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord; - status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(audioCategory), &audioCategory); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioSessionSetProperty(kAudioSessionProperty_AudioCategory) failed with status code=%d", (int)status); - goto done; - } - + // set preferred buffer size + Float32 preferredBufferSize = ((Float32)ptime / 1000.f); // in seconds + UInt32 size = sizeof(preferredBufferSize); + status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(preferredBufferSize), &preferredBufferSize); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration) failed with status=%d", (int)status); + TSK_OBJECT_SAFE_FREE(inst); + goto done; + } + status = AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, &size, &preferredBufferSize); + if(status == noErr) { + inst->frame_duration = (preferredBufferSize * 1000); + TSK_DEBUG_INFO("Frame duration=%d", inst->frame_duration); + } + else { + TSK_DEBUG_ERROR("AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, %f) failed", preferredBufferSize); + } + + + UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord; + status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(audioCategory), &audioCategory); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioSessionSetProperty(kAudioSessionProperty_AudioCategory) failed with status code=%d", (int)status); + goto done; + } + #elif TARGET_OS_MAC #if 1 - // set preferred buffer size - UInt32 preferredBufferSize = ((ptime * audioFormat->mSampleRate)/1000); // in bytes - UInt32 size = sizeof(preferredBufferSize); - status = AudioUnitSetProperty(inst->audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &preferredBufferSize, size); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); - } - status = AudioUnitGetProperty(inst->audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &preferredBufferSize, &size); - if(status == noErr){ - inst->frame_duration = ((preferredBufferSize * 1000)/audioFormat->mSampleRate); - TSK_DEBUG_INFO("Frame duration=%d", inst->frame_duration); - } - else { - TSK_DEBUG_ERROR("AudioUnitGetProperty(kAudioDevicePropertyBufferFrameSize, %lu) failed", (unsigned long)preferredBufferSize); - } + // set preferred buffer size + UInt32 preferredBufferSize = ((ptime * audioFormat->mSampleRate)/1000); // in bytes + UInt32 size = sizeof(preferredBufferSize); + status = AudioUnitSetProperty(inst->audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &preferredBufferSize, size); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); + } + status = AudioUnitGetProperty(inst->audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &preferredBufferSize, &size); + if(status == noErr) { + inst->frame_duration = ((preferredBufferSize * 1000)/audioFormat->mSampleRate); + TSK_DEBUG_INFO("Frame duration=%d", inst->frame_duration); + } + else { + TSK_DEBUG_ERROR("AudioUnitGetProperty(kAudioDevicePropertyBufferFrameSize, %lu) failed", (unsigned long)preferredBufferSize); + } #endif - + #endif - + done: - return (status == noErr) ? 0 : -2; + return (status == noErr) ? 0 : -2; } int tdav_audiounit_handle_mute(tdav_audiounit_handle_t* self, tsk_bool_t mute) { - tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - if(!inst || !inst->audioUnit){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; + if(!inst || !inst->audioUnit) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } #if TARGET_OS_IPHONE - OSStatus status = noErr; - status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_MuteOutput, - kAudioUnitScope_Output, kOutputBus, mute ? &kOne : &kZero, mute ? sizeof(kOne) : sizeof(kZero)); - - return (status == noErr) ? 0 : -2; + OSStatus status = noErr; + status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_MuteOutput, + kAudioUnitScope_Output, kOutputBus, mute ? &kOne : &kZero, mute ? sizeof(kOne) : sizeof(kZero)); + + return (status == noErr) ? 0 : -2; #else - return 0; + return 0; #endif } int tdav_audiounit_handle_interrupt(tdav_audiounit_handle_t* self, tsk_bool_t interrupt) { tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - if (!inst){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!inst) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } OSStatus status = noErr; if (inst->interrupted != interrupt && inst->started) { if (interrupt) { @@ -346,37 +347,38 @@ bail: int tdav_audiounit_handle_stop(tdav_audiounit_handle_t* self) { - tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; - OSStatus status = noErr; - if(!inst || (inst->started && !inst->audioUnit)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(inst); - if(inst->started && (status = AudioOutputUnitStop(inst->audioUnit))){ - TSK_DEBUG_ERROR("AudioOutputUnitStop failed with status=%ld", (signed long)status); - } + tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self; + OSStatus status = noErr; + if(!inst || (inst->started && !inst->audioUnit)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(inst); + if(inst->started && (status = AudioOutputUnitStop(inst->audioUnit))) { + TSK_DEBUG_ERROR("AudioOutputUnitStop failed with status=%ld", (signed long)status); + } inst->started = (status == noErr ? tsk_false : tsk_true); - tsk_safeobj_unlock(inst); - return (status != noErr) ? -2 : 0; + tsk_safeobj_unlock(inst); + return (status != noErr) ? -2 : 0; } -int tdav_audiounit_handle_destroy(tdav_audiounit_handle_t** self){ - if(!self || !*self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - tsk_list_lock(__audioUnitInstances); - if(tsk_object_get_refcount(*self)==1){ - tsk_list_remove_item_by_data(__audioUnitInstances, *self); - } - else { - tsk_object_unref(*self); - } - tsk_list_unlock(__audioUnitInstances); - *self = tsk_null; - return 0; +int tdav_audiounit_handle_destroy(tdav_audiounit_handle_t** self) +{ + if(!self || !*self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + tsk_list_lock(__audioUnitInstances); + if(tsk_object_get_refcount(*self)==1) { + tsk_list_remove_item_by_data(__audioUnitInstances, *self); + } + else { + tsk_object_unref(*self); + } + tsk_list_unlock(__audioUnitInstances); + *self = tsk_null; + return 0; } // @@ -384,39 +386,38 @@ int tdav_audiounit_handle_destroy(tdav_audiounit_handle_t** self){ // static tsk_object_t* tdav_audiounit_instance_ctor(tsk_object_t * self, va_list * app) { - tdav_audiounit_instance_t* inst = self; - if(inst){ - tsk_safeobj_init(inst); - } - return self; + tdav_audiounit_instance_t* inst = self; + if(inst) { + tsk_safeobj_init(inst); + } + return self; } static tsk_object_t* tdav_audiounit_instance_dtor(tsk_object_t * self) -{ - tdav_audiounit_instance_t* inst = self; - if(inst){ +{ + tdav_audiounit_instance_t* inst = self; + if(inst) { tsk_safeobj_lock(inst); - if(inst->audioUnit){ + if(inst->audioUnit) { AudioUnitUninitialize(inst->audioUnit); AudioComponentInstanceDispose(inst->audioUnit); inst->audioUnit = tsk_null; - } + } tsk_safeobj_unlock(inst); - - tsk_safeobj_deinit(inst); + + tsk_safeobj_deinit(inst); TSK_DEBUG_INFO("*** AudioUnit Instance destroyed ***"); - } - return self; + } + return self; } static int tdav_audiounit_instance_cmp(const tsk_object_t *_ai1, const tsk_object_t *_ai2) { - return (int)(_ai1 - _ai2); + return (int)(_ai1 - _ai2); } -static const tsk_object_def_t tdav_audiounit_instance_def_s = -{ - sizeof(tdav_audiounit_instance_t), - tdav_audiounit_instance_ctor, - tdav_audiounit_instance_dtor, - tdav_audiounit_instance_cmp, +static const tsk_object_def_t tdav_audiounit_instance_def_s = { + sizeof(tdav_audiounit_instance_t), + tdav_audiounit_instance_ctor, + tdav_audiounit_instance_dtor, + tdav_audiounit_instance_cmp, }; const tsk_object_def_t *tdav_audiounit_instance_def_t = &tdav_audiounit_instance_def_s; diff --git a/tinyDAV/src/audio/coreaudio/tdav_consumer_audioqueue.c b/tinyDAV/src/audio/coreaudio/tdav_consumer_audioqueue.c index 2f5fd90..882a988 100755 --- a/tinyDAV/src/audio/coreaudio/tdav_consumer_audioqueue.c +++ b/tinyDAV/src/audio/coreaudio/tdav_consumer_audioqueue.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> - * + * * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. - * + * * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -23,7 +23,7 @@ /**@file tdav_consumer_audioqueue.c * @brief Audio Consumer for MacOSX and iOS platforms. * - * @authors + * @authors * - Laurent Etiemble <laurent.etiemble(at)gmail.com> * - Mamadou Diop <diopmamadou(at)doubango(dot)org> * @@ -40,22 +40,23 @@ #include "tsk_memory.h" #include "tsk_debug.h" -static void __handle_output_buffer(void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer) { +static void __handle_output_buffer(void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer) +{ tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)userdata; - + if (!consumer->started) { return; } - - if(!tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), buffer->mAudioData, consumer->buffer_size)){ - // Put silence - memset(buffer->mAudioData, 0, consumer->buffer_size); - } - + + if(!tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), buffer->mAudioData, consumer->buffer_size)) { + // Put silence + memset(buffer->mAudioData, 0, consumer->buffer_size); + } + // Re-enqueue the buffer AudioQueueEnqueueBuffer(consumer->queue, buffer, 0, NULL); - // alert the jitter buffer - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(consumer)); + // alert the jitter buffer + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(consumer)); } /* ============ Media Consumer Interface ================= */ @@ -64,25 +65,25 @@ static void __handle_output_buffer(void *userdata, AudioQueueRef queue, AudioQue int tdav_consumer_audioqueue_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { OSStatus ret; - tsk_size_t i; - tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; - - if(!consumer || !codec && codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - /* codec should have ptime */ - - // Set audio category + tsk_size_t i; + tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; + + if(!consumer || !codec && codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); + /* codec should have ptime */ + + // Set audio category #if TARGET_OS_IPHONE - UInt32 category = kAudioSessionCategory_PlayAndRecord; - AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); + UInt32 category = kAudioSessionCategory_PlayAndRecord; + AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); #endif - + // Create the audio stream description AudioStreamBasicDescription *description = &(consumer->description); description->mSampleRate = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate; @@ -94,107 +95,107 @@ int tdav_consumer_audioqueue_prepare(tmedia_consumer_t* self, const tmedia_codec description->mBytesPerPacket = description->mBitsPerChannel / 8 * description->mChannelsPerFrame; description->mBytesPerFrame = description->mBytesPerPacket; description->mReserved = 0; - + int packetperbuffer = 1000 / TMEDIA_CONSUMER(consumer)->audio.ptime; consumer->buffer_size = description->mSampleRate * description->mBytesPerFrame / packetperbuffer; - + // Create the playback audio queue ret = AudioQueueNewOutput(&(consumer->description), __handle_output_buffer, consumer, - NULL, + NULL, NULL, 0, &(consumer->queue)); - + for(i = 0; i < CoreAudioPlayBuffers; i++) { // Create the buffer for the queue ret = AudioQueueAllocateBuffer(consumer->queue, consumer->buffer_size, &(consumer->buffers[i])); if (ret) { break; } - + // Clear the data memset(consumer->buffers[i]->mAudioData, 0, consumer->buffer_size); consumer->buffers[i]->mAudioDataByteSize = consumer->buffer_size; - + // Enqueue the buffer ret = AudioQueueEnqueueBuffer(consumer->queue, consumer->buffers[i], 0, NULL); if (ret) { break; } } - - return ret; + + return ret; } int tdav_consumer_audioqueue_start(tmedia_consumer_t* self) { OSStatus ret; - tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; - - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(consumer->started){ - TSK_DEBUG_WARN("Consumer already started"); - return 0; - } - - consumer->started = tsk_true; + tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; + + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(consumer->started) { + TSK_DEBUG_WARN("Consumer already started"); + return 0; + } + + consumer->started = tsk_true; ret = AudioQueueStart(consumer->queue, NULL); - - return ret; + + return ret; } int tdav_consumer_audioqueue_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; - - if(!consumer || !buffer || !size){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - // buffer is already decoded - return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); + tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; + + if(!consumer || !buffer || !size) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + // buffer is already decoded + return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); } int tdav_consumer_audioqueue_pause(tmedia_consumer_t* self) { OSStatus ret; - tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; - - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - + tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; + + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + ret = AudioQueuePause(consumer->queue); - - return ret; + + return ret; } int tdav_consumer_audioqueue_stop(tmedia_consumer_t* self) { OSStatus ret; - tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; - - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(!consumer->started){ - TSK_DEBUG_WARN("Consumer not started"); - return 0; - } - - consumer->started = tsk_false; + tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self; + + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(!consumer->started) { + TSK_DEBUG_WARN("Consumer not started"); + return 0; + } + + consumer->started = tsk_false; ret = AudioQueueStop(consumer->queue, false); - - return ret; + + return ret; } // @@ -203,64 +204,62 @@ int tdav_consumer_audioqueue_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_audioqueue_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_audioqueue_t *consumer = self; - if(consumer){ - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); - } - return self; + tdav_consumer_audioqueue_t *consumer = self; + if(consumer) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_audioqueue_dtor(tsk_object_t * self) -{ - tdav_consumer_audioqueue_t *consumer = self; - if(consumer){ - // Stop the consumer if not done - if(consumer->started){ - tdav_consumer_audioqueue_stop(self); - } - - // Free all buffers and dispose the queue +{ + tdav_consumer_audioqueue_t *consumer = self; + if(consumer) { + // Stop the consumer if not done + if(consumer->started) { + tdav_consumer_audioqueue_stop(self); + } + + // Free all buffers and dispose the queue if (consumer->queue) { - tsk_size_t i; - - for(i=0; i<CoreAudioPlayBuffers; i++){ - AudioQueueFreeBuffer(consumer->queue, consumer->buffers[i]); - } - + tsk_size_t i; + + for(i=0; i<CoreAudioPlayBuffers; i++) { + AudioQueueFreeBuffer(consumer->queue, consumer->buffers[i]); + } + AudioQueueDispose(consumer->queue, true); } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); - } - - return self; + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_audioqueue_def_s = -{ - sizeof(tdav_consumer_audioqueue_t), - tdav_consumer_audioqueue_ctor, - tdav_consumer_audioqueue_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_audioqueue_def_s = { + sizeof(tdav_consumer_audioqueue_t), + tdav_consumer_audioqueue_ctor, + tdav_consumer_audioqueue_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_audioqueue_plugin_def_s = -{ - &tdav_consumer_audioqueue_def_s, - - tmedia_audio, - "Apple CoreAudio consumer(AudioQueue)", - - tdav_consumer_audioqueue_set, - tdav_consumer_audioqueue_prepare, - tdav_consumer_audioqueue_start, - tdav_consumer_audioqueue_consume, - tdav_consumer_audioqueue_pause, - tdav_consumer_audioqueue_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_audioqueue_plugin_def_s = { + &tdav_consumer_audioqueue_def_s, + + tmedia_audio, + "Apple CoreAudio consumer(AudioQueue)", + + tdav_consumer_audioqueue_set, + tdav_consumer_audioqueue_prepare, + tdav_consumer_audioqueue_start, + tdav_consumer_audioqueue_consume, + tdav_consumer_audioqueue_pause, + tdav_consumer_audioqueue_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_audioqueue_plugin_def_t = &tdav_consumer_audioqueue_plugin_def_s; diff --git a/tinyDAV/src/audio/coreaudio/tdav_consumer_audiounit.c b/tinyDAV/src/audio/coreaudio/tdav_consumer_audiounit.c index 947d782..12ed8db 100755 --- a/tinyDAV/src/audio/coreaudio/tdav_consumer_audiounit.c +++ b/tinyDAV/src/audio/coreaudio/tdav_consumer_audiounit.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> - * + * * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. - * + * * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -38,339 +38,340 @@ static tsk_size_t tdav_consumer_audiounit_get(tdav_consumer_audiounit_t* self, void* data, tsk_size_t size); -static OSStatus __handle_output_buffer(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *inTimeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData) { - OSStatus status = noErr; - // tsk_size_t out_size; - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t* )inRefCon; - - if(!consumer->started || consumer->paused){ - goto done; - } - - if(!ioData){ - TSK_DEBUG_ERROR("Invalid argument"); - status = kNoDataError; - goto done; - } - // read from jitter buffer and fill ioData buffers - tsk_mutex_lock(consumer->ring.mutex); - for(int i=0; i<ioData->mNumberBuffers; i++){ - /* int ret = */ tdav_consumer_audiounit_get(consumer, ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize); - } - tsk_mutex_unlock(consumer->ring.mutex); - -done: +static OSStatus __handle_output_buffer(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + OSStatus status = noErr; + // tsk_size_t out_size; + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t* )inRefCon; + + if(!consumer->started || consumer->paused) { + goto done; + } + + if(!ioData) { + TSK_DEBUG_ERROR("Invalid argument"); + status = kNoDataError; + goto done; + } + // read from jitter buffer and fill ioData buffers + tsk_mutex_lock(consumer->ring.mutex); + for(int i=0; i<ioData->mNumberBuffers; i++) { + /* int ret = */ tdav_consumer_audiounit_get(consumer, ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize); + } + tsk_mutex_unlock(consumer->ring.mutex); + +done: return status; } static tsk_size_t tdav_consumer_audiounit_get(tdav_consumer_audiounit_t* self, void* data, tsk_size_t size) { - tsk_ssize_t retSize = 0; - + tsk_ssize_t retSize = 0; + #if DISABLE_JITTER_BUFFER - retSize = speex_buffer_read(self->ring.buffer, data, size); - if(retSize < size){ - memset(((uint8_t*)data)+retSize, 0, (size - retSize)); - } + retSize = speex_buffer_read(self->ring.buffer, data, size); + if(retSize < size) { + memset(((uint8_t*)data)+retSize, 0, (size - retSize)); + } #else - self->ring.leftBytes += size; - while (self->ring.leftBytes >= self->ring.chunck.size) { - self->ring.leftBytes -= self->ring.chunck.size; - retSize = (tsk_ssize_t)tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->ring.chunck.buffer, self->ring.chunck.size); - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self)); - speex_buffer_write(self->ring.buffer, self->ring.chunck.buffer, retSize); - } - // IMPORTANT: looks like there is a bug in speex: continously trying to read more than avail - // many times can corrupt the buffer. At least on OS X 1.5 - if(speex_buffer_get_available(self->ring.buffer) >= size){ - retSize = (tsk_ssize_t)speex_buffer_read(self->ring.buffer, data, (int)size); - } - else{ - memset(data, 0, size); - } + self->ring.leftBytes += size; + while (self->ring.leftBytes >= self->ring.chunck.size) { + self->ring.leftBytes -= self->ring.chunck.size; + retSize = (tsk_ssize_t)tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), self->ring.chunck.buffer, self->ring.chunck.size); + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(self)); + speex_buffer_write(self->ring.buffer, self->ring.chunck.buffer, retSize); + } + // IMPORTANT: looks like there is a bug in speex: continously trying to read more than avail + // many times can corrupt the buffer. At least on OS X 1.5 + if(speex_buffer_get_available(self->ring.buffer) >= size) { + retSize = (tsk_ssize_t)speex_buffer_read(self->ring.buffer, data, (int)size); + } + else { + memset(data, 0, size); + } #endif - return retSize; + return retSize; } /* ============ Media Consumer Interface ================= */ int tdav_consumer_audiounit_set(tmedia_consumer_t* self, const tmedia_param_t* param) { tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - if (param->plugin_type == tmedia_ppt_consumer) { - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "interrupt")) { - int32_t interrupt = *((uint8_t*)param->value) ? 1 : 0; + if (param->plugin_type == tmedia_ppt_consumer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "interrupt")) { + int32_t interrupt = *((uint8_t*)param->value) ? 1 : 0; return tdav_audiounit_handle_interrupt(consumer->audioUnitHandle, interrupt); } - } - } - return tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); + } + } + return tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); } static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - static UInt32 flagOne = 1; - AudioStreamBasicDescription audioFormat; + static UInt32 flagOne = 1; + AudioStreamBasicDescription audioFormat; #define kOutputBus 0 - - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - OSStatus status = noErr; - - if(!consumer || !codec || !codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(!consumer->audioUnitHandle){ - if(!(consumer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_CONSUMER(consumer)->session_id))){ - TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_CONSUMER(consumer)->session_id); - return -3; - } - } - - // enable - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), - kAudioOutputUnitProperty_EnableIO, - kAudioUnitScope_Output, - kOutputBus, - &flagOne, - sizeof(flagOne)); - if(status){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%d", (int32_t)status); - return -4; - } - else { - + + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; + OSStatus status = noErr; + + if(!consumer || !codec || !codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(!consumer->audioUnitHandle) { + if(!(consumer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_CONSUMER(consumer)->session_id))) { + TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_CONSUMER(consumer)->session_id); + return -3; + } + } + + // enable + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, + kOutputBus, + &flagOne, + sizeof(flagOne)); + if(status) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%d", (int32_t)status); + return -4; + } + else { + #if !TARGET_OS_IPHONE // strange: TARGET_OS_MAC is equal to '1' on Smulator - UInt32 param; - - // disable input - param = 0; - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, ¶m, sizeof(UInt32)); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); - return -4; - } - - // set default audio device - param = sizeof(AudioDeviceID); - AudioDeviceID outputDeviceID; - status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, ¶m, &outputDeviceID); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice) failed with status=%ld", (signed long)status); - return -4; - } - - // set the current device to the default input unit - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), - kAudioOutputUnitProperty_CurrentDevice, - kAudioUnitScope_Global, - 0, - &outputDeviceID, - sizeof(AudioDeviceID)); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_CurrentDevice) failed with status=%ld", (signed long)status); - return -4; - } - + UInt32 param; + + // disable input + param = 0; + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, ¶m, sizeof(UInt32)); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); + return -4; + } + + // set default audio device + param = sizeof(AudioDeviceID); + AudioDeviceID outputDeviceID; + status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, ¶m, &outputDeviceID); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice) failed with status=%ld", (signed long)status); + return -4; + } + + // set the current device to the default input unit + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, + 0, + &outputDeviceID, + sizeof(AudioDeviceID)); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_CurrentDevice) failed with status=%ld", (signed long)status); + return -4; + } + #endif - TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - + TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); + TSK_DEBUG_INFO("AudioUnit consumer: in.channels=%d, out.channles=%d, in.rate=%d, out.rate=%d, ptime=%d", TMEDIA_CONSUMER(consumer)->audio.in.channels, TMEDIA_CONSUMER(consumer)->audio.out.channels, TMEDIA_CONSUMER(consumer)->audio.in.rate, TMEDIA_CONSUMER(consumer)->audio.out.rate, TMEDIA_CONSUMER(consumer)->audio.ptime); - - audioFormat.mSampleRate = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate; - audioFormat.mFormatID = kAudioFormatLinearPCM; - audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; - audioFormat.mChannelsPerFrame = TMEDIA_CONSUMER(consumer)->audio.in.channels; - audioFormat.mFramesPerPacket = 1; - audioFormat.mBitsPerChannel = TMEDIA_CONSUMER(consumer)->audio.bits_per_sample; - audioFormat.mBytesPerPacket = audioFormat.mBitsPerChannel / 8 * audioFormat.mChannelsPerFrame; - audioFormat.mBytesPerFrame = audioFormat.mBytesPerPacket; - audioFormat.mReserved = 0; - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), - kAudioUnitProperty_StreamFormat, - kAudioUnitScope_Input, - kOutputBus, - &audioFormat, - sizeof(audioFormat)); - - if(status){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed with status=%ld", (signed long)status); - return -5; - } - else { - // configure - if(tdav_audiounit_handle_configure(consumer->audioUnitHandle, tsk_true, TMEDIA_CONSUMER(consumer)->audio.ptime, &audioFormat)){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_set_rate(%d) failed", TMEDIA_CONSUMER(consumer)->audio.out.rate); - return -4; - } - - // set callback function - AURenderCallbackStruct callback; - callback.inputProc = __handle_output_buffer; - callback.inputProcRefCon = consumer; - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), - kAudioUnitProperty_SetRenderCallback, - kAudioUnitScope_Input, - kOutputBus, - &callback, - sizeof(callback)); - if(status){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); - return -6; - } - } - } - - // allocate the chunck buffer and create the ring - consumer->ring.chunck.size = (TMEDIA_CONSUMER(consumer)->audio.ptime * audioFormat.mSampleRate * audioFormat.mBytesPerFrame) / 1000; - consumer->ring.size = kRingPacketCount * consumer->ring.chunck.size; - if(!(consumer->ring.chunck.buffer = tsk_realloc(consumer->ring.chunck.buffer, consumer->ring.chunck.size))){ - TSK_DEBUG_ERROR("Failed to allocate new buffer"); - return -7; - } - if(!consumer->ring.buffer){ - consumer->ring.buffer = speex_buffer_init((int)consumer->ring.size); - } - else { - int ret; - if((ret = (int)speex_buffer_resize(consumer->ring.buffer, (int)consumer->ring.size)) < 0){ - TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", (int)consumer->ring.size, ret); - return ret; - } - } - if(!consumer->ring.buffer){ - TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", (int)consumer->ring.size); - return -8; - } - if(!consumer->ring.mutex && !(consumer->ring.mutex = tsk_mutex_create_2(tsk_false))){ - TSK_DEBUG_ERROR("Failed to create mutex"); - return -9; - } - - // set maximum frames per slice as buffer size - //UInt32 numFrames = (UInt32)consumer->ring.chunck.size; - //status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), - // kAudioUnitProperty_MaximumFramesPerSlice, - // kAudioUnitScope_Global, - // 0, - // &numFrames, - // sizeof(numFrames)); - //if(status){ - // TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_MaximumFramesPerSlice, %u) failed with status=%d", (unsigned)numFrames, (int32_t)status); - // return -6; - //} - - TSK_DEBUG_INFO("AudioUnit consumer prepared"); + + audioFormat.mSampleRate = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate; + audioFormat.mFormatID = kAudioFormatLinearPCM; + audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; + audioFormat.mChannelsPerFrame = TMEDIA_CONSUMER(consumer)->audio.in.channels; + audioFormat.mFramesPerPacket = 1; + audioFormat.mBitsPerChannel = TMEDIA_CONSUMER(consumer)->audio.bits_per_sample; + audioFormat.mBytesPerPacket = audioFormat.mBitsPerChannel / 8 * audioFormat.mChannelsPerFrame; + audioFormat.mBytesPerFrame = audioFormat.mBytesPerPacket; + audioFormat.mReserved = 0; + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + kOutputBus, + &audioFormat, + sizeof(audioFormat)); + + if(status) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed with status=%ld", (signed long)status); + return -5; + } + else { + // configure + if(tdav_audiounit_handle_configure(consumer->audioUnitHandle, tsk_true, TMEDIA_CONSUMER(consumer)->audio.ptime, &audioFormat)) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_set_rate(%d) failed", TMEDIA_CONSUMER(consumer)->audio.out.rate); + return -4; + } + + // set callback function + AURenderCallbackStruct callback; + callback.inputProc = __handle_output_buffer; + callback.inputProcRefCon = consumer; + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), + kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Input, + kOutputBus, + &callback, + sizeof(callback)); + if(status) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); + return -6; + } + } + } + + // allocate the chunck buffer and create the ring + consumer->ring.chunck.size = (TMEDIA_CONSUMER(consumer)->audio.ptime * audioFormat.mSampleRate * audioFormat.mBytesPerFrame) / 1000; + consumer->ring.size = kRingPacketCount * consumer->ring.chunck.size; + if(!(consumer->ring.chunck.buffer = tsk_realloc(consumer->ring.chunck.buffer, consumer->ring.chunck.size))) { + TSK_DEBUG_ERROR("Failed to allocate new buffer"); + return -7; + } + if(!consumer->ring.buffer) { + consumer->ring.buffer = speex_buffer_init((int)consumer->ring.size); + } + else { + int ret; + if((ret = (int)speex_buffer_resize(consumer->ring.buffer, (int)consumer->ring.size)) < 0) { + TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", (int)consumer->ring.size, ret); + return ret; + } + } + if(!consumer->ring.buffer) { + TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", (int)consumer->ring.size); + return -8; + } + if(!consumer->ring.mutex && !(consumer->ring.mutex = tsk_mutex_create_2(tsk_false))) { + TSK_DEBUG_ERROR("Failed to create mutex"); + return -9; + } + + // set maximum frames per slice as buffer size + //UInt32 numFrames = (UInt32)consumer->ring.chunck.size; + //status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle), + // kAudioUnitProperty_MaximumFramesPerSlice, + // kAudioUnitScope_Global, + // 0, + // &numFrames, + // sizeof(numFrames)); + //if(status){ + // TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_MaximumFramesPerSlice, %u) failed with status=%d", (unsigned)numFrames, (int32_t)status); + // return -6; + //} + + TSK_DEBUG_INFO("AudioUnit consumer prepared"); return tdav_audiounit_handle_signal_consumer_prepared(consumer->audioUnitHandle); } static int tdav_consumer_audiounit_start(tmedia_consumer_t* self) { - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(consumer->paused){ - consumer->paused = tsk_false; - } - if(consumer->started){ - TSK_DEBUG_WARN("Already started"); - return 0; - } - else { - int ret = tdav_audiounit_handle_start(consumer->audioUnitHandle); - if(ret){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_start failed with error code=%d", ret); - return ret; - } - } - consumer->started = tsk_true; - TSK_DEBUG_INFO("AudioUnit consumer started"); - return 0; + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; + + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(consumer->paused) { + consumer->paused = tsk_false; + } + if(consumer->started) { + TSK_DEBUG_WARN("Already started"); + return 0; + } + else { + int ret = tdav_audiounit_handle_start(consumer->audioUnitHandle); + if(ret) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_start failed with error code=%d", ret); + return ret; + } + } + consumer->started = tsk_true; + TSK_DEBUG_INFO("AudioUnit consumer started"); + return 0; } static int tdav_consumer_audiounit_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) -{ - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - if(!consumer || !buffer || !size){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } +{ + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; + if(!consumer || !buffer || !size) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } #if DISABLE_JITTER_BUFFER - { - if(consumer->ring.buffer){ - tsk_mutex_lock(consumer->ring.mutex); - speex_buffer_write(consumer->ring.buffer, (void*)buffer, size); - tsk_mutex_unlock(consumer->ring.mutex); - return 0; - } - return -2; - } + { + if(consumer->ring.buffer) { + tsk_mutex_lock(consumer->ring.mutex); + speex_buffer_write(consumer->ring.buffer, (void*)buffer, size); + tsk_mutex_unlock(consumer->ring.mutex); + return 0; + } + return -2; + } #else - { - return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); - } + { + return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); + } #endif } static int tdav_consumer_audiounit_pause(tmedia_consumer_t* self) { - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - consumer->paused = tsk_true; - TSK_DEBUG_INFO("AudioUnit consumer paused"); - return 0; + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + consumer->paused = tsk_true; + TSK_DEBUG_INFO("AudioUnit consumer paused"); + return 0; } static int tdav_consumer_audiounit_stop(tmedia_consumer_t* self) { - tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; - - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(!consumer->started){ - TSK_DEBUG_INFO("Not started"); - return 0; - } - else { - int ret = tdav_audiounit_handle_stop(consumer->audioUnitHandle); - if(ret){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_stop failed with error code=%d", ret); - return ret; - } - } + tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self; + + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(!consumer->started) { + TSK_DEBUG_INFO("Not started"); + return 0; + } + else { + int ret = tdav_audiounit_handle_stop(consumer->audioUnitHandle); + if(ret) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_stop failed with error code=%d", ret); + return ret; + } + } #if TARGET_OS_IPHONE - //https://devforums.apple.com/thread/118595 - if(consumer->audioUnitHandle){ - tdav_audiounit_handle_destroy(&consumer->audioUnitHandle); - } + //https://devforums.apple.com/thread/118595 + if(consumer->audioUnitHandle) { + tdav_audiounit_handle_destroy(&consumer->audioUnitHandle); + } #endif - - consumer->started = tsk_false; - TSK_DEBUG_INFO("AudioUnit consumer stoppped"); - return 0; - + + consumer->started = tsk_false; + TSK_DEBUG_INFO("AudioUnit consumer stoppped"); + return 0; + } // @@ -379,67 +380,65 @@ static int tdav_consumer_audiounit_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_audiounit_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_audiounit_t *consumer = self; - if(consumer){ - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); - /* init self */ - } - return self; + tdav_consumer_audiounit_t *consumer = self; + if(consumer) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); + /* init self */ + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_audiounit_dtor(tsk_object_t * self) -{ - tdav_consumer_audiounit_t *consumer = self; - if(consumer){ - /* deinit self */ - // Stop the consumer if not done - if(consumer->started){ - tdav_consumer_audiounit_stop(self); - } - // destroy handle - if(consumer->audioUnitHandle){ - tdav_audiounit_handle_destroy(&consumer->audioUnitHandle); - } - TSK_FREE(consumer->ring.chunck.buffer); - if(consumer->ring.buffer){ - speex_buffer_destroy(consumer->ring.buffer); - } - if(consumer->ring.mutex){ - tsk_mutex_destroy(&consumer->ring.mutex); - } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); +{ + tdav_consumer_audiounit_t *consumer = self; + if(consumer) { + /* deinit self */ + // Stop the consumer if not done + if(consumer->started) { + tdav_consumer_audiounit_stop(self); + } + // destroy handle + if(consumer->audioUnitHandle) { + tdav_audiounit_handle_destroy(&consumer->audioUnitHandle); + } + TSK_FREE(consumer->ring.chunck.buffer); + if(consumer->ring.buffer) { + speex_buffer_destroy(consumer->ring.buffer); + } + if(consumer->ring.mutex) { + tsk_mutex_destroy(&consumer->ring.mutex); + } + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); TSK_DEBUG_INFO("*** AudioUnit Consumer destroyed ***"); - } - - return self; + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_audiounit_def_s = -{ - sizeof(tdav_consumer_audiounit_t), - tdav_consumer_audiounit_ctor, - tdav_consumer_audiounit_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_audiounit_def_s = { + sizeof(tdav_consumer_audiounit_t), + tdav_consumer_audiounit_ctor, + tdav_consumer_audiounit_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_audiounit_plugin_def_s = -{ - &tdav_consumer_audiounit_def_s, - - tmedia_audio, - "Apple CoreAudio consumer(AudioUnit)", - - tdav_consumer_audiounit_set, - tdav_consumer_audiounit_prepare, - tdav_consumer_audiounit_start, - tdav_consumer_audiounit_consume, - tdav_consumer_audiounit_pause, - tdav_consumer_audiounit_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_audiounit_plugin_def_s = { + &tdav_consumer_audiounit_def_s, + + tmedia_audio, + "Apple CoreAudio consumer(AudioUnit)", + + tdav_consumer_audiounit_set, + tdav_consumer_audiounit_prepare, + tdav_consumer_audiounit_start, + tdav_consumer_audiounit_consume, + tdav_consumer_audiounit_pause, + tdav_consumer_audiounit_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_audiounit_plugin_def_t = &tdav_consumer_audiounit_plugin_def_s; diff --git a/tinyDAV/src/audio/coreaudio/tdav_producer_audioqueue.c b/tinyDAV/src/audio/coreaudio/tdav_producer_audioqueue.c index d96fd67..b99b202 100755 --- a/tinyDAV/src/audio/coreaudio/tdav_producer_audioqueue.c +++ b/tinyDAV/src/audio/coreaudio/tdav_producer_audioqueue.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> - * + * * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. - * + * * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -23,7 +23,7 @@ /**@file tdav_producer_audioqueue.c * @brief Audio Producer for MacOSX and iOS platforms using AudioQueue. * - * @authors + * @authors * - Laurent Etiemble <laurent.etiemble(at)gmail.com> * - Mamadou Diop <diopmamadou(at)doubango(dot)org> * @@ -41,18 +41,19 @@ #include "tsk_memory.h" #include "tsk_debug.h" -static void __handle_input_buffer (void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer, const AudioTimeStamp *start_time, UInt32 number_packet_descriptions, const AudioStreamPacketDescription *packet_descriptions ) { - tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)userdata; - +static void __handle_input_buffer (void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer, const AudioTimeStamp *start_time, UInt32 number_packet_descriptions, const AudioStreamPacketDescription *packet_descriptions ) +{ + tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)userdata; + if (!producer->started) { return; } - - // Alert the session that there is new data to send - if(TMEDIA_PRODUCER(producer)->enc_cb.callback) { - TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data, buffer->mAudioData, buffer->mAudioDataByteSize); - } - + + // Alert the session that there is new data to send + if(TMEDIA_PRODUCER(producer)->enc_cb.callback) { + TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data, buffer->mAudioData, buffer->mAudioDataByteSize); + } + // Re-enqueue the buffer AudioQueueEnqueueBuffer(producer->queue, buffer, 0, NULL); } @@ -63,24 +64,24 @@ static void __handle_input_buffer (void *userdata, AudioQueueRef queue, AudioQue static int tdav_producer_audioqueue_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { OSStatus ret; - tsk_size_t i; - tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; - - if(!producer || !codec && codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); - /* codec should have ptime */ - - - // Set audio category + tsk_size_t i; + tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; + + if(!producer || !codec && codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + /* codec should have ptime */ + + + // Set audio category #if TARGET_OS_IPHONE - UInt32 category = kAudioSessionCategory_PlayAndRecord; - AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); + UInt32 category = kAudioSessionCategory_PlayAndRecord; + AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); #endif // Create the audio stream description AudioStreamBasicDescription *description = &(producer->description); @@ -93,95 +94,95 @@ static int tdav_producer_audioqueue_prepare(tmedia_producer_t* self, const tmedi description->mBytesPerPacket = description->mBitsPerChannel / 8 * description->mChannelsPerFrame; description->mBytesPerFrame = description->mBytesPerPacket; description->mReserved = 0; - + int packetperbuffer = 1000 / TMEDIA_PRODUCER(producer)->audio.ptime; producer->buffer_size = description->mSampleRate * description->mBytesPerFrame / packetperbuffer; - + // Create the record audio queue ret = AudioQueueNewInput(&(producer->description), - __handle_input_buffer, - producer, - NULL, - kCFRunLoopCommonModes, - 0, - &(producer->queue)); - + __handle_input_buffer, + producer, + NULL, + kCFRunLoopCommonModes, + 0, + &(producer->queue)); + for(i = 0; i < CoreAudioRecordBuffers; i++) { // Create the buffer for the queue ret = AudioQueueAllocateBuffer(producer->queue, producer->buffer_size, &(producer->buffers[i])); if (ret) { break; } - + // Clear the data memset(producer->buffers[i]->mAudioData, 0, producer->buffer_size); producer->buffers[i]->mAudioDataByteSize = producer->buffer_size; - + // Enqueue the buffer ret = AudioQueueEnqueueBuffer(producer->queue, producer->buffers[i], 0, NULL); if (ret) { break; } } - - return 0; + + return 0; } static int tdav_producer_audioqueue_start(tmedia_producer_t* self) { OSStatus ret; - tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; - - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(producer->started){ - TSK_DEBUG_WARN("Producer already started"); - return 0; - } - - producer->started = tsk_true; + tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; + + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(producer->started) { + TSK_DEBUG_WARN("Producer already started"); + return 0; + } + + producer->started = tsk_true; ret = AudioQueueStart(producer->queue, NULL); - - return ret; + + return ret; } static int tdav_producer_audioqueue_pause(tmedia_producer_t* self) { OSStatus ret; - tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; - - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - + tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; + + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + ret = AudioQueuePause(producer->queue); - - return ret; + + return ret; } static int tdav_producer_audioqueue_stop(tmedia_producer_t* self) { OSStatus ret; - tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; - - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(!producer->started){ - TSK_DEBUG_WARN("Producer not started"); - return 0; - } - - producer->started = tsk_false; + tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)self; + + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(!producer->started) { + TSK_DEBUG_WARN("Producer not started"); + return 0; + } + + producer->started = tsk_false; ret = AudioQueueStop(producer->queue, false); - - return ret; + + return ret; } @@ -191,62 +192,60 @@ static int tdav_producer_audioqueue_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_audioqueue_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_audioqueue_t *producer = self; - if(producer){ - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); - /* init self */ - // TODO - } - return self; + tdav_producer_audioqueue_t *producer = self; + if(producer) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); + /* init self */ + // TODO + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_audioqueue_dtor(tsk_object_t * self) -{ - tdav_producer_audioqueue_t *producer = self; - if(producer){ - // Stop the producer if not done - if(producer->started){ - tdav_producer_audioqueue_stop(self); - } - - // Free all buffers and dispose the queue +{ + tdav_producer_audioqueue_t *producer = self; + if(producer) { + // Stop the producer if not done + if(producer->started) { + tdav_producer_audioqueue_stop(self); + } + + // Free all buffers and dispose the queue if (producer->queue) { - tsk_size_t i; - - for(i=0; i<CoreAudioRecordBuffers; i++){ - AudioQueueFreeBuffer(producer->queue, producer->buffers[i]); - } + tsk_size_t i; + + for(i=0; i<CoreAudioRecordBuffers; i++) { + AudioQueueFreeBuffer(producer->queue, producer->buffers[i]); + } AudioQueueDispose(producer->queue, true); } - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); - } - - return self; + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_audioqueue_def_s = -{ - sizeof(tdav_producer_audioqueue_t), - tdav_producer_audioqueue_ctor, - tdav_producer_audioqueue_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_audioqueue_def_s = { + sizeof(tdav_producer_audioqueue_t), + tdav_producer_audioqueue_ctor, + tdav_producer_audioqueue_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_audioqueue_plugin_def_s = -{ - &tdav_producer_audioqueue_def_s, - - tmedia_audio, - "Apple CoreAudio producer (AudioQueue)", - - tdav_producer_audioqueue_set, - tdav_producer_audioqueue_prepare, - tdav_producer_audioqueue_start, - tdav_producer_audioqueue_pause, - tdav_producer_audioqueue_stop +static const tmedia_producer_plugin_def_t tdav_producer_audioqueue_plugin_def_s = { + &tdav_producer_audioqueue_def_s, + + tmedia_audio, + "Apple CoreAudio producer (AudioQueue)", + + tdav_producer_audioqueue_set, + tdav_producer_audioqueue_prepare, + tdav_producer_audioqueue_start, + tdav_producer_audioqueue_pause, + tdav_producer_audioqueue_stop }; const tmedia_producer_plugin_def_t *tdav_producer_audioqueue_plugin_def_t = &tdav_producer_audioqueue_plugin_def_s; diff --git a/tinyDAV/src/audio/coreaudio/tdav_producer_audiounit.c b/tinyDAV/src/audio/coreaudio/tdav_producer_audiounit.c index a88261e..7f8af7e 100755 --- a/tinyDAV/src/audio/coreaudio/tdav_producer_audiounit.c +++ b/tinyDAV/src/audio/coreaudio/tdav_producer_audiounit.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> - * + * * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. - * + * * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - * + * * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -35,322 +35,323 @@ #define kRingPacketCount 10 -static OSStatus __handle_input_buffer(void *inRefCon, - AudioUnitRenderActionFlags *ioActionFlags, - const AudioTimeStamp *inTimeStamp, - UInt32 inBusNumber, - UInt32 inNumberFrames, - AudioBufferList *ioData) { - OSStatus status = noErr; - tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)inRefCon; - - // holder - AudioBuffer buffer; - buffer.mData = tsk_null; - buffer.mDataByteSize = 0; - buffer.mNumberChannels = TMEDIA_PRODUCER(producer)->audio.channels; - - // list of holders - AudioBufferList buffers; - buffers.mNumberBuffers = 1; - buffers.mBuffers[0] = buffer; - - // render to get frames from the system - status = AudioUnitRender(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - ioActionFlags, - inTimeStamp, - inBusNumber, - inNumberFrames, - &buffers); - if(status == 0){ +static OSStatus __handle_input_buffer(void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + OSStatus status = noErr; + tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)inRefCon; + + // holder + AudioBuffer buffer; + buffer.mData = tsk_null; + buffer.mDataByteSize = 0; + buffer.mNumberChannels = TMEDIA_PRODUCER(producer)->audio.channels; + + // list of holders + AudioBufferList buffers; + buffers.mNumberBuffers = 1; + buffers.mBuffers[0] = buffer; + + // render to get frames from the system + status = AudioUnitRender(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + ioActionFlags, + inTimeStamp, + inBusNumber, + inNumberFrames, + &buffers); + if(status == 0) { // must not be done on async thread: doing it gives bad audio quality when audio+video call is done with CPU consuming codec (e.g. speex or g729) - speex_buffer_write(producer->ring.buffer, buffers.mBuffers[0].mData, buffers.mBuffers[0].mDataByteSize); + speex_buffer_write(producer->ring.buffer, buffers.mBuffers[0].mData, buffers.mBuffers[0].mDataByteSize); int avail = speex_buffer_get_available(producer->ring.buffer); while (producer->started && avail >= producer->ring.chunck.size) { avail -= speex_buffer_read(producer->ring.buffer, (void*)producer->ring.chunck.buffer, (int)producer->ring.chunck.size); TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data, - producer->ring.chunck.buffer, producer->ring.chunck.size); + producer->ring.chunck.buffer, producer->ring.chunck.size); } - } - + } + return status; } /* ============ Media Producer Interface ================= */ int tdav_producer_audiounit_set(tmedia_producer_t* self, const tmedia_param_t* param) -{ +{ tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; - if(param->plugin_type == tmedia_ppt_producer){ - if(param->value_type == tmedia_pvt_int32){ - if (tsk_striequals(param->key, "mute")) { - producer->muted = TSK_TO_INT32((uint8_t*)param->value); - return tdav_audiounit_handle_mute(((tdav_producer_audiounit_t*)self)->audioUnitHandle, producer->muted); - } + if(param->plugin_type == tmedia_ppt_producer) { + if(param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "mute")) { + producer->muted = TSK_TO_INT32((uint8_t*)param->value); + return tdav_audiounit_handle_mute(((tdav_producer_audiounit_t*)self)->audioUnitHandle, producer->muted); + } else if (tsk_striequals(param->key, "interrupt")) { - int32_t interrupt = *((uint8_t*)param->value) ? 1 : 0; + int32_t interrupt = *((uint8_t*)param->value) ? 1 : 0; return tdav_audiounit_handle_interrupt(producer->audioUnitHandle, interrupt); } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int tdav_producer_audiounit_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - static UInt32 flagOne = 1; - UInt32 param; - // static UInt32 flagZero = 0; + static UInt32 flagOne = 1; + UInt32 param; + // static UInt32 flagZero = 0; #define kInputBus 1 - - tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; - OSStatus status = noErr; - AudioStreamBasicDescription audioFormat; - AudioStreamBasicDescription deviceFormat; - - if(!producer || !codec || !codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(!producer->audioUnitHandle){ - if(!(producer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_PRODUCER(producer)->session_id))){ - TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_PRODUCER(producer)->session_id); - return -3; - } - } - - // enable - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioOutputUnitProperty_EnableIO, - kAudioUnitScope_Input, - kInputBus, - &flagOne, - sizeof(flagOne)); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); - return -4; - } - else { + + tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; + OSStatus status = noErr; + AudioStreamBasicDescription audioFormat; + AudioStreamBasicDescription deviceFormat; + + if(!producer || !codec || !codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(!producer->audioUnitHandle) { + if(!(producer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_PRODUCER(producer)->session_id))) { + TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_PRODUCER(producer)->session_id); + return -3; + } + } + + // enable + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Input, + kInputBus, + &flagOne, + sizeof(flagOne)); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); + return -4; + } + else { #if !TARGET_OS_IPHONE // strange: TARGET_OS_MAC is equal to '1' on Smulator - // disable output - param = 0; - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioOutputUnitProperty_EnableIO, - kAudioUnitScope_Output, - 0, - ¶m, - sizeof(UInt32)); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); - return -4; - } - - // set default audio device - param = sizeof(AudioDeviceID); - AudioDeviceID inputDeviceID; - status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, ¶m, &inputDeviceID); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice) failed with status=%ld", (signed long)status); - return -4; - } - - // set the current device to the default input unit - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioOutputUnitProperty_CurrentDevice, - kAudioUnitScope_Output, - 0, - &inputDeviceID, - sizeof(AudioDeviceID)); - if(status != noErr){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_CurrentDevice) failed with status=%ld", (signed long)status); - return -4; - } + // disable output + param = 0; + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, + 0, + ¶m, + sizeof(UInt32)); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_EnableIO) failed with status=%ld", (signed long)status); + return -4; + } + + // set default audio device + param = sizeof(AudioDeviceID); + AudioDeviceID inputDeviceID; + status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, ¶m, &inputDeviceID); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice) failed with status=%ld", (signed long)status); + return -4; + } + + // set the current device to the default input unit + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Output, + 0, + &inputDeviceID, + sizeof(AudioDeviceID)); + if(status != noErr) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_CurrentDevice) failed with status=%ld", (signed long)status); + return -4; + } #endif /* TARGET_OS_MAC */ - - /* codec should have ptime */ - TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + + /* codec should have ptime */ + TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); TSK_DEBUG_INFO("AudioUnit producer: channels=%d, rate=%d, ptime=%d", TMEDIA_PRODUCER(producer)->audio.channels, TMEDIA_PRODUCER(producer)->audio.rate, TMEDIA_PRODUCER(producer)->audio.ptime); - - // get device format - param = sizeof(AudioStreamBasicDescription); - status = AudioUnitGetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioUnitProperty_StreamFormat, - kAudioUnitScope_Input, - kInputBus, - &deviceFormat, ¶m); - if(status == noErr && deviceFormat.mSampleRate){ + + // get device format + param = sizeof(AudioStreamBasicDescription); + status = AudioUnitGetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + kInputBus, + &deviceFormat, ¶m); + if(status == noErr && deviceFormat.mSampleRate) { #if TARGET_OS_IPHONE - // iOS support 8Khz, 16kHz and 32kHz => do not override the sampleRate + // iOS support 8Khz, 16kHz and 32kHz => do not override the sampleRate #elif TARGET_OS_MAC - // For example, iSight supports only 48kHz - TMEDIA_PRODUCER(producer)->audio.rate = deviceFormat.mSampleRate; + // For example, iSight supports only 48kHz + TMEDIA_PRODUCER(producer)->audio.rate = deviceFormat.mSampleRate; #endif - } - - // set format - audioFormat.mSampleRate = TMEDIA_PRODUCER(producer)->audio.rate; - audioFormat.mFormatID = kAudioFormatLinearPCM; - audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; - audioFormat.mChannelsPerFrame = TMEDIA_PRODUCER(producer)->audio.channels; - audioFormat.mFramesPerPacket = 1; - audioFormat.mBitsPerChannel = TMEDIA_PRODUCER(producer)->audio.bits_per_sample; - audioFormat.mBytesPerPacket = audioFormat.mBitsPerChannel / 8 * audioFormat.mChannelsPerFrame; - audioFormat.mBytesPerFrame = audioFormat.mBytesPerPacket; - audioFormat.mReserved = 0; - if(audioFormat.mFormatID == kAudioFormatLinearPCM && audioFormat.mChannelsPerFrame == 1){ - audioFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; - } - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioUnitProperty_StreamFormat, - kAudioUnitScope_Output, - kInputBus, - &audioFormat, - sizeof(audioFormat)); - if(status){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed with status=%ld", (signed long)status); - return -5; - } - else { - - // configure - if(tdav_audiounit_handle_configure(producer->audioUnitHandle, tsk_false, TMEDIA_PRODUCER(producer)->audio.ptime, &audioFormat)){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_set_rate(%d) failed", TMEDIA_PRODUCER(producer)->audio.rate); - return -4; - } - - // set callback function - AURenderCallbackStruct callback; - callback.inputProc = __handle_input_buffer; - callback.inputProcRefCon = producer; - status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - kAudioOutputUnitProperty_SetInputCallback, - kAudioUnitScope_Output, - kInputBus, - &callback, - sizeof(callback)); - if(status){ - TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); - return -6; - } - else { - // disbale buffer allocation as we will provide ours - //status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), - // kAudioUnitProperty_ShouldAllocateBuffer, - // kAudioUnitScope_Output, - // kInputBus, - // &flagZero, - // sizeof(flagZero)); - - producer->ring.chunck.size = (TMEDIA_PRODUCER(producer)->audio.ptime * audioFormat.mSampleRate * audioFormat.mBytesPerFrame) / 1000; - // allocate our chunck buffer - if(!(producer->ring.chunck.buffer = tsk_realloc(producer->ring.chunck.buffer, producer->ring.chunck.size))){ - TSK_DEBUG_ERROR("Failed to allocate new buffer"); - return -7; - } - // create ringbuffer - producer->ring.size = kRingPacketCount * producer->ring.chunck.size; - if(!producer->ring.buffer){ - producer->ring.buffer = speex_buffer_init((int)producer->ring.size); - } - else { - int ret; - if((ret = speex_buffer_resize(producer->ring.buffer, producer->ring.size)) < 0){ - TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", (int)producer->ring.size, ret); - return ret; - } - } - if(!producer->ring.buffer){ - TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", (int)producer->ring.size); - return -9; - } - } - - } - } - - TSK_DEBUG_INFO("AudioUnit producer prepared"); - return tdav_audiounit_handle_signal_producer_prepared(producer->audioUnitHandle);; + } + + // set format + audioFormat.mSampleRate = TMEDIA_PRODUCER(producer)->audio.rate; + audioFormat.mFormatID = kAudioFormatLinearPCM; + audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; + audioFormat.mChannelsPerFrame = TMEDIA_PRODUCER(producer)->audio.channels; + audioFormat.mFramesPerPacket = 1; + audioFormat.mBitsPerChannel = TMEDIA_PRODUCER(producer)->audio.bits_per_sample; + audioFormat.mBytesPerPacket = audioFormat.mBitsPerChannel / 8 * audioFormat.mChannelsPerFrame; + audioFormat.mBytesPerFrame = audioFormat.mBytesPerPacket; + audioFormat.mReserved = 0; + if(audioFormat.mFormatID == kAudioFormatLinearPCM && audioFormat.mChannelsPerFrame == 1) { + audioFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; + } + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + kInputBus, + &audioFormat, + sizeof(audioFormat)); + if(status) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed with status=%ld", (signed long)status); + return -5; + } + else { + + // configure + if(tdav_audiounit_handle_configure(producer->audioUnitHandle, tsk_false, TMEDIA_PRODUCER(producer)->audio.ptime, &audioFormat)) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_set_rate(%d) failed", TMEDIA_PRODUCER(producer)->audio.rate); + return -4; + } + + // set callback function + AURenderCallbackStruct callback; + callback.inputProc = __handle_input_buffer; + callback.inputProcRefCon = producer; + status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + kAudioOutputUnitProperty_SetInputCallback, + kAudioUnitScope_Output, + kInputBus, + &callback, + sizeof(callback)); + if(status) { + TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%ld", (signed long)status); + return -6; + } + else { + // disbale buffer allocation as we will provide ours + //status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle), + // kAudioUnitProperty_ShouldAllocateBuffer, + // kAudioUnitScope_Output, + // kInputBus, + // &flagZero, + // sizeof(flagZero)); + + producer->ring.chunck.size = (TMEDIA_PRODUCER(producer)->audio.ptime * audioFormat.mSampleRate * audioFormat.mBytesPerFrame) / 1000; + // allocate our chunck buffer + if(!(producer->ring.chunck.buffer = tsk_realloc(producer->ring.chunck.buffer, producer->ring.chunck.size))) { + TSK_DEBUG_ERROR("Failed to allocate new buffer"); + return -7; + } + // create ringbuffer + producer->ring.size = kRingPacketCount * producer->ring.chunck.size; + if(!producer->ring.buffer) { + producer->ring.buffer = speex_buffer_init((int)producer->ring.size); + } + else { + int ret; + if((ret = speex_buffer_resize(producer->ring.buffer, producer->ring.size)) < 0) { + TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", (int)producer->ring.size, ret); + return ret; + } + } + if(!producer->ring.buffer) { + TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", (int)producer->ring.size); + return -9; + } + } + + } + } + + TSK_DEBUG_INFO("AudioUnit producer prepared"); + return tdav_audiounit_handle_signal_producer_prepared(producer->audioUnitHandle);; } static int tdav_producer_audiounit_start(tmedia_producer_t* self) { - tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; - - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(producer->paused){ - producer->paused = tsk_false; - return tsk_false; - } - - int ret; - if(producer->started){ - TSK_DEBUG_WARN("Already started"); - return 0; - } - else { - ret = tdav_audiounit_handle_start(producer->audioUnitHandle); - if(ret){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_start failed with error code=%d", ret); - return ret; - } - } - producer->started = tsk_true; - + tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; + + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(producer->paused) { + producer->paused = tsk_false; + return tsk_false; + } + + int ret; + if(producer->started) { + TSK_DEBUG_WARN("Already started"); + return 0; + } + else { + ret = tdav_audiounit_handle_start(producer->audioUnitHandle); + if(ret) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_start failed with error code=%d", ret); + return ret; + } + } + producer->started = tsk_true; + // apply parameters (because could be lost when the producer is restarted -handle recreated-) ret = tdav_audiounit_handle_mute(producer->audioUnitHandle, producer->muted); - TSK_DEBUG_INFO("AudioUnit producer started"); - return 0; + TSK_DEBUG_INFO("AudioUnit producer started"); + return 0; } static int tdav_producer_audiounit_pause(tmedia_producer_t* self) { tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - producer->paused = tsk_true; - TSK_DEBUG_INFO("AudioUnit producer paused"); - return 0; + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + producer->paused = tsk_true; + TSK_DEBUG_INFO("AudioUnit producer paused"); + return 0; } static int tdav_producer_audiounit_stop(tmedia_producer_t* self) { tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self; - - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if(!producer->started){ - TSK_DEBUG_INFO("Not started"); - return 0; - } - else { - int ret = tdav_audiounit_handle_stop(producer->audioUnitHandle); - if(ret){ - TSK_DEBUG_ERROR("tdav_audiounit_handle_stop failed with error code=%d", ret); - // do not return even if failed => we MUST stop the thread! - } + + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if(!producer->started) { + TSK_DEBUG_INFO("Not started"); + return 0; + } + else { + int ret = tdav_audiounit_handle_stop(producer->audioUnitHandle); + if(ret) { + TSK_DEBUG_ERROR("tdav_audiounit_handle_stop failed with error code=%d", ret); + // do not return even if failed => we MUST stop the thread! + } #if TARGET_OS_IPHONE - //https://devforums.apple.com/thread/118595 - if(producer->audioUnitHandle){ - tdav_audiounit_handle_destroy(&producer->audioUnitHandle); - } + //https://devforums.apple.com/thread/118595 + if(producer->audioUnitHandle) { + tdav_audiounit_handle_destroy(&producer->audioUnitHandle); + } #endif - } - producer->started = tsk_false; - TSK_DEBUG_INFO("AudioUnit producer stoppped"); - return 0; + } + producer->started = tsk_false; + TSK_DEBUG_INFO("AudioUnit producer stoppped"); + return 0; } @@ -360,61 +361,59 @@ static int tdav_producer_audiounit_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_audiounit_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_audiounit_t *producer = self; - if(producer){ - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); - /* init self */ - } - return self; + tdav_producer_audiounit_t *producer = self; + if(producer) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); + /* init self */ + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_audiounit_dtor(tsk_object_t * self) -{ - tdav_producer_audiounit_t *producer = self; - if(producer){ - // Stop the producer if not done - if(producer->started){ - tdav_producer_audiounit_stop(self); - } - - // Free all buffers and dispose the queue +{ + tdav_producer_audiounit_t *producer = self; + if(producer) { + // Stop the producer if not done + if(producer->started) { + tdav_producer_audiounit_stop(self); + } + + // Free all buffers and dispose the queue if (producer->audioUnitHandle) { - tdav_audiounit_handle_destroy(&producer->audioUnitHandle); + tdav_audiounit_handle_destroy(&producer->audioUnitHandle); } TSK_FREE(producer->ring.chunck.buffer); - if(producer->ring.buffer){ - speex_buffer_destroy(producer->ring.buffer); - } - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); - + if(producer->ring.buffer) { + speex_buffer_destroy(producer->ring.buffer); + } + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); + TSK_DEBUG_INFO("*** AudioUnit Producer destroyed ***"); - } - - return self; + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_audiounit_def_s = -{ - sizeof(tdav_producer_audiounit_t), - tdav_producer_audiounit_ctor, - tdav_producer_audiounit_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_audiounit_def_s = { + sizeof(tdav_producer_audiounit_t), + tdav_producer_audiounit_ctor, + tdav_producer_audiounit_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_audiounit_plugin_def_s = -{ - &tdav_producer_audiounit_def_s, - - tmedia_audio, - "Apple CoreAudio producer (AudioUnit)", - - tdav_producer_audiounit_set, - tdav_producer_audiounit_prepare, - tdav_producer_audiounit_start, - tdav_producer_audiounit_pause, - tdav_producer_audiounit_stop +static const tmedia_producer_plugin_def_t tdav_producer_audiounit_plugin_def_s = { + &tdav_producer_audiounit_def_s, + + tmedia_audio, + "Apple CoreAudio producer (AudioUnit)", + + tdav_producer_audiounit_set, + tdav_producer_audiounit_prepare, + tdav_producer_audiounit_start, + tdav_producer_audiounit_pause, + tdav_producer_audiounit_stop }; const tmedia_producer_plugin_def_t *tdav_producer_audiounit_plugin_def_t = &tdav_producer_audiounit_plugin_def_s; diff --git a/tinyDAV/src/audio/directsound/tdav_consumer_dsound.c b/tinyDAV/src/audio/directsound/tdav_consumer_dsound.c index 82e125b..cdf87f5 100755 --- a/tinyDAV/src/audio/directsound/tdav_consumer_dsound.c +++ b/tinyDAV/src/audio/directsound/tdav_consumer_dsound.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -48,117 +48,116 @@ extern void tdav_win32_print_error(const char* func, HRESULT hr); # define TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT 20 #endif /* TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT */ -typedef struct tdav_consumer_dsound_s -{ - TDAV_DECLARE_CONSUMER_AUDIO; +typedef struct tdav_consumer_dsound_s { + TDAV_DECLARE_CONSUMER_AUDIO; - tsk_bool_t started; - tsk_size_t bytes_per_notif_size; - uint8_t* bytes_per_notif_ptr; - tsk_thread_handle_t* tid[1]; + tsk_bool_t started; + tsk_size_t bytes_per_notif_size; + uint8_t* bytes_per_notif_ptr; + tsk_thread_handle_t* tid[1]; - LPDIRECTSOUND device; - LPDIRECTSOUNDBUFFER primaryBuffer; - LPDIRECTSOUNDBUFFER secondaryBuffer; - HANDLE notifEvents[TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT]; + LPDIRECTSOUND device; + LPDIRECTSOUNDBUFFER primaryBuffer; + LPDIRECTSOUNDBUFFER secondaryBuffer; + HANDLE notifEvents[TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT]; } tdav_consumer_dsound_t; static _inline int32_t __convert_volume(int32_t volume) { - static const int32_t __step = (DSBVOLUME_MAX - DSBVOLUME_MIN) / 100; - return (volume * __step) + DSBVOLUME_MIN; + static const int32_t __step = (DSBVOLUME_MAX - DSBVOLUME_MIN) / 100; + return (volume * __step) + DSBVOLUME_MIN; } static void* TSK_STDCALL _tdav_consumer_dsound_playback_thread(void *param) { - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)param; - - HRESULT hr; - LPVOID lpvAudio1, lpvAudio2; - DWORD dwBytesAudio1, dwBytesAudio2, dwEvent; - static const DWORD dwWriteCursor = 0; - tsk_size_t out_size; - - TSK_DEBUG_INFO("_tdav_consumer_dsound_playback_thread -- START"); - - SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); - - while (dsound->started) { - dwEvent = WaitForMultipleObjects(TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT, dsound->notifEvents, FALSE, INFINITE); - if (!dsound->started) { - break; - } - - // lock - hr = IDirectSoundBuffer_Lock( - dsound->secondaryBuffer, - dwWriteCursor/* Ignored because of DSBLOCK_FROMWRITECURSOR */, - (DWORD)dsound->bytes_per_notif_size, - &lpvAudio1, &dwBytesAudio1, - &lpvAudio2, &dwBytesAudio2, - DSBLOCK_FROMWRITECURSOR); - if (hr != DS_OK) { - tdav_win32_print_error("IDirectSoundBuffer_Lock", hr); - goto next; - } - - out_size = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(dsound), dsound->bytes_per_notif_ptr, dsound->bytes_per_notif_size); - if (out_size < dsound->bytes_per_notif_size) { - // fill with silence - memset(&dsound->bytes_per_notif_ptr[out_size], 0, (dsound->bytes_per_notif_size - out_size)); - } - if ((dwBytesAudio1 + dwBytesAudio2) == dsound->bytes_per_notif_size) { - memcpy(lpvAudio1, dsound->bytes_per_notif_ptr, dwBytesAudio1); - if (lpvAudio2 && dwBytesAudio2) { - memcpy(lpvAudio2, &dsound->bytes_per_notif_ptr[dwBytesAudio1], dwBytesAudio2); - } - } - else { - TSK_DEBUG_ERROR("Not expected: %d+%d#%d", dwBytesAudio1, dwBytesAudio2, dsound->bytes_per_notif_size); - } + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)param; + + HRESULT hr; + LPVOID lpvAudio1, lpvAudio2; + DWORD dwBytesAudio1, dwBytesAudio2, dwEvent; + static const DWORD dwWriteCursor = 0; + tsk_size_t out_size; + + TSK_DEBUG_INFO("_tdav_consumer_dsound_playback_thread -- START"); + + SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); + + while (dsound->started) { + dwEvent = WaitForMultipleObjects(TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT, dsound->notifEvents, FALSE, INFINITE); + if (!dsound->started) { + break; + } + + // lock + hr = IDirectSoundBuffer_Lock( + dsound->secondaryBuffer, + dwWriteCursor/* Ignored because of DSBLOCK_FROMWRITECURSOR */, + (DWORD)dsound->bytes_per_notif_size, + &lpvAudio1, &dwBytesAudio1, + &lpvAudio2, &dwBytesAudio2, + DSBLOCK_FROMWRITECURSOR); + if (hr != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_Lock", hr); + goto next; + } + + out_size = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(dsound), dsound->bytes_per_notif_ptr, dsound->bytes_per_notif_size); + if (out_size < dsound->bytes_per_notif_size) { + // fill with silence + memset(&dsound->bytes_per_notif_ptr[out_size], 0, (dsound->bytes_per_notif_size - out_size)); + } + if ((dwBytesAudio1 + dwBytesAudio2) == dsound->bytes_per_notif_size) { + memcpy(lpvAudio1, dsound->bytes_per_notif_ptr, dwBytesAudio1); + if (lpvAudio2 && dwBytesAudio2) { + memcpy(lpvAudio2, &dsound->bytes_per_notif_ptr[dwBytesAudio1], dwBytesAudio2); + } + } + else { + TSK_DEBUG_ERROR("Not expected: %d+%d#%d", dwBytesAudio1, dwBytesAudio2, dsound->bytes_per_notif_size); + } #if 0 - memset(lpvAudio1, rand(), dwBytesAudio1); + memset(lpvAudio1, rand(), dwBytesAudio1); #endif - // unlock - if ((hr = IDirectSoundBuffer_Unlock(dsound->secondaryBuffer, lpvAudio1, dwBytesAudio1, lpvAudio2, dwBytesAudio2)) != DS_OK) { - tdav_win32_print_error("IDirectSoundBuffer_UnLock", hr); - goto next; - } + // unlock + if ((hr = IDirectSoundBuffer_Unlock(dsound->secondaryBuffer, lpvAudio1, dwBytesAudio1, lpvAudio2, dwBytesAudio2)) != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_UnLock", hr); + goto next; + } next: - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(dsound)); - } + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(dsound)); + } - TSK_DEBUG_INFO("_tdav_consumer_dsound_playback_thread -- STOP"); - + TSK_DEBUG_INFO("_tdav_consumer_dsound_playback_thread -- STOP"); - return tsk_null; + + return tsk_null; } static int _tdav_consumer_dsound_unprepare(tdav_consumer_dsound_t *dsound) { - if(dsound){ - tsk_size_t i; - if(dsound->primaryBuffer){ - IDirectSoundBuffer_Release(dsound->primaryBuffer); - dsound->primaryBuffer = NULL; - } - if(dsound->secondaryBuffer){ - IDirectSoundBuffer_Release(dsound->secondaryBuffer); - dsound->secondaryBuffer = NULL; - } - if(dsound->device){ - IDirectSound_Release(dsound->device); - dsound->device = NULL; - } - for(i = 0; i<sizeof(dsound->notifEvents)/sizeof(dsound->notifEvents[0]); i++){ - if(dsound->notifEvents[i]){ - CloseHandle(dsound->notifEvents[i]); - dsound->notifEvents[i] = NULL; - } - } - } - return 0; + if(dsound) { + tsk_size_t i; + if(dsound->primaryBuffer) { + IDirectSoundBuffer_Release(dsound->primaryBuffer); + dsound->primaryBuffer = NULL; + } + if(dsound->secondaryBuffer) { + IDirectSoundBuffer_Release(dsound->secondaryBuffer); + dsound->secondaryBuffer = NULL; + } + if(dsound->device) { + IDirectSound_Release(dsound->device); + dsound->device = NULL; + } + for(i = 0; i<sizeof(dsound->notifEvents)/sizeof(dsound->notifEvents[0]); i++) { + if(dsound->notifEvents[i]) { + CloseHandle(dsound->notifEvents[i]); + dsound->notifEvents[i] = NULL; + } + } + } + return 0; } @@ -166,232 +165,232 @@ static int _tdav_consumer_dsound_unprepare(tdav_consumer_dsound_t *dsound) /* ============ Media Consumer Interface ================= */ static int tdav_consumer_dsound_set(tmedia_consumer_t* self, const tmedia_param_t* param) { - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; - int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); - - if(ret == 0){ - if(dsound->secondaryBuffer && tsk_striequals(param->key, "volume")){ - if(IDirectSoundBuffer_SetVolume(dsound->secondaryBuffer, __convert_volume(TMEDIA_CONSUMER(self)->audio.volume)) != DS_OK){ - TSK_DEBUG_ERROR("IDirectSoundBuffer_SetVolume() failed"); - ret = -1; - } - } - } - - return ret; + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; + int ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); + + if(ret == 0) { + if(dsound->secondaryBuffer && tsk_striequals(param->key, "volume")) { + if(IDirectSoundBuffer_SetVolume(dsound->secondaryBuffer, __convert_volume(TMEDIA_CONSUMER(self)->audio.volume)) != DS_OK) { + TSK_DEBUG_ERROR("IDirectSoundBuffer_SetVolume() failed"); + ret = -1; + } + } + } + + return ret; } static int tdav_consumer_dsound_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - HRESULT hr; - HWND hWnd; + HRESULT hr; + HWND hWnd; - WAVEFORMATEX wfx = {0}; - DSBUFFERDESC dsbd = {0}; + WAVEFORMATEX wfx = {0}; + DSBUFFERDESC dsbd = {0}; - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; - if(!dsound){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!dsound) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if(dsound->device || dsound->primaryBuffer || dsound->secondaryBuffer){ - TSK_DEBUG_ERROR("Consumer already prepared"); - return -2; - } + if(dsound->device || dsound->primaryBuffer || dsound->secondaryBuffer) { + TSK_DEBUG_ERROR("Consumer already prepared"); + return -2; + } - TMEDIA_CONSUMER(dsound)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(dsound)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(dsound)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); + TMEDIA_CONSUMER(dsound)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(dsound)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(dsound)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); #if 0 - TMEDIA_CONSUMER(dsound)->audio.out.rate = 48000; - TMEDIA_CONSUMER(dsound)->audio.out.channels = 2; + TMEDIA_CONSUMER(dsound)->audio.out.rate = 48000; + TMEDIA_CONSUMER(dsound)->audio.out.channels = 2; #endif - /* Create sound device */ - if((hr = DirectSoundCreate(NULL, &dsound->device, NULL) != DS_OK)){ - tdav_win32_print_error("DirectSoundCreate", hr); - return -3; - } - - /* Set CooperativeLevel */ - if((hWnd = GetForegroundWindow()) || (hWnd = GetDesktopWindow()) || (hWnd = GetConsoleWindow())){ - if((hr = IDirectSound_SetCooperativeLevel(dsound->device, hWnd, DSSCL_PRIORITY)) != DS_OK){ - tdav_win32_print_error("IDirectSound_SetCooperativeLevel", hr); - return -2; - } - } - - /* Creates the primary buffer and apply format */ - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = TMEDIA_CONSUMER(dsound)->audio.out.channels ? TMEDIA_CONSUMER(dsound)->audio.out.channels : TMEDIA_CONSUMER(dsound)->audio.in.channels; - wfx.nSamplesPerSec = TMEDIA_CONSUMER(dsound)->audio.out.rate ? TMEDIA_CONSUMER(dsound)->audio.out.rate : TMEDIA_CONSUMER(dsound)->audio.in.rate; - wfx.wBitsPerSample = TMEDIA_CONSUMER(dsound)->audio.bits_per_sample; - wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); - wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); - - /* Average bytes (count) for each notification */ - dsound->bytes_per_notif_size = ((wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(dsound)->audio.ptime)/1000); - if(!(dsound->bytes_per_notif_ptr = tsk_realloc(dsound->bytes_per_notif_ptr, dsound->bytes_per_notif_size))){ - TSK_DEBUG_ERROR("Failed to allocate buffer with size = %u", dsound->bytes_per_notif_size); - return -3; - } - - dsbd.dwSize = sizeof(DSBUFFERDESC); - dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER; - dsbd.dwBufferBytes = 0; - dsbd.lpwfxFormat = NULL; - - if((hr = IDirectSound_CreateSoundBuffer(dsound->device, &dsbd, &dsound->primaryBuffer, NULL)) != DS_OK){ - tdav_win32_print_error("IDirectSound_CreateSoundBuffer", hr); - return -4; - } - if((hr = IDirectSoundBuffer_SetFormat(dsound->primaryBuffer, &wfx)) != DS_OK){ - tdav_win32_print_error("IDirectSoundBuffer_SetFormat", hr); - return -5; - } - - /* Creates the secondary buffer and apply format */ - dsbd.dwFlags = (DSBCAPS_CTRLPOSITIONNOTIFY | DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME); - dsbd.dwBufferBytes = (DWORD)(TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT * dsound->bytes_per_notif_size); - dsbd.lpwfxFormat = &wfx; - - if((hr = IDirectSound_CreateSoundBuffer(dsound->device, &dsbd, &dsound->secondaryBuffer, NULL)) != DS_OK){ - tdav_win32_print_error("IDirectSound_CreateSoundBuffer", hr); - return -6; - } - - /* Set Volume */ - if(IDirectSoundBuffer_SetVolume(dsound->secondaryBuffer, __convert_volume(TMEDIA_CONSUMER(self)->audio.volume)) != DS_OK){ - TSK_DEBUG_ERROR("IDirectSoundBuffer_SetVolume() failed"); - } - - return 0; + /* Create sound device */ + if((hr = DirectSoundCreate(NULL, &dsound->device, NULL) != DS_OK)) { + tdav_win32_print_error("DirectSoundCreate", hr); + return -3; + } + + /* Set CooperativeLevel */ + if((hWnd = GetForegroundWindow()) || (hWnd = GetDesktopWindow()) || (hWnd = GetConsoleWindow())) { + if((hr = IDirectSound_SetCooperativeLevel(dsound->device, hWnd, DSSCL_PRIORITY)) != DS_OK) { + tdav_win32_print_error("IDirectSound_SetCooperativeLevel", hr); + return -2; + } + } + + /* Creates the primary buffer and apply format */ + wfx.wFormatTag = WAVE_FORMAT_PCM; + wfx.nChannels = TMEDIA_CONSUMER(dsound)->audio.out.channels ? TMEDIA_CONSUMER(dsound)->audio.out.channels : TMEDIA_CONSUMER(dsound)->audio.in.channels; + wfx.nSamplesPerSec = TMEDIA_CONSUMER(dsound)->audio.out.rate ? TMEDIA_CONSUMER(dsound)->audio.out.rate : TMEDIA_CONSUMER(dsound)->audio.in.rate; + wfx.wBitsPerSample = TMEDIA_CONSUMER(dsound)->audio.bits_per_sample; + wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); + wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); + + /* Average bytes (count) for each notification */ + dsound->bytes_per_notif_size = ((wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(dsound)->audio.ptime)/1000); + if(!(dsound->bytes_per_notif_ptr = tsk_realloc(dsound->bytes_per_notif_ptr, dsound->bytes_per_notif_size))) { + TSK_DEBUG_ERROR("Failed to allocate buffer with size = %u", dsound->bytes_per_notif_size); + return -3; + } + + dsbd.dwSize = sizeof(DSBUFFERDESC); + dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER; + dsbd.dwBufferBytes = 0; + dsbd.lpwfxFormat = NULL; + + if((hr = IDirectSound_CreateSoundBuffer(dsound->device, &dsbd, &dsound->primaryBuffer, NULL)) != DS_OK) { + tdav_win32_print_error("IDirectSound_CreateSoundBuffer", hr); + return -4; + } + if((hr = IDirectSoundBuffer_SetFormat(dsound->primaryBuffer, &wfx)) != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_SetFormat", hr); + return -5; + } + + /* Creates the secondary buffer and apply format */ + dsbd.dwFlags = (DSBCAPS_CTRLPOSITIONNOTIFY | DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME); + dsbd.dwBufferBytes = (DWORD)(TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT * dsound->bytes_per_notif_size); + dsbd.lpwfxFormat = &wfx; + + if((hr = IDirectSound_CreateSoundBuffer(dsound->device, &dsbd, &dsound->secondaryBuffer, NULL)) != DS_OK) { + tdav_win32_print_error("IDirectSound_CreateSoundBuffer", hr); + return -6; + } + + /* Set Volume */ + if(IDirectSoundBuffer_SetVolume(dsound->secondaryBuffer, __convert_volume(TMEDIA_CONSUMER(self)->audio.volume)) != DS_OK) { + TSK_DEBUG_ERROR("IDirectSoundBuffer_SetVolume() failed"); + } + + return 0; } static int tdav_consumer_dsound_start(tmedia_consumer_t* self) { - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; - - tsk_size_t i; - HRESULT hr; - LPDIRECTSOUNDNOTIFY lpDSBNotify; - DSBPOSITIONNOTIFY pPosNotify[TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT] = {0}; - - static DWORD dwMajorVersion = -1; - - // Get OS version - if(dwMajorVersion == -1){ - OSVERSIONINFO osvi; - ZeroMemory(&osvi, sizeof(OSVERSIONINFO)); - osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFO); - GetVersionEx(&osvi); - dwMajorVersion = osvi.dwMajorVersion; - } - - if(!dsound){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(!dsound->device || !dsound->primaryBuffer || !dsound->secondaryBuffer){ - TSK_DEBUG_ERROR("Consumer not prepared"); - return -2; - } - - if(dsound->started){ - return 0; - } - - if((hr = IDirectSoundBuffer_QueryInterface(dsound->secondaryBuffer, &IID_IDirectSoundNotify, (LPVOID*)&lpDSBNotify)) != DS_OK){ - tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); - return -3; - } - - /* Events associated to notification points */ - for(i = 0; i<TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT; i++){ - dsound->notifEvents[i] = CreateEvent(NULL, FALSE, FALSE, NULL); - // set notification point offset at the start of the buffer for Windows Vista and later and at the half of the buffer of XP and before - pPosNotify[i].dwOffset = (DWORD)((dsound->bytes_per_notif_size * i) + (dwMajorVersion > 5 ? (dsound->bytes_per_notif_size >> 1) : 1)); - pPosNotify[i].hEventNotify = dsound->notifEvents[i]; - } - if((hr = IDirectSoundNotify_SetNotificationPositions(lpDSBNotify, TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT, pPosNotify)) != DS_OK){ - IDirectSoundNotify_Release(lpDSBNotify); - tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); - return -4; - } - - if((hr = IDirectSoundNotify_Release(lpDSBNotify))){ - tdav_win32_print_error("IDirectSoundNotify_Release", hr); - } - - /* Start the buffer */ - if((hr = IDirectSoundBuffer_Play(dsound->secondaryBuffer, 0, 0, DSBPLAY_LOOPING)) != DS_OK){ - tdav_win32_print_error("IDirectSoundNotify_Release", hr); - return -5; - } - - /* start the reader thread */ - dsound->started = tsk_true; - tsk_thread_create(&dsound->tid[0], _tdav_consumer_dsound_playback_thread, dsound); - - return 0; + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; + + tsk_size_t i; + HRESULT hr; + LPDIRECTSOUNDNOTIFY lpDSBNotify; + DSBPOSITIONNOTIFY pPosNotify[TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT] = {0}; + + static DWORD dwMajorVersion = -1; + + // Get OS version + if(dwMajorVersion == -1) { + OSVERSIONINFO osvi; + ZeroMemory(&osvi, sizeof(OSVERSIONINFO)); + osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFO); + GetVersionEx(&osvi); + dwMajorVersion = osvi.dwMajorVersion; + } + + if(!dsound) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(!dsound->device || !dsound->primaryBuffer || !dsound->secondaryBuffer) { + TSK_DEBUG_ERROR("Consumer not prepared"); + return -2; + } + + if(dsound->started) { + return 0; + } + + if((hr = IDirectSoundBuffer_QueryInterface(dsound->secondaryBuffer, &IID_IDirectSoundNotify, (LPVOID*)&lpDSBNotify)) != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); + return -3; + } + + /* Events associated to notification points */ + for(i = 0; i<TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT; i++) { + dsound->notifEvents[i] = CreateEvent(NULL, FALSE, FALSE, NULL); + // set notification point offset at the start of the buffer for Windows Vista and later and at the half of the buffer of XP and before + pPosNotify[i].dwOffset = (DWORD)((dsound->bytes_per_notif_size * i) + (dwMajorVersion > 5 ? (dsound->bytes_per_notif_size >> 1) : 1)); + pPosNotify[i].hEventNotify = dsound->notifEvents[i]; + } + if((hr = IDirectSoundNotify_SetNotificationPositions(lpDSBNotify, TDAV_DSOUND_CONSUMER_NOTIF_POS_COUNT, pPosNotify)) != DS_OK) { + IDirectSoundNotify_Release(lpDSBNotify); + tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); + return -4; + } + + if((hr = IDirectSoundNotify_Release(lpDSBNotify))) { + tdav_win32_print_error("IDirectSoundNotify_Release", hr); + } + + /* Start the buffer */ + if((hr = IDirectSoundBuffer_Play(dsound->secondaryBuffer, 0, 0, DSBPLAY_LOOPING)) != DS_OK) { + tdav_win32_print_error("IDirectSoundNotify_Release", hr); + return -5; + } + + /* start the reader thread */ + dsound->started = tsk_true; + tsk_thread_create(&dsound->tid[0], _tdav_consumer_dsound_playback_thread, dsound); + + return 0; } static int tdav_consumer_dsound_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; - - if(!dsound || !buffer || !size){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - /* buffer is already decoded */ - return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(dsound), buffer, size, proto_hdr); + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; + + if(!dsound || !buffer || !size) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + /* buffer is already decoded */ + return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(dsound), buffer, size, proto_hdr); } static int tdav_consumer_dsound_pause(tmedia_consumer_t* self) { - return 0; + return 0; } static int tdav_consumer_dsound_stop(tmedia_consumer_t* self) { - tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; - - HRESULT hr; - - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(!dsound->started){ - return 0; - } - - /* should be done here */ - dsound->started = tsk_false; - - /* stop thread */ - if(dsound->tid[0]){ - tsk_thread_join(&(dsound->tid[0])); - } - - if((hr = IDirectSoundBuffer_Stop(dsound->secondaryBuffer)) != DS_OK){ - tdav_win32_print_error("IDirectSoundBuffer_Stop", hr); - } - if((hr = IDirectSoundBuffer_SetCurrentPosition(dsound->secondaryBuffer, 0)) != DS_OK){ - tdav_win32_print_error("IDirectSoundBuffer_SetCurrentPosition", hr); - } - - // unprepare - // will be prepared again before calling next start() - _tdav_consumer_dsound_unprepare(dsound); - - return 0; + tdav_consumer_dsound_t* dsound = (tdav_consumer_dsound_t*)self; + + HRESULT hr; + + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(!dsound->started) { + return 0; + } + + /* should be done here */ + dsound->started = tsk_false; + + /* stop thread */ + if(dsound->tid[0]) { + tsk_thread_join(&(dsound->tid[0])); + } + + if((hr = IDirectSoundBuffer_Stop(dsound->secondaryBuffer)) != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_Stop", hr); + } + if((hr = IDirectSoundBuffer_SetCurrentPosition(dsound->secondaryBuffer, 0)) != DS_OK) { + tdav_win32_print_error("IDirectSoundBuffer_SetCurrentPosition", hr); + } + + // unprepare + // will be prepared again before calling next start() + _tdav_consumer_dsound_unprepare(dsound); + + return 0; } @@ -401,56 +400,54 @@ static int tdav_consumer_dsound_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_dsound_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_dsound_t *consumer = self; - if(consumer){ - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); - /* init self */ - - } - return self; + tdav_consumer_dsound_t *consumer = self; + if(consumer) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); + /* init self */ + + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_dsound_dtor(tsk_object_t * self) -{ - tdav_consumer_dsound_t *dsound = self; - if(dsound){ - /* stop */ - if(dsound->started){ - tdav_consumer_dsound_stop(self); - } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(dsound)); - /* deinit self */ - _tdav_consumer_dsound_unprepare(dsound); - TSK_FREE(dsound->bytes_per_notif_ptr); - } - - return self; +{ + tdav_consumer_dsound_t *dsound = self; + if(dsound) { + /* stop */ + if(dsound->started) { + tdav_consumer_dsound_stop(self); + } + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(dsound)); + /* deinit self */ + _tdav_consumer_dsound_unprepare(dsound); + TSK_FREE(dsound->bytes_per_notif_ptr); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_dsound_def_s = -{ - sizeof(tdav_consumer_dsound_t), - tdav_consumer_dsound_ctor, - tdav_consumer_dsound_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_dsound_def_s = { + sizeof(tdav_consumer_dsound_t), + tdav_consumer_dsound_ctor, + tdav_consumer_dsound_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_dsound_plugin_def_s = -{ - &tdav_consumer_dsound_def_s, - - tmedia_audio, - "Microsoft DirectSound consumer", - - tdav_consumer_dsound_set, - tdav_consumer_dsound_prepare, - tdav_consumer_dsound_start, - tdav_consumer_dsound_consume, - tdav_consumer_dsound_pause, - tdav_consumer_dsound_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_dsound_plugin_def_s = { + &tdav_consumer_dsound_def_s, + + tmedia_audio, + "Microsoft DirectSound consumer", + + tdav_consumer_dsound_set, + tdav_consumer_dsound_prepare, + tdav_consumer_dsound_start, + tdav_consumer_dsound_consume, + tdav_consumer_dsound_pause, + tdav_consumer_dsound_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_dsound_plugin_def_t = &tdav_consumer_dsound_plugin_def_s; diff --git a/tinyDAV/src/audio/directsound/tdav_producer_dsound.c b/tinyDAV/src/audio/directsound/tdav_producer_dsound.c index c5ae167..ab9ca6f 100755 --- a/tinyDAV/src/audio/directsound/tdav_producer_dsound.c +++ b/tinyDAV/src/audio/directsound/tdav_producer_dsound.c @@ -53,98 +53,97 @@ extern void tdav_win32_print_error(const char* func, HRESULT hr); # define TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT 10 #endif /* TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT */ -typedef struct tdav_producer_dsound_s -{ - TDAV_DECLARE_PRODUCER_AUDIO; +typedef struct tdav_producer_dsound_s { + TDAV_DECLARE_PRODUCER_AUDIO; - tsk_bool_t started; - tsk_bool_t mute; - tsk_size_t bytes_per_notif_size; - tsk_thread_handle_t* tid[1]; + tsk_bool_t started; + tsk_bool_t mute; + tsk_size_t bytes_per_notif_size; + tsk_thread_handle_t* tid[1]; - LPDIRECTSOUNDCAPTURE device; - LPDIRECTSOUNDCAPTUREBUFFER captureBuffer; - HANDLE notifEvents[TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT]; + LPDIRECTSOUNDCAPTURE device; + LPDIRECTSOUNDCAPTUREBUFFER captureBuffer; + HANDLE notifEvents[TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT]; } tdav_producer_dsound_t; static void* TSK_STDCALL _tdav_producer_dsound_record_thread(void *param) { - tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)param; - - HRESULT hr; - LPVOID lpvAudio1, lpvAudio2; - DWORD dwBytesAudio1, dwBytesAudio2, dwEvent, dwIndex; - - TSK_DEBUG_INFO("_tdav_producer_dsound_record_thread -- START"); - - SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_TIME_CRITICAL); - - while (dsound->started) { - dwEvent = WaitForMultipleObjects(TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT, dsound->notifEvents, FALSE, INFINITE); - if (!dsound->started) { - break; - } - if (dwEvent < WAIT_OBJECT_0 || dwEvent >(WAIT_OBJECT_0 + TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT)) { - TSK_DEBUG_ERROR("Invalid dwEvent(%d)", dwEvent); - break; - } - dwIndex = (dwEvent - WAIT_OBJECT_0); - - // lock - if ((hr = IDirectSoundCaptureBuffer_Lock(dsound->captureBuffer, (DWORD)(dwIndex * dsound->bytes_per_notif_size), (DWORD)dsound->bytes_per_notif_size, &lpvAudio1, &dwBytesAudio1, &lpvAudio2, &dwBytesAudio2, 0)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCaptureBuffer_Lock", hr); - continue; - } - - if (TMEDIA_PRODUCER(dsound)->enc_cb.callback) { + tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)param; + + HRESULT hr; + LPVOID lpvAudio1, lpvAudio2; + DWORD dwBytesAudio1, dwBytesAudio2, dwEvent, dwIndex; + + TSK_DEBUG_INFO("_tdav_producer_dsound_record_thread -- START"); + + SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_TIME_CRITICAL); + + while (dsound->started) { + dwEvent = WaitForMultipleObjects(TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT, dsound->notifEvents, FALSE, INFINITE); + if (!dsound->started) { + break; + } + if (dwEvent < WAIT_OBJECT_0 || dwEvent >(WAIT_OBJECT_0 + TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT)) { + TSK_DEBUG_ERROR("Invalid dwEvent(%d)", dwEvent); + break; + } + dwIndex = (dwEvent - WAIT_OBJECT_0); + + // lock + if ((hr = IDirectSoundCaptureBuffer_Lock(dsound->captureBuffer, (DWORD)(dwIndex * dsound->bytes_per_notif_size), (DWORD)dsound->bytes_per_notif_size, &lpvAudio1, &dwBytesAudio1, &lpvAudio2, &dwBytesAudio2, 0)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCaptureBuffer_Lock", hr); + continue; + } + + if (TMEDIA_PRODUCER(dsound)->enc_cb.callback) { #if SEND_SILENCE_ON_MUTE - if (dsound->mute) { - memset(lpvAudio1, 0, dwBytesAudio1); - if(lpvAudio2){ - memset(lpvAudio2, 0, dwBytesAudio2); - } - } + if (dsound->mute) { + memset(lpvAudio1, 0, dwBytesAudio1); + if(lpvAudio2) { + memset(lpvAudio2, 0, dwBytesAudio2); + } + } #endif - TMEDIA_PRODUCER(dsound)->enc_cb.callback(TMEDIA_PRODUCER(dsound)->enc_cb.callback_data, lpvAudio1, dwBytesAudio1); - if (lpvAudio2) { - TMEDIA_PRODUCER(dsound)->enc_cb.callback(TMEDIA_PRODUCER(dsound)->enc_cb.callback_data, lpvAudio2, dwBytesAudio2); - } - } + TMEDIA_PRODUCER(dsound)->enc_cb.callback(TMEDIA_PRODUCER(dsound)->enc_cb.callback_data, lpvAudio1, dwBytesAudio1); + if (lpvAudio2) { + TMEDIA_PRODUCER(dsound)->enc_cb.callback(TMEDIA_PRODUCER(dsound)->enc_cb.callback_data, lpvAudio2, dwBytesAudio2); + } + } - // unlock - if ((hr = IDirectSoundCaptureBuffer_Unlock(dsound->captureBuffer, lpvAudio1, dwBytesAudio1, lpvAudio2, dwBytesAudio2)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCaptureBuffer_Unlock", hr); - continue; - } - } + // unlock + if ((hr = IDirectSoundCaptureBuffer_Unlock(dsound->captureBuffer, lpvAudio1, dwBytesAudio1, lpvAudio2, dwBytesAudio2)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCaptureBuffer_Unlock", hr); + continue; + } + } - TSK_DEBUG_INFO("_tdav_producer_dsound_record_thread -- STOP"); + TSK_DEBUG_INFO("_tdav_producer_dsound_record_thread -- STOP"); - return tsk_null; + return tsk_null; } static int _tdav_producer_dsound_unprepare(tdav_producer_dsound_t* dsound) { - if (dsound) { - tsk_size_t i; - if (dsound->captureBuffer) { - IDirectSoundCaptureBuffer_Release(dsound->captureBuffer); - dsound->captureBuffer = NULL; - } - if (dsound->device) { - IDirectSoundCapture_Release(dsound->device); - dsound->device = NULL; - } - for (i = 0; i < (sizeof(dsound->notifEvents) / sizeof(dsound->notifEvents[0])); i++){ - if (dsound->notifEvents[i]) { - CloseHandle(dsound->notifEvents[i]); - dsound->notifEvents[i] = NULL; - } - } - } - return 0; + if (dsound) { + tsk_size_t i; + if (dsound->captureBuffer) { + IDirectSoundCaptureBuffer_Release(dsound->captureBuffer); + dsound->captureBuffer = NULL; + } + if (dsound->device) { + IDirectSoundCapture_Release(dsound->device); + dsound->device = NULL; + } + for (i = 0; i < (sizeof(dsound->notifEvents) / sizeof(dsound->notifEvents[0])); i++) { + if (dsound->notifEvents[i]) { + CloseHandle(dsound->notifEvents[i]); + dsound->notifEvents[i] = NULL; + } + } + } + return 0; } @@ -153,191 +152,191 @@ static int _tdav_producer_dsound_unprepare(tdav_producer_dsound_t* dsound) /* ============ Media Producer Interface ================= */ static int tdav_producer_dsound_set(tmedia_producer_t* self, const tmedia_param_t* param) { - tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; - if (param->plugin_type == tmedia_ppt_producer) { - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "volume")) { - return 0; - } - else if (tsk_striequals(param->key, "mute")) { - dsound->mute = (TSK_TO_INT32((uint8_t*)param->value) != 0); + tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; + if (param->plugin_type == tmedia_ppt_producer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "volume")) { + return 0; + } + else if (tsk_striequals(param->key, "mute")) { + dsound->mute = (TSK_TO_INT32((uint8_t*)param->value) != 0); #if !SEND_SILENCE_ON_MUTE - if (dsound->started) { - if (dsound->mute) { - IDirectSoundCaptureBuffer_Stop(dsound->captureBuffer); - } - else { - IDirectSoundCaptureBuffer_Start(dsound->captureBuffer, DSBPLAY_LOOPING); - } - } + if (dsound->started) { + if (dsound->mute) { + IDirectSoundCaptureBuffer_Stop(dsound->captureBuffer); + } + else { + IDirectSoundCaptureBuffer_Start(dsound->captureBuffer, DSBPLAY_LOOPING); + } + } #endif - return 0; - } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); + return 0; + } + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int tdav_producer_dsound_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - HRESULT hr; + HRESULT hr; - WAVEFORMATEX wfx = { 0 }; - DSCBUFFERDESC dsbd = { 0 }; + WAVEFORMATEX wfx = { 0 }; + DSCBUFFERDESC dsbd = { 0 }; - tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; + tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; - if (!dsound || !codec) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!dsound || !codec) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (dsound->device || dsound->captureBuffer) { - TSK_DEBUG_ERROR("Producer already prepared"); - return -2; - } + if (dsound->device || dsound->captureBuffer) { + TSK_DEBUG_ERROR("Producer already prepared"); + return -2; + } - TMEDIA_PRODUCER(dsound)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER(dsound)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); - TMEDIA_PRODUCER(dsound)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(dsound)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(dsound)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); + TMEDIA_PRODUCER(dsound)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); #if 0 - TMEDIA_PRODUCER(dsound)->audio.rate = 48000; - TMEDIA_PRODUCER(dsound)->audio.channels = 1; + TMEDIA_PRODUCER(dsound)->audio.rate = 48000; + TMEDIA_PRODUCER(dsound)->audio.channels = 1; #endif - /* Create capture device */ - if ((hr = DirectSoundCaptureCreate(NULL, &dsound->device, NULL) != DS_OK)) { - tdav_win32_print_error("DirectSoundCaptureCreate", hr); - return -3; - } - - /* Creates the capture buffer */ - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = TMEDIA_PRODUCER(dsound)->audio.channels; - wfx.nSamplesPerSec = TMEDIA_PRODUCER(dsound)->audio.rate; - wfx.wBitsPerSample = TMEDIA_PRODUCER(dsound)->audio.bits_per_sample; - wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample / 8); - wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); - - /* Average bytes (count) for each notification */ - dsound->bytes_per_notif_size = ((wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(dsound)->audio.ptime) / 1000); - - dsbd.dwSize = sizeof(DSCBUFFERDESC); - dsbd.dwBufferBytes = (DWORD)(TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT * dsound->bytes_per_notif_size); - dsbd.lpwfxFormat = &wfx; - - if ((hr = IDirectSoundCapture_CreateCaptureBuffer(dsound->device, &dsbd, &dsound->captureBuffer, NULL)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCapture_CreateCaptureBuffer", hr); - return -4; - } - - return 0; + /* Create capture device */ + if ((hr = DirectSoundCaptureCreate(NULL, &dsound->device, NULL) != DS_OK)) { + tdav_win32_print_error("DirectSoundCaptureCreate", hr); + return -3; + } + + /* Creates the capture buffer */ + wfx.wFormatTag = WAVE_FORMAT_PCM; + wfx.nChannels = TMEDIA_PRODUCER(dsound)->audio.channels; + wfx.nSamplesPerSec = TMEDIA_PRODUCER(dsound)->audio.rate; + wfx.wBitsPerSample = TMEDIA_PRODUCER(dsound)->audio.bits_per_sample; + wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample / 8); + wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); + + /* Average bytes (count) for each notification */ + dsound->bytes_per_notif_size = ((wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(dsound)->audio.ptime) / 1000); + + dsbd.dwSize = sizeof(DSCBUFFERDESC); + dsbd.dwBufferBytes = (DWORD)(TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT * dsound->bytes_per_notif_size); + dsbd.lpwfxFormat = &wfx; + + if ((hr = IDirectSoundCapture_CreateCaptureBuffer(dsound->device, &dsbd, &dsound->captureBuffer, NULL)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCapture_CreateCaptureBuffer", hr); + return -4; + } + + return 0; } static int tdav_producer_dsound_start(tmedia_producer_t* self) { - tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; - - tsk_size_t i; - DWORD dwOffset; - HRESULT hr; - LPDIRECTSOUNDNOTIFY lpDSBNotify; - DSBPOSITIONNOTIFY pPosNotify[TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT] = { 0 }; - - if (!dsound) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if (!dsound->device || !dsound->captureBuffer) { - TSK_DEBUG_ERROR("Producer not prepared"); - return -2; - } - - if (dsound->started) { - return 0; - } - - if ((hr = IDirectSoundCaptureBuffer_QueryInterface(dsound->captureBuffer, &IID_IDirectSoundNotify, (LPVOID*)&lpDSBNotify)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCaptureBuffer_QueryInterface", hr); - return -3; - } - - /* Events associated to notification points */ - dwOffset = (DWORD)(dsound->bytes_per_notif_size - 1); - for (i = 0; i < (sizeof(dsound->notifEvents) / sizeof(dsound->notifEvents[0])); i++){ - dsound->notifEvents[i] = CreateEvent(NULL, FALSE, FALSE, NULL); - pPosNotify[i].dwOffset = dwOffset; - pPosNotify[i].hEventNotify = dsound->notifEvents[i]; - dwOffset += (DWORD)dsound->bytes_per_notif_size; - } - if ((hr = IDirectSoundNotify_SetNotificationPositions(lpDSBNotify, TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT, pPosNotify)) != DS_OK) { - IDirectSoundNotify_Release(lpDSBNotify); - tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); - return -4; - } - - if ((hr = IDirectSoundNotify_Release(lpDSBNotify))) { - tdav_win32_print_error("IDirectSoundNotify_Release", hr); - } - - /* Start the buffer */ - if ((hr = IDirectSoundCaptureBuffer_Start(dsound->captureBuffer, DSBPLAY_LOOPING)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCaptureBuffer_Start", hr); - return -5; - } - - /* start the reader thread */ - dsound->started = tsk_true; - tsk_thread_create(&dsound->tid[0], _tdav_producer_dsound_record_thread, dsound); - - return 0; + tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; + + tsk_size_t i; + DWORD dwOffset; + HRESULT hr; + LPDIRECTSOUNDNOTIFY lpDSBNotify; + DSBPOSITIONNOTIFY pPosNotify[TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT] = { 0 }; + + if (!dsound) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if (!dsound->device || !dsound->captureBuffer) { + TSK_DEBUG_ERROR("Producer not prepared"); + return -2; + } + + if (dsound->started) { + return 0; + } + + if ((hr = IDirectSoundCaptureBuffer_QueryInterface(dsound->captureBuffer, &IID_IDirectSoundNotify, (LPVOID*)&lpDSBNotify)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCaptureBuffer_QueryInterface", hr); + return -3; + } + + /* Events associated to notification points */ + dwOffset = (DWORD)(dsound->bytes_per_notif_size - 1); + for (i = 0; i < (sizeof(dsound->notifEvents) / sizeof(dsound->notifEvents[0])); i++) { + dsound->notifEvents[i] = CreateEvent(NULL, FALSE, FALSE, NULL); + pPosNotify[i].dwOffset = dwOffset; + pPosNotify[i].hEventNotify = dsound->notifEvents[i]; + dwOffset += (DWORD)dsound->bytes_per_notif_size; + } + if ((hr = IDirectSoundNotify_SetNotificationPositions(lpDSBNotify, TDAV_DSOUND_PRODUCER_NOTIF_POS_COUNT, pPosNotify)) != DS_OK) { + IDirectSoundNotify_Release(lpDSBNotify); + tdav_win32_print_error("IDirectSoundBuffer_QueryInterface", hr); + return -4; + } + + if ((hr = IDirectSoundNotify_Release(lpDSBNotify))) { + tdav_win32_print_error("IDirectSoundNotify_Release", hr); + } + + /* Start the buffer */ + if ((hr = IDirectSoundCaptureBuffer_Start(dsound->captureBuffer, DSBPLAY_LOOPING)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCaptureBuffer_Start", hr); + return -5; + } + + /* start the reader thread */ + dsound->started = tsk_true; + tsk_thread_create(&dsound->tid[0], _tdav_producer_dsound_record_thread, dsound); + + return 0; } static int tdav_producer_dsound_pause(tmedia_producer_t* self) { - return 0; + return 0; } static int tdav_producer_dsound_stop(tmedia_producer_t* self) { - tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; + tdav_producer_dsound_t* dsound = (tdav_producer_dsound_t*)self; - HRESULT hr; + HRESULT hr; - if (!self) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (!dsound->started) { - return 0; - } + if (!dsound->started) { + return 0; + } - // should be done here - dsound->started = tsk_false; + // should be done here + dsound->started = tsk_false; #if !SEND_SILENCE_ON_MUTE - if (dsound->mute && dsound->notifEvents[0]) { - // thread is paused -> raise event now that "started" is equal to false - SetEvent(dsound->notifEvents[0]); - } + if (dsound->mute && dsound->notifEvents[0]) { + // thread is paused -> raise event now that "started" is equal to false + SetEvent(dsound->notifEvents[0]); + } #endif - // stop thread - if (dsound->tid[0]) { - tsk_thread_join(&(dsound->tid[0])); - } + // stop thread + if (dsound->tid[0]) { + tsk_thread_join(&(dsound->tid[0])); + } - if ((hr = IDirectSoundCaptureBuffer_Stop(dsound->captureBuffer)) != DS_OK) { - tdav_win32_print_error("IDirectSoundCaptureBuffer_Stop", hr); - } + if ((hr = IDirectSoundCaptureBuffer_Stop(dsound->captureBuffer)) != DS_OK) { + tdav_win32_print_error("IDirectSoundCaptureBuffer_Stop", hr); + } - // unprepare - // will be prepared again before next start() - _tdav_producer_dsound_unprepare(dsound); + // unprepare + // will be prepared again before next start() + _tdav_producer_dsound_unprepare(dsound); - return 0; + return 0; } @@ -347,54 +346,52 @@ static int tdav_producer_dsound_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_dsound_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_dsound_t *producer = self; - if (producer) { - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); - /* init self */ - - } - return self; + tdav_producer_dsound_t *producer = self; + if (producer) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); + /* init self */ + + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_dsound_dtor(tsk_object_t * self) { - tdav_producer_dsound_t *dsound = self; - if (dsound) { - /* stop */ - if (dsound->started) { - tdav_producer_dsound_stop(self); - } - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(dsound)); - /* deinit self */ - _tdav_producer_dsound_unprepare(dsound); - } - - return self; + tdav_producer_dsound_t *dsound = self; + if (dsound) { + /* stop */ + if (dsound->started) { + tdav_producer_dsound_stop(self); + } + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(dsound)); + /* deinit self */ + _tdav_producer_dsound_unprepare(dsound); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_dsound_def_s = -{ - sizeof(tdav_producer_dsound_t), - tdav_producer_dsound_ctor, - tdav_producer_dsound_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_dsound_def_s = { + sizeof(tdav_producer_dsound_t), + tdav_producer_dsound_ctor, + tdav_producer_dsound_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_dsound_plugin_def_s = -{ - &tdav_producer_dsound_def_s, +static const tmedia_producer_plugin_def_t tdav_producer_dsound_plugin_def_s = { + &tdav_producer_dsound_def_s, - tmedia_audio, - "Microsoft DirectSound producer", + tmedia_audio, + "Microsoft DirectSound producer", - tdav_producer_dsound_set, - tdav_producer_dsound_prepare, - tdav_producer_dsound_start, - tdav_producer_dsound_pause, - tdav_producer_dsound_stop + tdav_producer_dsound_set, + tdav_producer_dsound_prepare, + tdav_producer_dsound_start, + tdav_producer_dsound_pause, + tdav_producer_dsound_stop }; const tmedia_producer_plugin_def_t *tdav_producer_dsound_plugin_def_t = &tdav_producer_dsound_plugin_def_s; diff --git a/tinyDAV/src/audio/oss/tdav_consumer_oss.c b/tinyDAV/src/audio/oss/tdav_consumer_oss.c index 0370210..6ad43e7 100755 --- a/tinyDAV/src/audio/oss/tdav_consumer_oss.c +++ b/tinyDAV/src/audio/oss/tdav_consumer_oss.c @@ -1,17 +1,17 @@ /* Copyright (C) 2014 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -39,287 +39,286 @@ #define OSS_DEBUG_ERROR(FMT, ...) TSK_DEBUG_ERROR("[OSS Consumer] " FMT, ##__VA_ARGS__) #define OSS_DEBUG_FATAL(FMT, ...) TSK_DEBUG_FATAL("[OSS Consumer] " FMT, ##__VA_ARGS__) -typedef struct tdav_consumer_oss_s -{ - TDAV_DECLARE_CONSUMER_AUDIO; - - tsk_bool_t b_started; - tsk_bool_t b_prepared; - tsk_bool_t b_muted; - int n_bits_per_sample; - - int fd; - tsk_thread_handle_t* tid[1]; - - tsk_size_t n_buff_size_in_bytes; - tsk_size_t n_buff_size_in_samples; - uint8_t* p_buff_ptr; - - tsk_size_t n_buff16_size_in_bytes; - tsk_size_t n_buff16_size_in_samples; - uint16_t* p_buff16_ptr; - - TSK_DECLARE_SAFEOBJ; +typedef struct tdav_consumer_oss_s { + TDAV_DECLARE_CONSUMER_AUDIO; + + tsk_bool_t b_started; + tsk_bool_t b_prepared; + tsk_bool_t b_muted; + int n_bits_per_sample; + + int fd; + tsk_thread_handle_t* tid[1]; + + tsk_size_t n_buff_size_in_bytes; + tsk_size_t n_buff_size_in_samples; + uint8_t* p_buff_ptr; + + tsk_size_t n_buff16_size_in_bytes; + tsk_size_t n_buff16_size_in_samples; + uint16_t* p_buff16_ptr; + + TSK_DECLARE_SAFEOBJ; } tdav_consumer_oss_t; static int __oss_from_16bits_to_8bits(const void* p_src, void* p_dst, tsk_size_t n_samples) { - tsk_size_t i; - uint16_t *_p_src = (uint16_t*)p_src; - uint8_t *_p_dst = (uint8_t*)p_dst; - - if (!p_src || !p_dst || !n_samples) { - OSS_DEBUG_ERROR("invalid parameter"); - return -1; - } - for (i = 0; i < n_samples; ++i) { - _p_dst[i] = _p_src[i]; - } - return 0; + tsk_size_t i; + uint16_t *_p_src = (uint16_t*)p_src; + uint8_t *_p_dst = (uint8_t*)p_dst; + + if (!p_src || !p_dst || !n_samples) { + OSS_DEBUG_ERROR("invalid parameter"); + return -1; + } + for (i = 0; i < n_samples; ++i) { + _p_dst[i] = _p_src[i]; + } + return 0; } static void* TSK_STDCALL _tdav_consumer_oss_playback_thread(void *param) { - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)param; - int err; - void* p_buffer = ((p_oss->n_bits_per_sample == 8) ? (void*)p_oss->p_buff16_ptr: (void*)p_oss->p_buff_ptr); - tsk_size_t n_buffer_in_bytes = (p_oss->n_bits_per_sample == 8) ? p_oss->n_buff16_size_in_bytes : p_oss->n_buff_size_in_bytes; - tsk_size_t n_buffer_in_samples = p_oss->n_buff_size_in_samples; - - const void* _p_buffer; - tsk_size_t _n_buffer_in_bytes; - - OSS_DEBUG_INFO("__playback_thread -- START"); - - tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); - - while (p_oss->b_started) { - tsk_safeobj_lock(p_oss); - err = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(p_oss), p_buffer, n_buffer_in_bytes); // requires 16bits, thread-safe - if (err >= 0) { - _p_buffer = p_buffer; - _n_buffer_in_bytes = n_buffer_in_bytes; - if (err < n_buffer_in_bytes) { - memset(((uint8_t*)p_buffer) + err, 0, (n_buffer_in_bytes - err)); - } - if (p_oss->n_bits_per_sample == 8) { - __oss_from_16bits_to_8bits(p_buffer, p_oss->p_buff_ptr, n_buffer_in_samples); - _p_buffer = p_oss->p_buff_ptr; - _n_buffer_in_bytes >>= 1; - } - if ((err = write(p_oss->fd, _p_buffer, _n_buffer_in_bytes)) != _n_buffer_in_bytes) { - OSS_DEBUG_ERROR ("Failed to read data from audio interface failed (%d -> %s)", err , strerror(errno)); - tsk_safeobj_unlock(p_oss); - goto bail; - } - } - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(p_oss)); - - tsk_safeobj_unlock(p_oss); - } + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)param; + int err; + void* p_buffer = ((p_oss->n_bits_per_sample == 8) ? (void*)p_oss->p_buff16_ptr: (void*)p_oss->p_buff_ptr); + tsk_size_t n_buffer_in_bytes = (p_oss->n_bits_per_sample == 8) ? p_oss->n_buff16_size_in_bytes : p_oss->n_buff_size_in_bytes; + tsk_size_t n_buffer_in_samples = p_oss->n_buff_size_in_samples; + + const void* _p_buffer; + tsk_size_t _n_buffer_in_bytes; + + OSS_DEBUG_INFO("__playback_thread -- START"); + + tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); + + while (p_oss->b_started) { + tsk_safeobj_lock(p_oss); + err = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(p_oss), p_buffer, n_buffer_in_bytes); // requires 16bits, thread-safe + if (err >= 0) { + _p_buffer = p_buffer; + _n_buffer_in_bytes = n_buffer_in_bytes; + if (err < n_buffer_in_bytes) { + memset(((uint8_t*)p_buffer) + err, 0, (n_buffer_in_bytes - err)); + } + if (p_oss->n_bits_per_sample == 8) { + __oss_from_16bits_to_8bits(p_buffer, p_oss->p_buff_ptr, n_buffer_in_samples); + _p_buffer = p_oss->p_buff_ptr; + _n_buffer_in_bytes >>= 1; + } + if ((err = write(p_oss->fd, _p_buffer, _n_buffer_in_bytes)) != _n_buffer_in_bytes) { + OSS_DEBUG_ERROR ("Failed to read data from audio interface failed (%d -> %s)", err , strerror(errno)); + tsk_safeobj_unlock(p_oss); + goto bail; + } + } + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(p_oss)); + + tsk_safeobj_unlock(p_oss); + } bail: - OSS_DEBUG_INFO("__playback_thread -- STOP"); - return tsk_null; + OSS_DEBUG_INFO("__playback_thread -- STOP"); + return tsk_null; } /* ============ Media Consumer Interface ================= */ static int tdav_consumer_oss_set(tmedia_consumer_t* self, const tmedia_param_t* param) { - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; - int ret = 0; + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; + int ret = 0; - ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); + ret = tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); - return ret; + return ret; } static int tdav_consumer_oss_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; - int err = 0, channels, sample_rate, bits_per_sample; + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; + int err = 0, channels, sample_rate, bits_per_sample; - if (!p_oss || !codec && codec->plugin) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!p_oss || !codec && codec->plugin) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } - tsk_safeobj_lock(p_oss); + tsk_safeobj_lock(p_oss); - if (p_oss->fd == -1) { - if ((p_oss->fd = open("/dev/dsp", O_WRONLY)) < 0) { - OSS_DEBUG_ERROR("open('/dev/dsp') failed: %s", strerror(errno)); - err = -2; - goto bail; - } - } + if (p_oss->fd == -1) { + if ((p_oss->fd = open("/dev/dsp", O_WRONLY)) < 0) { + OSS_DEBUG_ERROR("open('/dev/dsp') failed: %s", strerror(errno)); + err = -2; + goto bail; + } + } - TMEDIA_CONSUMER(p_oss)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(p_oss)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); TMEDIA_CONSUMER(p_oss)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); TMEDIA_CONSUMER(p_oss)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - - // Set using requested - channels = TMEDIA_CONSUMER(p_oss)->audio.in.channels; - sample_rate = TMEDIA_CONSUMER(p_oss)->audio.in.rate; - bits_per_sample = TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample; // 16 - - // Prepare - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_BITS, &bits_per_sample)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_BITS, %d) failed: %d->%s", bits_per_sample, err, strerror(errno)); - goto bail; - } - if (bits_per_sample != 16 && bits_per_sample != 8) { - OSS_DEBUG_ERROR("bits_per_sample=%d not supported", bits_per_sample); - err = -3; - goto bail; - } - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_CHANNELS, %d) failed: %d->%s", channels, err, strerror(errno)); - goto bail; - } - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_RATE, &sample_rate)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_RATE, %d) failed: %d->%s", sample_rate, err, strerror(errno)); - goto bail; - } - - p_oss->n_buff_size_in_bytes = (TMEDIA_CONSUMER(p_oss)->audio.ptime * sample_rate * ((bits_per_sample >> 3) * channels)) / 1000; - if (!(p_oss->p_buff_ptr = tsk_realloc(p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes))) { - OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); - err = -4; - goto bail; - } - p_oss->n_buff_size_in_samples = (p_oss->n_buff_size_in_bytes / (bits_per_sample >> 3)); - if (bits_per_sample == 8) { - p_oss->n_buff16_size_in_bytes = p_oss->n_buff_size_in_bytes << 1; - if (!(p_oss->p_buff16_ptr = tsk_realloc(p_oss->p_buff16_ptr, p_oss->n_buff16_size_in_bytes))) { - OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); - err = -5; - goto bail; - } - p_oss->n_buff16_size_in_samples = p_oss->n_buff_size_in_samples; - } - - OSS_DEBUG_INFO("prepared: req_bits_per_sample=%d; req_channels=%d; req_rate=%d, resp_bits_per_sample=%d; resp_channels=%d; resp_rate=%d /// n_buff_size_in_samples=%u;n_buff_size_in_bytes=%u", - TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample, TMEDIA_CONSUMER(p_oss)->audio.in.channels, TMEDIA_CONSUMER(p_oss)->audio.in.rate, - bits_per_sample, channels, sample_rate, - p_oss->n_buff_size_in_samples, p_oss->n_buff_size_in_bytes); - - // Set using supported (up to the resampler to convert to requested) - TMEDIA_CONSUMER(p_oss)->audio.out.channels = channels; - TMEDIA_CONSUMER(p_oss)->audio.out.rate = sample_rate; - // TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample = bits_per_sample; - - p_oss->n_bits_per_sample = bits_per_sample; - p_oss->b_prepared = tsk_true; + + // Set using requested + channels = TMEDIA_CONSUMER(p_oss)->audio.in.channels; + sample_rate = TMEDIA_CONSUMER(p_oss)->audio.in.rate; + bits_per_sample = TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample; // 16 + + // Prepare + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_BITS, &bits_per_sample)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_BITS, %d) failed: %d->%s", bits_per_sample, err, strerror(errno)); + goto bail; + } + if (bits_per_sample != 16 && bits_per_sample != 8) { + OSS_DEBUG_ERROR("bits_per_sample=%d not supported", bits_per_sample); + err = -3; + goto bail; + } + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_CHANNELS, %d) failed: %d->%s", channels, err, strerror(errno)); + goto bail; + } + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_RATE, &sample_rate)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_RATE, %d) failed: %d->%s", sample_rate, err, strerror(errno)); + goto bail; + } + + p_oss->n_buff_size_in_bytes = (TMEDIA_CONSUMER(p_oss)->audio.ptime * sample_rate * ((bits_per_sample >> 3) * channels)) / 1000; + if (!(p_oss->p_buff_ptr = tsk_realloc(p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes))) { + OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); + err = -4; + goto bail; + } + p_oss->n_buff_size_in_samples = (p_oss->n_buff_size_in_bytes / (bits_per_sample >> 3)); + if (bits_per_sample == 8) { + p_oss->n_buff16_size_in_bytes = p_oss->n_buff_size_in_bytes << 1; + if (!(p_oss->p_buff16_ptr = tsk_realloc(p_oss->p_buff16_ptr, p_oss->n_buff16_size_in_bytes))) { + OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); + err = -5; + goto bail; + } + p_oss->n_buff16_size_in_samples = p_oss->n_buff_size_in_samples; + } + + OSS_DEBUG_INFO("prepared: req_bits_per_sample=%d; req_channels=%d; req_rate=%d, resp_bits_per_sample=%d; resp_channels=%d; resp_rate=%d /// n_buff_size_in_samples=%u;n_buff_size_in_bytes=%u", + TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample, TMEDIA_CONSUMER(p_oss)->audio.in.channels, TMEDIA_CONSUMER(p_oss)->audio.in.rate, + bits_per_sample, channels, sample_rate, + p_oss->n_buff_size_in_samples, p_oss->n_buff_size_in_bytes); + + // Set using supported (up to the resampler to convert to requested) + TMEDIA_CONSUMER(p_oss)->audio.out.channels = channels; + TMEDIA_CONSUMER(p_oss)->audio.out.rate = sample_rate; + // TMEDIA_CONSUMER(p_oss)->audio.bits_per_sample = bits_per_sample; + + p_oss->n_bits_per_sample = bits_per_sample; + p_oss->b_prepared = tsk_true; bail: - if (err) { - if (p_oss->fd != -1) { - close(p_oss->fd); - p_oss->fd = -1; - } - } - tsk_safeobj_unlock(p_oss); - - return err; + if (err) { + if (p_oss->fd != -1) { + close(p_oss->fd); + p_oss->fd = -1; + } + } + tsk_safeobj_unlock(p_oss); + + return err; } static int tdav_consumer_oss_start(tmedia_consumer_t* self) { - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; - int err = 0; + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; + int err = 0; - if (! p_oss) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (! p_oss) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } - tsk_safeobj_lock(p_oss); + tsk_safeobj_lock(p_oss); - if (!p_oss->b_prepared) { - OSS_DEBUG_WARN("Not prepared"); - err = -2; - goto bail; - } + if (!p_oss->b_prepared) { + OSS_DEBUG_WARN("Not prepared"); + err = -2; + goto bail; + } - if (p_oss->b_started) { - OSS_DEBUG_WARN("Already started"); - goto bail; - } + if (p_oss->b_started) { + OSS_DEBUG_WARN("Already started"); + goto bail; + } - /* start thread */ - p_oss->b_started = tsk_true; - tsk_thread_create(&p_oss->tid[0], _tdav_consumer_oss_playback_thread, p_oss); + /* start thread */ + p_oss->b_started = tsk_true; + tsk_thread_create(&p_oss->tid[0], _tdav_consumer_oss_playback_thread, p_oss); - OSS_DEBUG_INFO("started"); + OSS_DEBUG_INFO("started"); bail: - tsk_safeobj_unlock(p_oss); - return err; + tsk_safeobj_unlock(p_oss); + return err; } static int tdav_consumer_oss_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - int err = 0; - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; - - if (!p_oss || !buffer || !size) { - OSS_DEBUG_ERROR("Invalid paramter"); - return -1; - } - - //tsk_safeobj_lock(p_oss); - - if (!p_oss->b_started) { - OSS_DEBUG_WARN("Not started"); - err = -2; - goto bail; - } - if ((err = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(p_oss), buffer, size, proto_hdr))/*thread-safe*/) { - OSS_DEBUG_WARN("Failed to put audio data to the jitter buffer"); - goto bail; - } - + int err = 0; + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; + + if (!p_oss || !buffer || !size) { + OSS_DEBUG_ERROR("Invalid paramter"); + return -1; + } + + //tsk_safeobj_lock(p_oss); + + if (!p_oss->b_started) { + OSS_DEBUG_WARN("Not started"); + err = -2; + goto bail; + } + if ((err = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(p_oss), buffer, size, proto_hdr))/*thread-safe*/) { + OSS_DEBUG_WARN("Failed to put audio data to the jitter buffer"); + goto bail; + } + bail: - //tsk_safeobj_unlock(p_oss); - return err; + //tsk_safeobj_unlock(p_oss); + return err; } static int tdav_consumer_oss_pause(tmedia_consumer_t* self) { - return 0; + return 0; } static int tdav_consumer_oss_stop(tmedia_consumer_t* self) { - tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; - int err; - - if (!p_oss) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(p_oss); - - /* should be done here */ - p_oss->b_started = tsk_false; - - /* stop thread */ - if (p_oss->tid[0]) { - tsk_thread_join(&(p_oss->tid[0])); - } - if (p_oss->fd != -1) { - close(p_oss->fd); - p_oss->fd = -1; - } - p_oss->b_prepared = tsk_false; - - OSS_DEBUG_INFO("stopped"); - - tsk_safeobj_unlock(p_oss); - - return 0; + tdav_consumer_oss_t* p_oss = (tdav_consumer_oss_t*)self; + int err; + + if (!p_oss) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(p_oss); + + /* should be done here */ + p_oss->b_started = tsk_false; + + /* stop thread */ + if (p_oss->tid[0]) { + tsk_thread_join(&(p_oss->tid[0])); + } + if (p_oss->fd != -1) { + close(p_oss->fd); + p_oss->fd = -1; + } + p_oss->b_prepared = tsk_false; + + OSS_DEBUG_INFO("stopped"); + + tsk_safeobj_unlock(p_oss); + + return 0; } @@ -329,68 +328,66 @@ static int tdav_consumer_oss_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_oss_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_oss_t *p_oss = self; - if (p_oss) { - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(p_oss)); - /* init self */ - - p_oss->fd = -1; - tsk_safeobj_init(p_oss); - - OSS_DEBUG_INFO("created"); - } - return self; + tdav_consumer_oss_t *p_oss = self; + if (p_oss) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(p_oss)); + /* init self */ + + p_oss->fd = -1; + tsk_safeobj_init(p_oss); + + OSS_DEBUG_INFO("created"); + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_oss_dtor(tsk_object_t * self) -{ - tdav_consumer_oss_t *p_oss = self; - if (p_oss) { - - /* stop */ - if (p_oss->b_started) { - tdav_consumer_oss_stop(self); - } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(p_oss)); - /* deinit self */ - if (p_oss->fd > 0) { - close(p_oss->fd); - p_oss->fd = -1; - } - TSK_FREE(p_oss->p_buff_ptr); - TSK_FREE(p_oss->p_buff16_ptr); - tsk_safeobj_deinit(p_oss); - - OSS_DEBUG_INFO("*** destroyed ***"); - } - - return self; +{ + tdav_consumer_oss_t *p_oss = self; + if (p_oss) { + + /* stop */ + if (p_oss->b_started) { + tdav_consumer_oss_stop(self); + } + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(p_oss)); + /* deinit self */ + if (p_oss->fd > 0) { + close(p_oss->fd); + p_oss->fd = -1; + } + TSK_FREE(p_oss->p_buff_ptr); + TSK_FREE(p_oss->p_buff16_ptr); + tsk_safeobj_deinit(p_oss); + + OSS_DEBUG_INFO("*** destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_oss_def_s = -{ - sizeof(tdav_consumer_oss_t), - tdav_consumer_oss_ctor, - tdav_consumer_oss_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_oss_def_s = { + sizeof(tdav_consumer_oss_t), + tdav_consumer_oss_ctor, + tdav_consumer_oss_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_oss_plugin_def_s = -{ - &tdav_consumer_oss_def_s, - - tmedia_audio, - "Linux OSS consumer", - - tdav_consumer_oss_set, - tdav_consumer_oss_prepare, - tdav_consumer_oss_start, - tdav_consumer_oss_consume, - tdav_consumer_oss_pause, - tdav_consumer_oss_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_oss_plugin_def_s = { + &tdav_consumer_oss_def_s, + + tmedia_audio, + "Linux OSS consumer", + + tdav_consumer_oss_set, + tdav_consumer_oss_prepare, + tdav_consumer_oss_start, + tdav_consumer_oss_consume, + tdav_consumer_oss_pause, + tdav_consumer_oss_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_oss_plugin_def_t = &tdav_consumer_oss_plugin_def_s; diff --git a/tinyDAV/src/audio/oss/tdav_producer_oss.c b/tinyDAV/src/audio/oss/tdav_producer_oss.c index d61fb96..61a7afa 100755 --- a/tinyDAV/src/audio/oss/tdav_producer_oss.c +++ b/tinyDAV/src/audio/oss/tdav_producer_oss.c @@ -1,17 +1,17 @@ /* Copyright (C) 2014 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -39,264 +39,263 @@ #define OSS_DEBUG_ERROR(FMT, ...) TSK_DEBUG_ERROR("[OSS Producer] " FMT, ##__VA_ARGS__) #define OSS_DEBUG_FATAL(FMT, ...) TSK_DEBUG_FATAL("[OSS Producer] " FMT, ##__VA_ARGS__) -typedef struct tdav_producer_oss_s -{ - TDAV_DECLARE_PRODUCER_AUDIO; - - tsk_bool_t b_started; - tsk_bool_t b_prepared; - tsk_bool_t b_muted; - int n_bits_per_sample; - - int fd; - tsk_thread_handle_t* tid[1]; - - tsk_size_t n_buff_size_in_bytes; - tsk_size_t n_buff_size_in_samples; - uint8_t* p_buff_ptr; - - tsk_size_t n_buff16_size_in_bytes; - tsk_size_t n_buff16_size_in_samples; - uint16_t* p_buff16_ptr; - - TSK_DECLARE_SAFEOBJ; +typedef struct tdav_producer_oss_s { + TDAV_DECLARE_PRODUCER_AUDIO; + + tsk_bool_t b_started; + tsk_bool_t b_prepared; + tsk_bool_t b_muted; + int n_bits_per_sample; + + int fd; + tsk_thread_handle_t* tid[1]; + + tsk_size_t n_buff_size_in_bytes; + tsk_size_t n_buff_size_in_samples; + uint8_t* p_buff_ptr; + + tsk_size_t n_buff16_size_in_bytes; + tsk_size_t n_buff16_size_in_samples; + uint16_t* p_buff16_ptr; + + TSK_DECLARE_SAFEOBJ; } tdav_producer_oss_t; static int __oss_from_8bits_to_16bits(const void* p_src, void* p_dst, tsk_size_t n_samples) { - tsk_size_t i; - const uint8_t *_p_src = (const uint8_t*)p_src; - uint16_t *_p_dst = (uint16_t*)p_dst; - - if (!p_src || !p_dst || !n_samples) { - OSS_DEBUG_ERROR("invalid parameter"); - return -1; - } - for (i = 0; i < n_samples; ++i) { - _p_dst[i] = _p_src[i]; - } - return 0; + tsk_size_t i; + const uint8_t *_p_src = (const uint8_t*)p_src; + uint16_t *_p_dst = (uint16_t*)p_dst; + + if (!p_src || !p_dst || !n_samples) { + OSS_DEBUG_ERROR("invalid parameter"); + return -1; + } + for (i = 0; i < n_samples; ++i) { + _p_dst[i] = _p_src[i]; + } + return 0; } static void* TSK_STDCALL _tdav_producer_oss_record_thread(void *param) { - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)param; - int err; - const void* p_buffer = ((p_oss->n_bits_per_sample == 8) ? (const void*)p_oss->p_buff16_ptr: (const void*)p_oss->p_buff_ptr); - tsk_size_t n_buffer_in_bytes = (p_oss->n_bits_per_sample == 8) ? p_oss->n_buff16_size_in_bytes : p_oss->n_buff_size_in_bytes; - - OSS_DEBUG_INFO("__record_thread -- START"); - - tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); - - while (p_oss->b_started) { - tsk_safeobj_lock(p_oss); - if ((err = read(p_oss->fd, p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes)) != p_oss->n_buff_size_in_bytes) { - OSS_DEBUG_ERROR ("Failed to read data from audio interface failed (%d -> %s)", err , strerror(errno)); - tsk_safeobj_unlock(p_oss); - goto bail; - } - if (p_oss->n_bits_per_sample == 8) { - if ((err = __oss_from_8bits_to_16bits(p_oss->p_buff_ptr, p_oss->p_buff16_ptr, p_oss->n_buff_size_in_samples))) { - tsk_safeobj_unlock(p_oss); - goto bail; - } - } - if (!p_oss->b_muted && TMEDIA_PRODUCER(p_oss)->enc_cb.callback) { - TMEDIA_PRODUCER(p_oss)->enc_cb.callback(TMEDIA_PRODUCER(p_oss)->enc_cb.callback_data, p_buffer, n_buffer_in_bytes); - } - tsk_safeobj_unlock(p_oss); - } + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)param; + int err; + const void* p_buffer = ((p_oss->n_bits_per_sample == 8) ? (const void*)p_oss->p_buff16_ptr: (const void*)p_oss->p_buff_ptr); + tsk_size_t n_buffer_in_bytes = (p_oss->n_bits_per_sample == 8) ? p_oss->n_buff16_size_in_bytes : p_oss->n_buff_size_in_bytes; + + OSS_DEBUG_INFO("__record_thread -- START"); + + tsk_thread_set_priority_2(TSK_THREAD_PRIORITY_TIME_CRITICAL); + + while (p_oss->b_started) { + tsk_safeobj_lock(p_oss); + if ((err = read(p_oss->fd, p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes)) != p_oss->n_buff_size_in_bytes) { + OSS_DEBUG_ERROR ("Failed to read data from audio interface failed (%d -> %s)", err , strerror(errno)); + tsk_safeobj_unlock(p_oss); + goto bail; + } + if (p_oss->n_bits_per_sample == 8) { + if ((err = __oss_from_8bits_to_16bits(p_oss->p_buff_ptr, p_oss->p_buff16_ptr, p_oss->n_buff_size_in_samples))) { + tsk_safeobj_unlock(p_oss); + goto bail; + } + } + if (!p_oss->b_muted && TMEDIA_PRODUCER(p_oss)->enc_cb.callback) { + TMEDIA_PRODUCER(p_oss)->enc_cb.callback(TMEDIA_PRODUCER(p_oss)->enc_cb.callback_data, p_buffer, n_buffer_in_bytes); + } + tsk_safeobj_unlock(p_oss); + } bail: - OSS_DEBUG_INFO("__record_thread -- STOP"); - return tsk_null; + OSS_DEBUG_INFO("__record_thread -- STOP"); + return tsk_null; } /* ============ Media Producer Interface ================= */ static int tdav_producer_oss_set(tmedia_producer_t* self, const tmedia_param_t* param) -{ - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; - if (param->plugin_type == tmedia_ppt_producer) { - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "volume")) { - return 0; - } - else if(tsk_striequals(param->key, "mute")){ - p_oss->b_muted = (TSK_TO_INT32((uint8_t*)param->value) != 0); - return 0; - } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); +{ + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + if (param->plugin_type == tmedia_ppt_producer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "volume")) { + return 0; + } + else if(tsk_striequals(param->key, "mute")) { + p_oss->b_muted = (TSK_TO_INT32((uint8_t*)param->value) != 0); + return 0; + } + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int tdav_producer_oss_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; - int err = 0, channels, sample_rate, bits_per_sample; - - if (!p_oss || !codec && codec->plugin) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(p_oss); - - if (p_oss->fd == -1) { - if ((p_oss->fd = open("/dev/dsp", O_RDONLY)) < 0) { - OSS_DEBUG_ERROR("open('/dev/dsp') failed: %s", strerror(errno)); - err = -2; - goto bail; - } - } - - // Set using requested - channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - sample_rate = TMEDIA_CODEC_RATE_ENCODING(codec); - bits_per_sample = TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample; // 16 - - // Prepare - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_BITS, &bits_per_sample)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_BITS, %d) failed: %d->%s", bits_per_sample, err, strerror(errno)); - goto bail; - } - if (bits_per_sample != 16 && bits_per_sample != 8) { - OSS_DEBUG_ERROR("bits_per_sample=%d not supported", bits_per_sample); - err = -3; - goto bail; - } - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_CHANNELS, %d) failed: %d->%s", channels, err, strerror(errno)); - goto bail; - } - if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_RATE, &sample_rate)) != 0) { - OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_RATE, %d) failed: %d->%s", sample_rate, err, strerror(errno)); - goto bail; - } - - p_oss->n_buff_size_in_bytes = (TMEDIA_PRODUCER(p_oss)->audio.ptime * sample_rate * ((bits_per_sample >> 3) * channels)) / 1000; - if (!(p_oss->p_buff_ptr = tsk_realloc(p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes))) { - OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); - err = -4; - goto bail; - } - p_oss->n_buff_size_in_samples = (p_oss->n_buff_size_in_bytes / (bits_per_sample >> 3)); - if (bits_per_sample == 8) { - p_oss->n_buff16_size_in_bytes = p_oss->n_buff_size_in_bytes << 1; - if (!(p_oss->p_buff16_ptr = tsk_realloc(p_oss->p_buff16_ptr, p_oss->n_buff16_size_in_bytes))) { - OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); - err = -5; - goto bail; - } - p_oss->n_buff16_size_in_samples = p_oss->n_buff_size_in_samples; - } - - OSS_DEBUG_INFO("prepared: req_bits_per_sample=%d; req_channels=%d; req_rate=%d, resp_bits_per_sample=%d; resp_channels=%d; resp_rate=%d /// n_buff_size_in_samples=%u;n_buff_size_in_bytes=%u", - TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample, TMEDIA_PRODUCER(p_oss)->audio.channels, TMEDIA_PRODUCER(p_oss)->audio.rate, - bits_per_sample, channels, sample_rate, - p_oss->n_buff_size_in_samples, p_oss->n_buff_size_in_bytes); - - // Set using supported (up to the resampler to convert to requested) - TMEDIA_PRODUCER(p_oss)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER(p_oss)->audio.channels = channels; - TMEDIA_PRODUCER(p_oss)->audio.rate = sample_rate; - // TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample = bits_per_sample; - - p_oss->n_bits_per_sample = bits_per_sample; - p_oss->b_prepared = tsk_true; + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + int err = 0, channels, sample_rate, bits_per_sample; + + if (!p_oss || !codec && codec->plugin) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(p_oss); + + if (p_oss->fd == -1) { + if ((p_oss->fd = open("/dev/dsp", O_RDONLY)) < 0) { + OSS_DEBUG_ERROR("open('/dev/dsp') failed: %s", strerror(errno)); + err = -2; + goto bail; + } + } + + // Set using requested + channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + sample_rate = TMEDIA_CODEC_RATE_ENCODING(codec); + bits_per_sample = TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample; // 16 + + // Prepare + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_BITS, &bits_per_sample)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_BITS, %d) failed: %d->%s", bits_per_sample, err, strerror(errno)); + goto bail; + } + if (bits_per_sample != 16 && bits_per_sample != 8) { + OSS_DEBUG_ERROR("bits_per_sample=%d not supported", bits_per_sample); + err = -3; + goto bail; + } + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_CHANNELS, %d) failed: %d->%s", channels, err, strerror(errno)); + goto bail; + } + if ((err = ioctl(p_oss->fd, SOUND_PCM_WRITE_RATE, &sample_rate)) != 0) { + OSS_DEBUG_ERROR("ioctl(SOUND_PCM_WRITE_RATE, %d) failed: %d->%s", sample_rate, err, strerror(errno)); + goto bail; + } + + p_oss->n_buff_size_in_bytes = (TMEDIA_PRODUCER(p_oss)->audio.ptime * sample_rate * ((bits_per_sample >> 3) * channels)) / 1000; + if (!(p_oss->p_buff_ptr = tsk_realloc(p_oss->p_buff_ptr, p_oss->n_buff_size_in_bytes))) { + OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); + err = -4; + goto bail; + } + p_oss->n_buff_size_in_samples = (p_oss->n_buff_size_in_bytes / (bits_per_sample >> 3)); + if (bits_per_sample == 8) { + p_oss->n_buff16_size_in_bytes = p_oss->n_buff_size_in_bytes << 1; + if (!(p_oss->p_buff16_ptr = tsk_realloc(p_oss->p_buff16_ptr, p_oss->n_buff16_size_in_bytes))) { + OSS_DEBUG_ERROR("Failed to allocate buffer with size = %u", p_oss->n_buff_size_in_bytes); + err = -5; + goto bail; + } + p_oss->n_buff16_size_in_samples = p_oss->n_buff_size_in_samples; + } + + OSS_DEBUG_INFO("prepared: req_bits_per_sample=%d; req_channels=%d; req_rate=%d, resp_bits_per_sample=%d; resp_channels=%d; resp_rate=%d /// n_buff_size_in_samples=%u;n_buff_size_in_bytes=%u", + TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample, TMEDIA_PRODUCER(p_oss)->audio.channels, TMEDIA_PRODUCER(p_oss)->audio.rate, + bits_per_sample, channels, sample_rate, + p_oss->n_buff_size_in_samples, p_oss->n_buff_size_in_bytes); + + // Set using supported (up to the resampler to convert to requested) + TMEDIA_PRODUCER(p_oss)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(p_oss)->audio.channels = channels; + TMEDIA_PRODUCER(p_oss)->audio.rate = sample_rate; + // TMEDIA_PRODUCER(p_oss)->audio.bits_per_sample = bits_per_sample; + + p_oss->n_bits_per_sample = bits_per_sample; + p_oss->b_prepared = tsk_true; bail: - if (err) { - if (p_oss->fd != -1) { - close(p_oss->fd); - p_oss->fd = -1; - } - } - tsk_safeobj_unlock(p_oss); - - return err; + if (err) { + if (p_oss->fd != -1) { + close(p_oss->fd); + p_oss->fd = -1; + } + } + tsk_safeobj_unlock(p_oss); + + return err; } static int tdav_producer_oss_start(tmedia_producer_t* self) { - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; - int err = 0; + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + int err = 0; - if (! p_oss) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (! p_oss) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } - tsk_safeobj_lock(p_oss); + tsk_safeobj_lock(p_oss); - if (!p_oss->b_prepared) { - OSS_DEBUG_WARN("Not prepared"); - err = -2; - goto bail; - } + if (!p_oss->b_prepared) { + OSS_DEBUG_WARN("Not prepared"); + err = -2; + goto bail; + } - if (p_oss->b_started) { - OSS_DEBUG_WARN("Already started"); - goto bail; - } + if (p_oss->b_started) { + OSS_DEBUG_WARN("Already started"); + goto bail; + } - /* start thread */ - p_oss->b_started = tsk_true; - tsk_thread_create(&p_oss->tid[0], _tdav_producer_oss_record_thread, p_oss); + /* start thread */ + p_oss->b_started = tsk_true; + tsk_thread_create(&p_oss->tid[0], _tdav_producer_oss_record_thread, p_oss); - OSS_DEBUG_INFO("started"); + OSS_DEBUG_INFO("started"); bail: - tsk_safeobj_unlock(p_oss); - return err; + tsk_safeobj_unlock(p_oss); + return err; } static int tdav_producer_oss_pause(tmedia_producer_t* self) { - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + + if (!p_oss) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (!p_oss) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - OSS_DEBUG_INFO("paused"); + OSS_DEBUG_INFO("paused"); - return 0; + return 0; } static int tdav_producer_oss_stop(tmedia_producer_t* self) { - tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; - int err; - - if (!p_oss) { - OSS_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(p_oss); - - /* should be done here */ - p_oss->b_started = tsk_false; - - /* stop thread */ - if (p_oss->tid[0]) { - tsk_thread_join(&(p_oss->tid[0])); - } - if (p_oss->fd != -1) { - close(p_oss->fd); - p_oss->fd = -1; - } - p_oss->b_prepared = tsk_false; - - OSS_DEBUG_INFO("stopped"); - - tsk_safeobj_unlock(p_oss); - - return 0; + tdav_producer_oss_t* p_oss = (tdav_producer_oss_t*)self; + int err; + + if (!p_oss) { + OSS_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(p_oss); + + /* should be done here */ + p_oss->b_started = tsk_false; + + /* stop thread */ + if (p_oss->tid[0]) { + tsk_thread_join(&(p_oss->tid[0])); + } + if (p_oss->fd != -1) { + close(p_oss->fd); + p_oss->fd = -1; + } + p_oss->b_prepared = tsk_false; + + OSS_DEBUG_INFO("stopped"); + + tsk_safeobj_unlock(p_oss); + + return 0; } @@ -306,63 +305,61 @@ static int tdav_producer_oss_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_oss_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_oss_t *p_oss = (tdav_producer_oss_t*)self; - if (p_oss) { - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(p_oss)); - /* init self */ - p_oss->fd = -1; - tsk_safeobj_init(p_oss); - } - return self; + tdav_producer_oss_t *p_oss = (tdav_producer_oss_t*)self; + if (p_oss) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(p_oss)); + /* init self */ + p_oss->fd = -1; + tsk_safeobj_init(p_oss); + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_oss_dtor(tsk_object_t * self) -{ - tdav_producer_oss_t *p_oss = (tdav_producer_oss_t *)self; - if (p_oss) { - /* stop */ - if (p_oss->b_started) { - tdav_producer_oss_stop((tmedia_producer_t*)p_oss); - } - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(p_oss)); - /* deinit self */ - if (p_oss->fd != -1) { - close(p_oss->fd); - p_oss->fd = -1; - } - TSK_FREE(p_oss->p_buff_ptr); - TSK_FREE(p_oss->p_buff16_ptr); - tsk_safeobj_deinit(p_oss); - - OSS_DEBUG_INFO("*** destroyed ***"); - } - - return self; +{ + tdav_producer_oss_t *p_oss = (tdav_producer_oss_t *)self; + if (p_oss) { + /* stop */ + if (p_oss->b_started) { + tdav_producer_oss_stop((tmedia_producer_t*)p_oss); + } + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(p_oss)); + /* deinit self */ + if (p_oss->fd != -1) { + close(p_oss->fd); + p_oss->fd = -1; + } + TSK_FREE(p_oss->p_buff_ptr); + TSK_FREE(p_oss->p_buff16_ptr); + tsk_safeobj_deinit(p_oss); + + OSS_DEBUG_INFO("*** destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_oss_def_s = -{ - sizeof(tdav_producer_oss_t), - tdav_producer_oss_ctor, - tdav_producer_oss_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_oss_def_s = { + sizeof(tdav_producer_oss_t), + tdav_producer_oss_ctor, + tdav_producer_oss_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_oss_plugin_def_s = -{ - &tdav_producer_oss_def_s, - - tmedia_audio, - "Linux OSS producer", - - tdav_producer_oss_set, - tdav_producer_oss_prepare, - tdav_producer_oss_start, - tdav_producer_oss_pause, - tdav_producer_oss_stop +static const tmedia_producer_plugin_def_t tdav_producer_oss_plugin_def_s = { + &tdav_producer_oss_def_s, + + tmedia_audio, + "Linux OSS producer", + + tdav_producer_oss_set, + tdav_producer_oss_prepare, + tdav_producer_oss_start, + tdav_producer_oss_pause, + tdav_producer_oss_stop }; const tmedia_producer_plugin_def_t *tdav_producer_oss_plugin_def_t = &tdav_producer_oss_plugin_def_s; diff --git a/tinyDAV/src/audio/tdav_consumer_audio.c b/tinyDAV/src/audio/tdav_consumer_audio.c index 73d9688..a07944d 100755 --- a/tinyDAV/src/audio/tdav_consumer_audio.c +++ b/tinyDAV/src/audio/tdav_consumer_audio.c @@ -36,7 +36,7 @@ #if TSK_UNDER_WINDOWS # include <Winsock2.h> // timeval #elif defined(__SYMBIAN32__) -# include <_timeval.h> +# include <_timeval.h> #else # include <sys/time.h> #endif @@ -51,29 +51,29 @@ /** Initialize audio consumer */ int tdav_consumer_audio_init(tdav_consumer_audio_t* self) { - int ret; + int ret; - TSK_DEBUG_INFO("tdav_consumer_audio_init()"); + TSK_DEBUG_INFO("tdav_consumer_audio_init()"); - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - /* base */ - if ((ret = tmedia_consumer_init(TMEDIA_CONSUMER(self)))){ - return ret; - } + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + /* base */ + if ((ret = tmedia_consumer_init(TMEDIA_CONSUMER(self)))) { + return ret; + } - /* self (should be update by prepare() by using the codec's info)*/ - TMEDIA_CONSUMER(self)->audio.bits_per_sample = TDAV_BITS_PER_SAMPLE_DEFAULT; - TMEDIA_CONSUMER(self)->audio.ptime = TDAV_PTIME_DEFAULT; - TMEDIA_CONSUMER(self)->audio.in.channels = TDAV_CHANNELS_DEFAULT; - TMEDIA_CONSUMER(self)->audio.in.rate = TDAV_RATE_DEFAULT; - TMEDIA_CONSUMER(self)->audio.gain = TSK_MIN(tmedia_defaults_get_audio_consumer_gain(), TDAV_AUDIO_GAIN_MAX); + /* self (should be update by prepare() by using the codec's info)*/ + TMEDIA_CONSUMER(self)->audio.bits_per_sample = TDAV_BITS_PER_SAMPLE_DEFAULT; + TMEDIA_CONSUMER(self)->audio.ptime = TDAV_PTIME_DEFAULT; + TMEDIA_CONSUMER(self)->audio.in.channels = TDAV_CHANNELS_DEFAULT; + TMEDIA_CONSUMER(self)->audio.in.rate = TDAV_RATE_DEFAULT; + TMEDIA_CONSUMER(self)->audio.gain = TSK_MIN(tmedia_defaults_get_audio_consumer_gain(), TDAV_AUDIO_GAIN_MAX); - tsk_safeobj_init(self); + tsk_safeobj_init(self); - return 0; + return 0; } /** @@ -87,159 +87,160 @@ int tdav_consumer_audio_init(tdav_consumer_audio_t* self) */ int tdav_consumer_audio_cmp(const tsk_object_t* consumer1, const tsk_object_t* consumer2) { - int ret; - tsk_subsat_int32_ptr(consumer1, consumer2, &ret); - return ret; + int ret; + tsk_subsat_int32_ptr(consumer1, consumer2, &ret); + return ret; } int tdav_consumer_audio_set(tdav_consumer_audio_t* self, const tmedia_param_t* param) { - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if (param->plugin_type == tmedia_ppt_consumer){ - if (param->value_type == tmedia_pvt_int32){ - if (tsk_striequals(param->key, "gain")){ - int32_t gain = *((int32_t*)param->value); - if (gain < TDAV_AUDIO_GAIN_MAX && gain >= 0){ - TMEDIA_CONSUMER(self)->audio.gain = (uint8_t)gain; - TSK_DEBUG_INFO("audio consumer gain=%u", gain); - } - else{ - TSK_DEBUG_ERROR("%u is invalid as gain value", gain); - return -2; - } - } - else if (tsk_striequals(param->key, "volume")){ - TMEDIA_CONSUMER(self)->audio.volume = TSK_TO_INT32((uint8_t*)param->value); - TMEDIA_CONSUMER(self)->audio.volume = TSK_CLAMP(0, TMEDIA_CONSUMER(self)->audio.volume, 100); - } - } - } - - return 0; + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if (param->plugin_type == tmedia_ppt_consumer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "gain")) { + int32_t gain = *((int32_t*)param->value); + if (gain < TDAV_AUDIO_GAIN_MAX && gain >= 0) { + TMEDIA_CONSUMER(self)->audio.gain = (uint8_t)gain; + TSK_DEBUG_INFO("audio consumer gain=%u", gain); + } + else { + TSK_DEBUG_ERROR("%u is invalid as gain value", gain); + return -2; + } + } + else if (tsk_striequals(param->key, "volume")) { + TMEDIA_CONSUMER(self)->audio.volume = TSK_TO_INT32((uint8_t*)param->value); + TMEDIA_CONSUMER(self)->audio.volume = TSK_CLAMP(0, TMEDIA_CONSUMER(self)->audio.volume, 100); + } + } + } + + return 0; } /* put data (bytes not shorts) into the jitter buffer (consumers always have ptime of 20ms) */ int tdav_consumer_audio_put(tdav_consumer_audio_t* self, const void* data, tsk_size_t data_size, const tsk_object_t* proto_hdr) { - int ret; + int ret; - if (!self || !data || !self->jitterbuffer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!self || !data || !self->jitterbuffer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - tsk_safeobj_lock(self); + tsk_safeobj_lock(self); - if (!TMEDIA_JITTER_BUFFER(self->jitterbuffer)->opened){ - uint32_t rate = TMEDIA_CONSUMER(self)->audio.out.rate ? TMEDIA_CONSUMER(self)->audio.out.rate : TMEDIA_CONSUMER(self)->audio.in.rate; - uint32_t channels = TMEDIA_CONSUMER(self)->audio.out.channels ? TMEDIA_CONSUMER(self)->audio.out.channels : tmedia_defaults_get_audio_channels_playback(); - if ((ret = tmedia_jitterbuffer_open(self->jitterbuffer, TMEDIA_CONSUMER(self)->audio.ptime, rate, channels))){ - TSK_DEBUG_ERROR("Failed to open jitterbuffer (%d)", ret); - tsk_safeobj_unlock(self); - return ret; - } - } + if (!TMEDIA_JITTER_BUFFER(self->jitterbuffer)->opened) { + uint32_t rate = TMEDIA_CONSUMER(self)->audio.out.rate ? TMEDIA_CONSUMER(self)->audio.out.rate : TMEDIA_CONSUMER(self)->audio.in.rate; + uint32_t channels = TMEDIA_CONSUMER(self)->audio.out.channels ? TMEDIA_CONSUMER(self)->audio.out.channels : tmedia_defaults_get_audio_channels_playback(); + if ((ret = tmedia_jitterbuffer_open(self->jitterbuffer, TMEDIA_CONSUMER(self)->audio.ptime, rate, channels))) { + TSK_DEBUG_ERROR("Failed to open jitterbuffer (%d)", ret); + tsk_safeobj_unlock(self); + return ret; + } + } - ret = tmedia_jitterbuffer_put(self->jitterbuffer, (void*)data, data_size, proto_hdr); + ret = tmedia_jitterbuffer_put(self->jitterbuffer, (void*)data, data_size, proto_hdr); - tsk_safeobj_unlock(self); + tsk_safeobj_unlock(self); - return ret; + return ret; } /* get data from the jitter buffer (consumers should always have ptime of 20ms) */ tsk_size_t tdav_consumer_audio_get(tdav_consumer_audio_t* self, void* out_data, tsk_size_t out_size) { - tsk_size_t ret_size = 0; - if (!self || !self->jitterbuffer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return 0; - } - - tsk_safeobj_lock(self); - - if (!TMEDIA_JITTER_BUFFER(self->jitterbuffer)->opened){ - int ret; - uint32_t frame_duration = TMEDIA_CONSUMER(self)->audio.ptime; - uint32_t rate = TMEDIA_CONSUMER(self)->audio.out.rate ? TMEDIA_CONSUMER(self)->audio.out.rate : TMEDIA_CONSUMER(self)->audio.in.rate; - uint32_t channels = TMEDIA_CONSUMER(self)->audio.out.channels ? TMEDIA_CONSUMER(self)->audio.out.channels : tmedia_defaults_get_audio_channels_playback(); - if ((ret = tmedia_jitterbuffer_open(TMEDIA_JITTER_BUFFER(self->jitterbuffer), frame_duration, rate, channels))){ - TSK_DEBUG_ERROR("Failed to open jitterbuffer (%d)", ret); - tsk_safeobj_unlock(self); - return 0; - } - } - ret_size = tmedia_jitterbuffer_get(TMEDIA_JITTER_BUFFER(self->jitterbuffer), out_data, out_size); - - tsk_safeobj_unlock(self); - - // denoiser - if (self->denoise && self->denoise->opened && (self->denoise->echo_supp_enabled || self->denoise->noise_supp_enabled)) { - if (self->denoise->echo_supp_enabled) { - // Echo process last frame - if (self->denoise->playback_frame && self->denoise->playback_frame->size) { - tmedia_denoise_echo_playback(self->denoise, self->denoise->playback_frame->data, (uint32_t)self->denoise->playback_frame->size); - } - if (ret_size){ - // save - tsk_buffer_copy(self->denoise->playback_frame, 0, out_data, ret_size); - } - } + tsk_size_t ret_size = 0; + if (!self || !self->jitterbuffer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return 0; + } + + tsk_safeobj_lock(self); + + if (!TMEDIA_JITTER_BUFFER(self->jitterbuffer)->opened) { + int ret; + uint32_t frame_duration = TMEDIA_CONSUMER(self)->audio.ptime; + uint32_t rate = TMEDIA_CONSUMER(self)->audio.out.rate ? TMEDIA_CONSUMER(self)->audio.out.rate : TMEDIA_CONSUMER(self)->audio.in.rate; + uint32_t channels = TMEDIA_CONSUMER(self)->audio.out.channels ? TMEDIA_CONSUMER(self)->audio.out.channels : tmedia_defaults_get_audio_channels_playback(); + if ((ret = tmedia_jitterbuffer_open(TMEDIA_JITTER_BUFFER(self->jitterbuffer), frame_duration, rate, channels))) { + TSK_DEBUG_ERROR("Failed to open jitterbuffer (%d)", ret); + tsk_safeobj_unlock(self); + return 0; + } + } + ret_size = tmedia_jitterbuffer_get(TMEDIA_JITTER_BUFFER(self->jitterbuffer), out_data, out_size); + + tsk_safeobj_unlock(self); + + // denoiser + if (self->denoise && self->denoise->opened && (self->denoise->echo_supp_enabled || self->denoise->noise_supp_enabled)) { + if (self->denoise->echo_supp_enabled) { + // Echo process last frame + if (self->denoise->playback_frame && self->denoise->playback_frame->size) { + tmedia_denoise_echo_playback(self->denoise, self->denoise->playback_frame->data, (uint32_t)self->denoise->playback_frame->size); + } + if (ret_size) { + // save + tsk_buffer_copy(self->denoise->playback_frame, 0, out_data, ret_size); + } + } #if 1 // suppress noise if not supported by remote party's encoder - // suppress noise - if (self->denoise->noise_supp_enabled && ret_size) { - tmedia_denoise_process_playback(self->denoise, out_data, (uint32_t)ret_size); - } + // suppress noise + if (self->denoise->noise_supp_enabled && ret_size) { + tmedia_denoise_process_playback(self->denoise, out_data, (uint32_t)ret_size); + } #endif - } + } - return ret_size; + return ret_size; } int tdav_consumer_audio_tick(tdav_consumer_audio_t* self) { - if (!self || !self->jitterbuffer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return 0; - } - return tmedia_jitterbuffer_tick(TMEDIA_JITTER_BUFFER(self->jitterbuffer)); + if (!self || !self->jitterbuffer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return 0; + } + return tmedia_jitterbuffer_tick(TMEDIA_JITTER_BUFFER(self->jitterbuffer)); } /* set denioiser */ void tdav_consumer_audio_set_denoise(tdav_consumer_audio_t* self, struct tmedia_denoise_s* denoise) { - tsk_safeobj_lock(self); - TSK_OBJECT_SAFE_FREE(self->denoise); - self->denoise = (struct tmedia_denoise_s*)tsk_object_ref(denoise); - tsk_safeobj_unlock(self); + tsk_safeobj_lock(self); + TSK_OBJECT_SAFE_FREE(self->denoise); + self->denoise = (struct tmedia_denoise_s*)tsk_object_ref(denoise); + tsk_safeobj_unlock(self); } void tdav_consumer_audio_set_jitterbuffer(tdav_consumer_audio_t* self, struct tmedia_jitterbuffer_s* jitterbuffer) { - tsk_safeobj_lock(self); - TSK_OBJECT_SAFE_FREE(self->jitterbuffer); - self->jitterbuffer = (struct tmedia_jitterbuffer_s*)tsk_object_ref(jitterbuffer); - tsk_safeobj_unlock(self); + tsk_safeobj_lock(self); + TSK_OBJECT_SAFE_FREE(self->jitterbuffer); + self->jitterbuffer = (struct tmedia_jitterbuffer_s*)tsk_object_ref(jitterbuffer); + tsk_safeobj_unlock(self); } /** Reset jitterbuffer */ -int tdav_consumer_audio_reset(tdav_consumer_audio_t* self){ - int ret; - if (!self) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - tsk_safeobj_lock(self); - ret = tmedia_jitterbuffer_reset(TMEDIA_JITTER_BUFFER(self->jitterbuffer)); - tsk_safeobj_unlock(self); - - return ret; +int tdav_consumer_audio_reset(tdav_consumer_audio_t* self) +{ + int ret; + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + tsk_safeobj_lock(self); + ret = tmedia_jitterbuffer_reset(TMEDIA_JITTER_BUFFER(self->jitterbuffer)); + tsk_safeobj_unlock(self); + + return ret; } /* tsk_safeobj_lock(self); */ @@ -248,25 +249,25 @@ int tdav_consumer_audio_reset(tdav_consumer_audio_t* self){ /** DeInitialize audio consumer */ int tdav_consumer_audio_deinit(tdav_consumer_audio_t* self) { - int ret; + int ret; - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - /* base */ - if ((ret = tmedia_consumer_deinit(TMEDIA_CONSUMER(self)))){ - /* return ret; */ - } + /* base */ + if ((ret = tmedia_consumer_deinit(TMEDIA_CONSUMER(self)))) { + /* return ret; */ + } - /* self */ - TSK_OBJECT_SAFE_FREE(self->denoise); - TSK_OBJECT_SAFE_FREE(self->resampler); - TSK_OBJECT_SAFE_FREE(self->jitterbuffer); + /* self */ + TSK_OBJECT_SAFE_FREE(self->denoise); + TSK_OBJECT_SAFE_FREE(self->resampler); + TSK_OBJECT_SAFE_FREE(self->jitterbuffer); - tsk_safeobj_deinit(self); + tsk_safeobj_deinit(self); - return 0; + return 0; } diff --git a/tinyDAV/src/audio/tdav_jitterbuffer.c b/tinyDAV/src/audio/tdav_jitterbuffer.c index 4fd1010..b2cd287 100755 --- a/tinyDAV/src/audio/tdav_jitterbuffer.c +++ b/tinyDAV/src/audio/tdav_jitterbuffer.c @@ -36,8 +36,8 @@ #include <string.h> #include <limits.h> -#define jb_warn(...) (warnf ? warnf(__VA_ARGS__) : (void)0) -#define jb_err(...) (errf ? errf(__VA_ARGS__) : (void)0) +#define jb_warn(...) (warnf ? warnf(__VA_ARGS__) : (void)0) +#define jb_err(...) (errf ? errf(__VA_ARGS__) : (void)0) #define jb_dbg(...) (dbgf ? dbgf(__VA_ARGS__) : (void)0) //public functions @@ -47,38 +47,40 @@ void jb_reset_all(jitterbuffer *jb); void jb_destroy(jitterbuffer *jb); void jb_set_settings(jitterbuffer *jb, jb_settings *settings); -void jb_get_info(jitterbuffer *jb, jb_info *stats); -void jb_get_settings(jitterbuffer *jb, jb_settings *settings); -float jb_guess_mos(float p, long d, int codec); +void jb_get_info(jitterbuffer *jb, jb_info *stats); +void jb_get_settings(jitterbuffer *jb, jb_settings *settings); +float jb_guess_mos(float p, long d, int codec); int jb_has_frames(jitterbuffer *jb); -void jb_put(jitterbuffer *jb, void *data, int type, long ms, long ts, long now, int codec); +void jb_put(jitterbuffer *jb, void *data, int type, long ms, long ts, long now, int codec); int jb_get(jitterbuffer *jb, void **data, long now, long interpl); //private functions -static void set_default_settings(jitterbuffer *jb); -static void reset(jitterbuffer *jb); -static long find_pointer(long *array, long max_index, long value); static void frame_free(jb_frame *frame); - -static void put_control(jitterbuffer *jb, void *data, int type, long ts); -static void put_voice(jitterbuffer *jb, void *data, int type, long ms, long ts, int codec); -static void put_history(jitterbuffer *jb, long ts, long now, long ms, int codec); +static void set_default_settings(jitterbuffer *jb); +static void reset(jitterbuffer *jb); +static long find_pointer(long *array, long max_index, long value); +static void frame_free(jb_frame *frame); + +static void put_control(jitterbuffer *jb, void *data, int type, long ts); +static void put_voice(jitterbuffer *jb, void *data, int type, long ms, long ts, int codec); +static void put_history(jitterbuffer *jb, long ts, long now, long ms, int codec); static void calculate_info(jitterbuffer *jb, long ts, long now, int codec); -static int get_control(jitterbuffer *jb, void **data); -static int get_voice(jitterbuffer *jb, void **data, long now, long interpl); +static int get_control(jitterbuffer *jb, void **data); +static int get_voice(jitterbuffer *jb, void **data, long now, long interpl); static int get_voicecase(jitterbuffer *jb, void **data, long now, long interpl, long diff); -static int get_next_frametype(jitterbuffer *jb, long ts); -static long get_next_framets(jitterbuffer *jb); -static jb_frame *get_frame(jitterbuffer *jb, long ts); +static int get_next_frametype(jitterbuffer *jb, long ts); +static long get_next_framets(jitterbuffer *jb); +static jb_frame *get_frame(jitterbuffer *jb, long ts); static jb_frame *get_all_frames(jitterbuffer *jb); //debug... -static jb_output_function_t warnf, errf, dbgf; -void jb_setoutput(jb_output_function_t warn, jb_output_function_t err, jb_output_function_t dbg) { +static jb_output_function_t warnf, errf, dbgf; +void jb_setoutput(jb_output_function_t warn, jb_output_function_t err, jb_output_function_t dbg) +{ warnf = warn; errf = err; dbgf = dbg; @@ -90,74 +92,74 @@ void jb_setoutput(jb_output_function_t warn, jb_output_function_t err, jb_output * return NULL if malloc doesn't work * else return jb with default_settings. */ -jitterbuffer *jb_new() +jitterbuffer *jb_new() { - jitterbuffer *jb; - - jb_dbg("N"); - jb = tsk_calloc(1, sizeof(jitterbuffer)); - if (!jb) { - jb_err("cannot allocate jitterbuffer\n"); - return NULL; - } - set_default_settings(jb); - reset(jb); - return jb; + jitterbuffer *jb; + + jb_dbg("N"); + jb = tsk_calloc(1, sizeof(jitterbuffer)); + if (!jb) { + jb_err("cannot allocate jitterbuffer\n"); + return NULL; + } + set_default_settings(jb); + reset(jb); + return jb; } /*********** - * empty voice messages - * reset statistics + * empty voice messages + * reset statistics * keep the settings */ -void jb_reset(jitterbuffer *jb) +void jb_reset(jitterbuffer *jb) { - jb_frame *frame; - - jb_dbg("R"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_reset()\n"); - return; - } - - //free voice - while(jb->voiceframes) { - frame = get_all_frames(jb); - frame_free(frame); - } - //reset stats - memset(&(jb->info),0,sizeof(jb_info) ); - // set default settings - reset(jb); + jb_frame *frame; + + jb_dbg("R"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_reset()\n"); + return; + } + + //free voice + while(jb->voiceframes) { + frame = get_all_frames(jb); + frame_free(frame); + } + //reset stats + memset(&(jb->info),0,sizeof(jb_info) ); + // set default settings + reset(jb); } /*********** * empty nonvoice messages * empty voice messages - * reset statistics + * reset statistics * reset settings to default */ -void jb_reset_all(jitterbuffer *jb) +void jb_reset_all(jitterbuffer *jb) { - jb_frame *frame; - - jb_dbg("r"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_reset_all()\n"); - return; - } - - // free nonvoice - while(jb->controlframes) { - frame = jb->controlframes; - jb->controlframes = frame->next; - frame_free(frame); - } - // free voice and reset statistics is done by jb_reset - jb_reset(jb); - set_default_settings(jb); + jb_frame *frame; + + jb_dbg("r"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_reset_all()\n"); + return; + } + + // free nonvoice + while(jb->controlframes) { + frame = jb->controlframes; + jb->controlframes = frame->next; + frame_free(frame); + } + // free voice and reset statistics is done by jb_reset + jb_reset(jb); + set_default_settings(jb); } @@ -166,54 +168,54 @@ void jb_reset_all(jitterbuffer *jb) * free all the [non]voice frames with reset_all * free the jitterbuffer */ -void jb_destroy(jitterbuffer *jb) +void jb_destroy(jitterbuffer *jb) { - jb_dbg("D"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_destroy()\n"); - return; - } - - jb_reset_all(jb); - free(jb); + jb_dbg("D"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_destroy()\n"); + return; + } + + jb_reset_all(jb); + free(jb); } /*********** - * Set settings for the jitterbuffer. + * Set settings for the jitterbuffer. * Only if a setting is defined it will be written * in the jb->settings. * This means that no setting can be set to zero */ -void jb_set_settings(jitterbuffer *jb, jb_settings *settings) +void jb_set_settings(jitterbuffer *jb, jb_settings *settings) { - jb_dbg("S"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_set_settings()\n"); - return; - } - - if (settings->min_jb) { - jb->settings.min_jb = settings->min_jb; - } - if (settings->max_jb) { - jb->settings.max_jb = settings->max_jb; - } - if (settings->max_successive_interp) { - jb->settings.max_successive_interp = settings->max_successive_interp; - } - if (settings->extra_delay) { - jb->settings.extra_delay = settings->extra_delay; - } - if (settings->wait_grow) { - jb->settings.wait_grow = settings->wait_grow; - } - if (settings->wait_shrink) { - jb->settings.wait_shrink = settings->wait_shrink; - } - if (settings->max_diff) { - jb->settings.max_diff = settings->max_diff; - } + jb_dbg("S"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_set_settings()\n"); + return; + } + + if (settings->min_jb) { + jb->settings.min_jb = settings->min_jb; + } + if (settings->max_jb) { + jb->settings.max_jb = settings->max_jb; + } + if (settings->max_successive_interp) { + jb->settings.max_successive_interp = settings->max_successive_interp; + } + if (settings->extra_delay) { + jb->settings.extra_delay = settings->extra_delay; + } + if (settings->wait_grow) { + jb->settings.wait_grow = settings->wait_grow; + } + if (settings->wait_shrink) { + jb->settings.wait_shrink = settings->wait_shrink; + } + if (settings->max_diff) { + jb->settings.max_diff = settings->max_diff; + } } @@ -223,34 +225,35 @@ void jb_set_settings(jitterbuffer *jb, jb_settings *settings) * delay and delay_target will be calculated * *stats = info */ -void jb_get_info(jitterbuffer *jb, jb_info *stats) +void jb_get_info(jitterbuffer *jb, jb_info *stats) { - long max_index, pointer; - - jb_dbg("I"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_get_info()\n"); - return; - } - - jb->info.delay = jb->current - jb->min; - jb->info.delay_target = jb->target - jb->min; - - //calculate the losspct... - max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? -jb->hist_pointer : JB_HISTORY_SIZE-1; - if (max_index>1) { - pointer = find_pointer(&jb->hist_sorted_delay[0], max_index, -jb->current); - jb->info.losspct = ((max_index - pointer)*100/max_index); - if (jb->info.losspct < 0) { - jb->info.losspct = 0; - } - } else { - jb->info.losspct = 0; - } - - *stats = jb->info; + long max_index, pointer; + + jb_dbg("I"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_get_info()\n"); + return; + } + + jb->info.delay = jb->current - jb->min; + jb->info.delay_target = jb->target - jb->min; + + //calculate the losspct... + max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? + jb->hist_pointer : JB_HISTORY_SIZE-1; + if (max_index>1) { + pointer = find_pointer(&jb->hist_sorted_delay[0], max_index, + jb->current); + jb->info.losspct = ((max_index - pointer)*100/max_index); + if (jb->info.losspct < 0) { + jb->info.losspct = 0; + } + } + else { + jb->info.losspct = 0; + } + + *stats = jb->info; } @@ -258,56 +261,56 @@ jb->current); * gives the settings for this jitterbuffer * *settings = settings */ -void jb_get_settings(jitterbuffer *jb, jb_settings *settings) +void jb_get_settings(jitterbuffer *jb, jb_settings *settings) { - jb_dbg("S"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_get_settings()\n"); - return; - } - - *settings = jb->settings; + jb_dbg("S"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_get_settings()\n"); + return; + } + + *settings = jb->settings; } /*********** - * returns an estimate on the MOS with given loss, delay and codec + * returns an estimate on the MOS with given loss, delay and codec * if the formula is not present the default will be used * please use the JB_CODEC_OTHER if you want to define your own formula - * + * */ -float jb_guess_mos(float p, long d, int codec) +float jb_guess_mos(float p, long d, int codec) { - float result; - - switch (codec) { - case JB_CODEC_GSM_EFR: - result = (4.31f - 0.23f*p - 0.0071f*d); - break; - - case JB_CODEC_G723_1: - result = (3.99f - 0.16f*p - 0.0071f*d); - break; - - case JB_CODEC_G729: - case JB_CODEC_G729A: - result = (4.13f - 0.14f*p - 0.0071f*d); - break; + float result; + + switch (codec) { + case JB_CODEC_GSM_EFR: + result = (4.31f - 0.23f*p - 0.0071f*d); + break; + + case JB_CODEC_G723_1: + result = (3.99f - 0.16f*p - 0.0071f*d); + break; + + case JB_CODEC_G729: + case JB_CODEC_G729A: + result = (4.13f - 0.14f*p - 0.0071f*d); + break; case JB_CODEC_G711x_PLC: - result = (4.42f - 0.087f*p - 0.0071f*d); - break; + result = (4.42f - 0.087f*p - 0.0071f*d); + break; case JB_CODEC_G711x: - result = (4.42f - 0.63f*p - 0.0071f*d); - break; - + result = (4.42f - 0.63f*p - 0.0071f*d); + break; + case JB_CODEC_OTHER: default: - result = (4.42f - 0.63f*p - 0.0071f*d); + result = (4.42f - 0.63f*p - 0.0071f*d); - } - return result; + } + return result; } @@ -316,69 +319,74 @@ float jb_guess_mos(float p, long d, int codec) */ int jb_has_frames(jitterbuffer *jb) { - jb_dbg("H"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_has_frames()\n"); - return JB_NOJB; - } - - if(jb->controlframes || jb->voiceframes) { - return JB_OK; - } else { - return JB_EMPTY; - } + jb_dbg("H"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_has_frames()\n"); + return JB_NOJB; + } + + if(jb->controlframes || jb->voiceframes) { + return JB_OK; + } + else { + return JB_EMPTY; + } } /*********** - * Put a packet into the jitterbuffers + * Put a packet into the jitterbuffers * Only the timestamps of voicepackets are put in the history * this because the jitterbuffer only works for voicepackets * don't put packets twice in history and queue (e.g. transmitting every frame twice) * keep track of statistics */ -void jb_put(jitterbuffer *jb, void *data, int type, long ms, long ts, long now, int codec) -{ - long pointer, max_index; - - if (jb == NULL) { - jb_err("no jitterbuffer in jb_put()\n"); - return; - } - - jb->info.frames_received++; - - if (type == JB_TYPE_CONTROL) { - //put the packet into the contol-queue of the jitterbuffer - jb_dbg("pC"); - put_control(jb,data,type,ts); - - } else if (type == JB_TYPE_VOICE) { - // only add voice that aren't already in the buffer - max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE-1; - pointer = find_pointer(&jb->hist_sorted_timestamp[0], max_index, ts); - if (jb->hist_sorted_timestamp[pointer]==ts) { //timestamp already in queue - jb_dbg("pT"); - free(data); - jb->info.frames_dropped_twice++; - } else { //add - jb_dbg("pV"); - /* add voicepacket to history */ - put_history(jb,ts,now,ms,codec); - /*calculate jitterbuffer size*/ - calculate_info(jb, ts, now, codec); - /*put the packet into the queue of the jitterbuffer*/ - put_voice(jb,data,type,ms,ts,codec); - } - - } else if (type == JB_TYPE_SILENCE){ //silence - jb_dbg("pS"); - put_voice(jb,data,type,ms,ts,codec); - - } else {//should NEVER happen - jb_err("jb_put(): type not known\n"); - free(data); - } +void jb_put(jitterbuffer *jb, void *data, int type, long ms, long ts, long now, int codec) +{ + long pointer, max_index; + + if (jb == NULL) { + jb_err("no jitterbuffer in jb_put()\n"); + return; + } + + jb->info.frames_received++; + + if (type == JB_TYPE_CONTROL) { + //put the packet into the contol-queue of the jitterbuffer + jb_dbg("pC"); + put_control(jb,data,type,ts); + + } + else if (type == JB_TYPE_VOICE) { + // only add voice that aren't already in the buffer + max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE-1; + pointer = find_pointer(&jb->hist_sorted_timestamp[0], max_index, ts); + if (jb->hist_sorted_timestamp[pointer]==ts) { //timestamp already in queue + jb_dbg("pT"); + free(data); + jb->info.frames_dropped_twice++; + } + else { //add + jb_dbg("pV"); + /* add voicepacket to history */ + put_history(jb,ts,now,ms,codec); + /*calculate jitterbuffer size*/ + calculate_info(jb, ts, now, codec); + /*put the packet into the queue of the jitterbuffer*/ + put_voice(jb,data,type,ms,ts,codec); + } + + } + else if (type == JB_TYPE_SILENCE) { //silence + jb_dbg("pS"); + put_voice(jb,data,type,ms,ts,codec); + + } + else { //should NEVER happen + jb_err("jb_put(): type not known\n"); + free(data); + } } @@ -389,48 +397,48 @@ void jb_put(jitterbuffer *jb, void *data, int type, long ms, long ts, long now, * returns JB_INTERP if interpolating is required * returns JB_EMPTY if no voice frame is in the jitterbuffer (only during silence) */ -int jb_get(jitterbuffer *jb, void **data, long now, long interpl) +int jb_get(jitterbuffer *jb, void **data, long now, long interpl) { - int result; - - jb_dbg("A"); - if (jb == NULL) { - jb_err("no jitterbuffer in jb_get()\n"); - return JB_NOJB; - } - - result = get_control(jb, data); - if (result != JB_OK ) { //no control message available maybe there is voice... - result = get_voice(jb, data, now, interpl); - } - return result; + int result; + + jb_dbg("A"); + if (jb == NULL) { + jb_err("no jitterbuffer in jb_get()\n"); + return JB_NOJB; + } + + result = get_control(jb, data); + if (result != JB_OK ) { //no control message available maybe there is voice... + result = get_voice(jb, data, now, interpl); + } + return result; } /*********** - * set all the settings to default + * set all the settings to default */ -static void set_default_settings(jitterbuffer *jb) +static void set_default_settings(jitterbuffer *jb) { - jb->settings.min_jb = JB_MIN_SIZE; - jb->settings.max_jb = JB_MAX_SIZE; - jb->settings.max_successive_interp = JB_MAX_SUCCESSIVE_INTERP; - jb->settings.extra_delay = JB_ALLOW_EXTRA_DELAY; - jb->settings.wait_grow = JB_WAIT_GROW; - jb->settings.wait_shrink = JB_WAIT_SHRINK; - jb->settings.max_diff = JB_MAX_DIFF; + jb->settings.min_jb = JB_MIN_SIZE; + jb->settings.max_jb = JB_MAX_SIZE; + jb->settings.max_successive_interp = JB_MAX_SUCCESSIVE_INTERP; + jb->settings.extra_delay = JB_ALLOW_EXTRA_DELAY; + jb->settings.wait_grow = JB_WAIT_GROW; + jb->settings.wait_shrink = JB_WAIT_SHRINK; + jb->settings.max_diff = JB_MAX_DIFF; } /*********** - * reset the jitterbuffer so we can start in silence and + * reset the jitterbuffer so we can start in silence and * we start with a new history */ static void reset(jitterbuffer *jb) { - jb->hist_pointer = 0; //start over - jb->silence_begin_ts = 0; //no begin_ts defined - jb->info.silence =1; //we always start in silence + jb->hist_pointer = 0; //start over + jb->silence_begin_ts = 0; //no begin_ts defined + jb->info.silence =1; //we always start in silence } @@ -442,214 +450,221 @@ static void reset(jitterbuffer *jb) * if value doesn't exist return first pointer where array[low]>value * int low; //the lowest index being examined * int max_index; //the highest index being examined - * int mid; //the middle index between low and max_index. + * int mid; //the middle index between low and max_index. * mid ==(low+max_index)/2 * at the end low is the position of value or where array[low]>value - */ -static long find_pointer(long *array, long max_index, long value) + */ +static long find_pointer(long *array, long max_index, long value) { - register long low, mid, high; - low = 0; - high = max_index; - while (low<=high) { - mid= (low+high)/2; - if (array[mid] < value) { - low = mid+1; - } else { - high = mid-1; - } - } - while(low < max_index && (array[low]==array[(low+1)]) ) { - low++; - } - return low; + register long low, mid, high; + low = 0; + high = max_index; + while (low<=high) { + mid= (low+high)/2; + if (array[mid] < value) { + low = mid+1; + } + else { + high = mid-1; + } + } + while(low < max_index && (array[low]==array[(low+1)]) ) { + low++; + } + return low; } /*********** * free the given frame, afterwards the framepointer is undefined */ -static void frame_free(jb_frame *frame) +static void frame_free(jb_frame *frame) { - if (frame->data) { - free(frame->data); - } - free(frame); + if (frame->data) { + free(frame->data); + } + free(frame); } /*********** * put a nonvoice frame into the nonvoice queue */ -static void put_control(jitterbuffer *jb, void *data, int type, long ts) +static void put_control(jitterbuffer *jb, void *data, int type, long ts) { - jb_frame *frame, *p; - - frame = malloc(sizeof(jb_frame)); - if(!frame) { - jb_err("cannot allocate frame\n"); - return; - } - frame->data = data; - frame->ts = ts; - frame->type = type; - frame->next = NULL; - data = NULL;//to avoid stealing memory - - p = jb->controlframes; - if (p) { //there are already control messages - if (ts < p->ts) { - jb->controlframes = frame; - frame->next = p; - } else { - while (p->next && (ts >=p->next->ts)) {//sort on timestamps! so find place to put... - p = p->next; - } - if (p->next) { - frame->next = p->next; - } - p->next = frame; - } - } else { - jb->controlframes = frame; - } + jb_frame *frame, *p; + + frame = malloc(sizeof(jb_frame)); + if(!frame) { + jb_err("cannot allocate frame\n"); + return; + } + frame->data = data; + frame->ts = ts; + frame->type = type; + frame->next = NULL; + data = NULL;//to avoid stealing memory + + p = jb->controlframes; + if (p) { //there are already control messages + if (ts < p->ts) { + jb->controlframes = frame; + frame->next = p; + } + else { + while (p->next && (ts >=p->next->ts)) {//sort on timestamps! so find place to put... + p = p->next; + } + if (p->next) { + frame->next = p->next; + } + p->next = frame; + } + } + else { + jb->controlframes = frame; + } } /*********** - * put a voice or silence frame into the jitterbuffer + * put a voice or silence frame into the jitterbuffer */ -static void put_voice(jitterbuffer *jb, void *data, int type, long ms, long ts, int codec) +static void put_voice(jitterbuffer *jb, void *data, int type, long ms, long ts, int codec) { - jb_frame *frame, *p; - frame = malloc(sizeof(jb_frame)); - if(!frame) { - jb_err("cannot allocate frame\n"); - return; - } - - frame->data = data; - frame->ts = ts; - frame->ms = ms; - frame->type = type; - frame->codec = codec; - - data = NULL; //to avoid stealing the memory location - /* - * frames are a circular list, jb->voiceframes points to to the lowest ts, - * jb->voiceframes->prev points to the highest ts - */ - if(!jb->voiceframes) { /* queue is empty */ - jb->voiceframes = frame; - frame->next = frame; - frame->prev = frame; - } else { - p = jb->voiceframes; - if(ts < p->prev->ts) { //frame is out of order - jb->info.frames_ooo++; - } - if (ts < p->ts) { //frame is lowest, let voiceframes point to it! - jb->voiceframes = frame; - } else { - while(ts < p->prev->ts ) { - p = p->prev; - } - } - frame->next = p; - frame->prev = p->prev; - frame->next->prev = frame; - frame->prev->next = frame; - } + jb_frame *frame, *p; + frame = malloc(sizeof(jb_frame)); + if(!frame) { + jb_err("cannot allocate frame\n"); + return; + } + + frame->data = data; + frame->ts = ts; + frame->ms = ms; + frame->type = type; + frame->codec = codec; + + data = NULL; //to avoid stealing the memory location + /* + * frames are a circular list, jb->voiceframes points to to the lowest ts, + * jb->voiceframes->prev points to the highest ts + */ + if(!jb->voiceframes) { /* queue is empty */ + jb->voiceframes = frame; + frame->next = frame; + frame->prev = frame; + } + else { + p = jb->voiceframes; + if(ts < p->prev->ts) { //frame is out of order + jb->info.frames_ooo++; + } + if (ts < p->ts) { //frame is lowest, let voiceframes point to it! + jb->voiceframes = frame; + } + else { + while(ts < p->prev->ts ) { + p = p->prev; + } + } + frame->next = p; + frame->prev = p->prev; + frame->next->prev = frame; + frame->prev->next = frame; + } } /*********** * puts the timestamps of a received packet in the history of *jb * for later calculations of the size of jitterbuffer *jb. - * - * summary of function: - * - calculate delay difference - * - delete old value from hist & sorted_history_delay & sorted_history_timestamp if needed + * + * summary of function: + * - calculate delay difference + * - delete old value from hist & sorted_history_delay & sorted_history_timestamp if needed * - add new value to history & sorted_history_delay & sorted_history_timestamp - * - we keep sorted_history_delay for calculations + * - we keep sorted_history_delay for calculations * - we keep sorted_history_timestamp for ensuring each timestamp isn't put twice in the buffer. */ -static void put_history(jitterbuffer *jb, long ts, long now, long ms, int codec) +static void put_history(jitterbuffer *jb, long ts, long now, long ms, int codec) { - jb_hist_element out, in; - long max_index, pointer, location; - - // max_index is the highest possible index - max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE-1; - location = (jb->hist_pointer % JB_HISTORY_SIZE); - - // we want to delete a value from the jitterbuffer - // only when we are through the history. - if (jb->hist_pointer > JB_HISTORY_SIZE-1) { - /* the value we need to delete from sorted histories */ - out = jb->hist[location]; - //delete delay from hist_sorted_delay - pointer = find_pointer(&jb->hist_sorted_delay[0], max_index, out.delay); - /* move over pointer is the position of kicked*/ - if (pointer<max_index) { //only move if we have something to move - memmove( &(jb->hist_sorted_delay[pointer]), - &(jb->hist_sorted_delay[pointer+1]), - ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); - } - - //delete timestamp from hist_sorted_timestamp - pointer = find_pointer(&jb->hist_sorted_timestamp[0], max_index, out.ts); - /* move over pointer is the position of kicked*/ - if (pointer<max_index) { //only move if we have something to move - memmove( &(jb->hist_sorted_timestamp[pointer]), - &(jb->hist_sorted_timestamp[pointer+1]), - ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); - } - } - - in.delay = now - ts; //delay of current packet - in.ts = ts; //timestamp of current packet - in.ms = ms; //length of current packet - in.codec = codec; //codec of current packet - - /* adding the new delay to the sorted history - * first special cases: - * - delay is the first history stamp - * - delay > highest history stamp - */ - if (max_index==0 || in.delay >= jb->hist_sorted_delay[max_index-1]) { - jb->hist_sorted_delay[max_index] = in.delay; - } else { - pointer = find_pointer(&jb->hist_sorted_delay[0], (max_index-1), in.delay); - /* move over and add delay */ - memmove( &(jb->hist_sorted_delay[pointer+1]), - &(jb->hist_sorted_delay[pointer]), - ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); - jb->hist_sorted_delay[pointer] = in.delay; - } - - /* adding the new timestamp to the sorted history - * first special cases: - * - timestamp is the first history stamp - * - timestamp > highest history stamp - */ - if (max_index==0 || in.ts >= jb->hist_sorted_timestamp[max_index-1]) { - jb->hist_sorted_timestamp[max_index] = in.ts; - } else { - - pointer = find_pointer(&jb->hist_sorted_timestamp[0], (max_index-1), in.ts); - /* move over and add timestamp */ - memmove( &(jb->hist_sorted_timestamp[pointer+1]), - &(jb->hist_sorted_timestamp[pointer]), - ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); - jb->hist_sorted_timestamp[pointer] = in.ts; - } - - /* put the jb_hist_element in the history - * then increase hist_pointer for next time - */ - jb->hist[location] = in; - jb->hist_pointer++; + jb_hist_element out, in; + long max_index, pointer, location; + + // max_index is the highest possible index + max_index = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE-1; + location = (jb->hist_pointer % JB_HISTORY_SIZE); + + // we want to delete a value from the jitterbuffer + // only when we are through the history. + if (jb->hist_pointer > JB_HISTORY_SIZE-1) { + /* the value we need to delete from sorted histories */ + out = jb->hist[location]; + //delete delay from hist_sorted_delay + pointer = find_pointer(&jb->hist_sorted_delay[0], max_index, out.delay); + /* move over pointer is the position of kicked*/ + if (pointer<max_index) { //only move if we have something to move + memmove( &(jb->hist_sorted_delay[pointer]), + &(jb->hist_sorted_delay[pointer+1]), + ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); + } + + //delete timestamp from hist_sorted_timestamp + pointer = find_pointer(&jb->hist_sorted_timestamp[0], max_index, out.ts); + /* move over pointer is the position of kicked*/ + if (pointer<max_index) { //only move if we have something to move + memmove( &(jb->hist_sorted_timestamp[pointer]), + &(jb->hist_sorted_timestamp[pointer+1]), + ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); + } + } + + in.delay = now - ts; //delay of current packet + in.ts = ts; //timestamp of current packet + in.ms = ms; //length of current packet + in.codec = codec; //codec of current packet + + /* adding the new delay to the sorted history + * first special cases: + * - delay is the first history stamp + * - delay > highest history stamp + */ + if (max_index==0 || in.delay >= jb->hist_sorted_delay[max_index-1]) { + jb->hist_sorted_delay[max_index] = in.delay; + } + else { + pointer = find_pointer(&jb->hist_sorted_delay[0], (max_index-1), in.delay); + /* move over and add delay */ + memmove( &(jb->hist_sorted_delay[pointer+1]), + &(jb->hist_sorted_delay[pointer]), + ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); + jb->hist_sorted_delay[pointer] = in.delay; + } + + /* adding the new timestamp to the sorted history + * first special cases: + * - timestamp is the first history stamp + * - timestamp > highest history stamp + */ + if (max_index==0 || in.ts >= jb->hist_sorted_timestamp[max_index-1]) { + jb->hist_sorted_timestamp[max_index] = in.ts; + } + else { + + pointer = find_pointer(&jb->hist_sorted_timestamp[0], (max_index-1), in.ts); + /* move over and add timestamp */ + memmove( &(jb->hist_sorted_timestamp[pointer+1]), + &(jb->hist_sorted_timestamp[pointer]), + ((JB_HISTORY_SIZE-(pointer+1)) * sizeof(long)) ); + jb->hist_sorted_timestamp[pointer] = in.ts; + } + + /* put the jb_hist_element in the history + * then increase hist_pointer for next time + */ + jb->hist[location] = in; + jb->hist_pointer++; } @@ -659,123 +674,128 @@ static void put_history(jitterbuffer *jb, long ts, long now, long ms, int codec) * Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications * by: Kouhei Fujimoto & Shingo Ata & Masayuki Murata * http://www.nal.ics.es.osaka-u.ac.jp/achievements/web2002/pdf/journal/k-fujimo02TSJ-AdaptivePlayoutBuffer.pdf - * + * * it calculates jitter and minimum delay * get the best delay for the specified codec - + */ -static void calculate_info(jitterbuffer *jb, long ts, long now, int codec) +static void calculate_info(jitterbuffer *jb, long ts, long now, int codec) { - long diff, size, max_index, d, d1, d2, n; - float p, p1, p2, A, B; - //size = how many items there in the history - size = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE; - max_index = size-1; - - /* - * the Inter-Quartile Range can be used for estimating jitter - * http://www.slac.stanford.edu/comp/net/wan-mon/tutorial.html#variable - * just take the square root of the iqr for jitter - */ - jb->info.iqr = jb->hist_sorted_delay[max_index*3/4] - jb->hist_sorted_delay[max_index/4]; - - - /* - * The RTP way of calculating jitter. - * This one is used at the moment, although it is not correct. - * But in this way the other side understands us. - */ - diff = now - ts - jb->last_delay; - if (!jb->last_delay) { - diff = 0; //this to make sure we won't get odd jitter due first ts. - } - jb->last_delay = now - ts; - if (diff <0){ - diff = -diff; - } - jb->info.jitter = jb->info.jitter + (diff - jb->info.jitter)/16; - - /* jb->min is minimum delay in hist_sorted_delay, we don't look at the lowest 2% */ - /* because sometimes there are odd delays in there */ - jb->min = jb->hist_sorted_delay[(max_index*2/100)]; - - /* - * calculating the preferred size of the jitterbuffer: - * instead of calculating the optimum delay using the Pareto equation - * I use look at the array of sorted delays and choose my optimum from there - * always walk trough a percentage of the history this because imagine following tail: - * [...., 12, 300, 301 ,302] - * her we want to discard last three but that won't happen if we won't walk the array - * the number of frames we walk depends on how scattered the sorted delays are. - * For that we look at the iqr. The dependencies of the iqr are based on - * tests we've done here in the lab. But are not optimized. - */ - //init: - //the higest delay.. - d = d1= d2 = jb->hist_sorted_delay[max_index]- jb->min; - A=B=LONG_MIN; - p = p2 =0; - n=0; - p1 = 5; //always look at the top 5% - if (jb->info.iqr >200) { //with more jitter look at more delays - p1=25; - } else if (jb->info.iqr >100) { - p1=20; - } else if (jb->info.iqr >50){ - p1=11; - } - - //find the optimum delay.. - while(max_index>10 && (B > A ||p2<p1)) { // By MDI: from ">=" to ">" - //the packetloss with this delay - p2 =(n*100.0f/size); - // estimate MOS-value - B = jb_guess_mos(p2,d2,codec); - if (B > A) { - p = p2; - d = d2; - A = B; - } - d1 = d2; - //find next delay != delay so the same delay isn't calculated twice - //don't look further if we have seen half of the history - while((d2>=d1) && ((n*2)<max_index) ) { - n++; - d2 = jb->hist_sorted_delay[(max_index-n)] - jb->min; - } - } - //the targeted size of the jitterbuffer - if (jb->settings.min_jb && (jb->settings.min_jb > d) ) { - jb->target = jb->min + jb->settings.min_jb; - } else if (jb->settings.max_jb && (jb->settings.max_jb > d) ){ - jb->target = jb->min + jb->settings.max_jb; - } else { - jb->target = jb->min + d; - } + long diff, size, max_index, d, d1, d2, n; + float p, p1, p2, A, B; + //size = how many items there in the history + size = (jb->hist_pointer < JB_HISTORY_SIZE) ? jb->hist_pointer : JB_HISTORY_SIZE; + max_index = size-1; + + /* + * the Inter-Quartile Range can be used for estimating jitter + * http://www.slac.stanford.edu/comp/net/wan-mon/tutorial.html#variable + * just take the square root of the iqr for jitter + */ + jb->info.iqr = jb->hist_sorted_delay[max_index*3/4] - jb->hist_sorted_delay[max_index/4]; + + + /* + * The RTP way of calculating jitter. + * This one is used at the moment, although it is not correct. + * But in this way the other side understands us. + */ + diff = now - ts - jb->last_delay; + if (!jb->last_delay) { + diff = 0; //this to make sure we won't get odd jitter due first ts. + } + jb->last_delay = now - ts; + if (diff <0) { + diff = -diff; + } + jb->info.jitter = jb->info.jitter + (diff - jb->info.jitter)/16; + + /* jb->min is minimum delay in hist_sorted_delay, we don't look at the lowest 2% */ + /* because sometimes there are odd delays in there */ + jb->min = jb->hist_sorted_delay[(max_index*2/100)]; + + /* + * calculating the preferred size of the jitterbuffer: + * instead of calculating the optimum delay using the Pareto equation + * I use look at the array of sorted delays and choose my optimum from there + * always walk trough a percentage of the history this because imagine following tail: + * [...., 12, 300, 301 ,302] + * her we want to discard last three but that won't happen if we won't walk the array + * the number of frames we walk depends on how scattered the sorted delays are. + * For that we look at the iqr. The dependencies of the iqr are based on + * tests we've done here in the lab. But are not optimized. + */ + //init: + //the higest delay.. + d = d1= d2 = jb->hist_sorted_delay[max_index]- jb->min; + A=B=LONG_MIN; + p = p2 =0; + n=0; + p1 = 5; //always look at the top 5% + if (jb->info.iqr >200) { //with more jitter look at more delays + p1=25; + } + else if (jb->info.iqr >100) { + p1=20; + } + else if (jb->info.iqr >50) { + p1=11; + } + + //find the optimum delay.. + while(max_index>10 && (B > A ||p2<p1)) { // By MDI: from ">=" to ">" + //the packetloss with this delay + p2 =(n*100.0f/size); + // estimate MOS-value + B = jb_guess_mos(p2,d2,codec); + if (B > A) { + p = p2; + d = d2; + A = B; + } + d1 = d2; + //find next delay != delay so the same delay isn't calculated twice + //don't look further if we have seen half of the history + while((d2>=d1) && ((n*2)<max_index) ) { + n++; + d2 = jb->hist_sorted_delay[(max_index-n)] - jb->min; + } + } + //the targeted size of the jitterbuffer + if (jb->settings.min_jb && (jb->settings.min_jb > d) ) { + jb->target = jb->min + jb->settings.min_jb; + } + else if (jb->settings.max_jb && (jb->settings.max_jb > d) ) { + jb->target = jb->min + jb->settings.max_jb; + } + else { + jb->target = jb->min + d; + } } /*********** * if there is a nonvoice frame it will be returned [*data] and the frame * will be made free - */ -static int get_control(jitterbuffer *jb, void **data) + */ +static int get_control(jitterbuffer *jb, void **data) { - jb_frame *frame; - int result; - - frame = jb->controlframes; - if (frame) { - jb_dbg("gC"); - *data = frame->data; - frame->data = NULL; - jb->controlframes = frame->next; - frame_free(frame); - result = JB_OK; - } else { - result = JB_NOFRAME; - } - return result; + jb_frame *frame; + int result; + + frame = jb->controlframes; + if (frame) { + jb_dbg("gC"); + *data = frame->data; + frame->data = NULL; + jb->controlframes = frame->next; + frame_free(frame); + result = JB_OK; + } + else { + result = JB_NOFRAME; + } + return result; } @@ -784,68 +804,72 @@ static int get_control(jitterbuffer *jb, void **data) * returns JB_NOFRAME if it's no time to play voice and or no frame available * returns JB_INTERP if interpolating is required * returns JB_EMPTY if no voice frame is in the jitterbuffer (only during silence) - * + * * if the next frame is a silence frame we will go in silence-mode * each new instance of the jitterbuffer will start in silence mode * in silence mode we will set the jitterbuffer to the size we want - * when we are not in silence mode get_voicecase will handle the rest. + * when we are not in silence mode get_voicecase will handle the rest. */ -static int get_voice(jitterbuffer *jb, void **data, long now, long interpl) +static int get_voice(jitterbuffer *jb, void **data, long now, long interpl) { - jb_frame *frame; - long diff; - int result; - - diff = jb->target - jb->current; - - //if the next frame is a silence frame, go in silence mode... - if((get_next_frametype(jb, now - jb->current) == JB_TYPE_SILENCE) ) { - jb_dbg("gs"); - frame = get_frame(jb, now - jb->current); - *data = frame->data; - frame->data = NULL; - jb->info.silence =1; - jb->silence_begin_ts = frame->ts; - frame_free(frame); - result = JB_OK; - } else { - if(jb->info.silence) { // we are in silence - /* - * During silence we can set the jitterbuffer size to the size - * we want... - */ - if (diff) { - jb->current = jb->target; - } - frame = get_frame(jb, now - jb->current); - if (frame) { - if (jb->silence_begin_ts && frame->ts < jb->silence_begin_ts) { - jb_dbg("gL"); - /* voice frame is late, next!*/ - jb->info.frames_late++; - frame_free(frame); - result = get_voice(jb, data, now, interpl); - } else { - jb_dbg("gP"); - /* voice frame */ - jb->info.silence = 0; - jb->silence_begin_ts = 0; - jb->next_voice_time = frame->ts + frame->ms; - jb->info.last_voice_ms = frame->ms; - *data = frame->data; - frame->data = NULL; - frame_free(frame); - result = JB_OK; + jb_frame *frame; + long diff; + int result; + + diff = jb->target - jb->current; + + //if the next frame is a silence frame, go in silence mode... + if((get_next_frametype(jb, now - jb->current) == JB_TYPE_SILENCE) ) { + jb_dbg("gs"); + frame = get_frame(jb, now - jb->current); + *data = frame->data; + frame->data = NULL; + jb->info.silence =1; + jb->silence_begin_ts = frame->ts; + frame_free(frame); + result = JB_OK; + } + else { + if(jb->info.silence) { // we are in silence + /* + * During silence we can set the jitterbuffer size to the size + * we want... + */ + if (diff) { + jb->current = jb->target; + } + frame = get_frame(jb, now - jb->current); + if (frame) { + if (jb->silence_begin_ts && frame->ts < jb->silence_begin_ts) { + jb_dbg("gL"); + /* voice frame is late, next!*/ + jb->info.frames_late++; + frame_free(frame); + result = get_voice(jb, data, now, interpl); + } + else { + jb_dbg("gP"); + /* voice frame */ + jb->info.silence = 0; + jb->silence_begin_ts = 0; + jb->next_voice_time = frame->ts + frame->ms; + jb->info.last_voice_ms = frame->ms; + *data = frame->data; + frame->data = NULL; + frame_free(frame); + result = JB_OK; + } + } + else { //no frame + jb_dbg("gS"); + result = JB_EMPTY; + } + } + else { //voice case + result = get_voicecase(jb,data,now,interpl,diff); } - } else { //no frame - jb_dbg("gS"); - result = JB_EMPTY; - } - } else { //voice case - result = get_voicecase(jb,data,now,interpl,diff); - } - } - return result; + } + return result; } @@ -856,117 +880,125 @@ static int get_voice(jitterbuffer *jb, void **data, long now, long interpl) * - diff < 0, we may need to shrink * - everything else */ -static int get_voicecase(jitterbuffer *jb, void **data, long now, long interpl, long diff) +static int get_voicecase(jitterbuffer *jb, void **data, long now, long interpl, long diff) { - jb_frame *frame; - int result; - - // * - difference is way off, reset - if (diff > jb->settings.max_diff || -diff > jb->settings.max_diff) { - jb_err("wakko diff in get_voicecase\n"); - reset(jb); //reset hist because the timestamps are wakko. - result = JB_NOFRAME; - //- diff > 0, we may need to grow - } else if ((diff > 0) && - (now > (jb->last_adjustment + jb->settings.wait_grow) - || (now + jb->current + interpl) < get_next_framets(jb) ) ) { //grow - /* first try to grow */ - if (diff<interpl/2) { - jb_dbg("ag"); - jb->current +=diff; - } else { - jb_dbg("aG"); - /* grow by interp frame len */ - jb->current += interpl; - } - jb->last_adjustment = now; - result = get_voice(jb, data, now, interpl); - //- diff < 0, we may need to shrink - } else if ( (diff < 0) - && (now > (jb->last_adjustment + jb->settings.wait_shrink)) - && ((-diff) > jb->settings.extra_delay) ) { - /* now try to shrink - * if there is a frame shrink by frame length - * otherwise shrink by interpl - */ - jb->last_adjustment = now; - - frame = get_frame(jb, now - jb->current); - if(frame) { - jb_dbg("as"); - /* shrink by frame size we're throwing out */ - jb->info.frames_dropped++; - jb->current -= frame->ms; - frame_free(frame); - } else { - jb_dbg("aS"); - /* shrink by interpl */ - jb->current -= interpl; - } - result = get_voice(jb, data, now, interpl); - } else { - /* if it is not the time to play a result = JB_NOFRAME - * else We try to play a frame if a frame is available - * and not late it is played otherwise - * if available it is dropped and the next is tried - * last option is interpolating - */ - if (now - jb->current < jb->next_voice_time) { - jb_dbg("aN"); - result = JB_NOFRAME; - } else { - frame = get_frame(jb, now - jb->current); - if (frame) { //there is a frame - /* voice frame is late */ - if(frame->ts < jb->next_voice_time) { //late - jb_dbg("aL"); - jb->info.frames_late++; - frame_free(frame); - result = get_voice(jb, data, now, interpl); - } else { - jb_dbg("aP"); - /* normal case; return the frame, increment stuff */ - *data = frame->data; - frame->data = NULL; - jb->next_voice_time = frame->ts + frame->ms; - jb->cnt_successive_interp = 0; - frame_free(frame); - result = JB_OK; + jb_frame *frame; + int result; + + // * - difference is way off, reset + if (diff > jb->settings.max_diff || -diff > jb->settings.max_diff) { + jb_err("wakko diff in get_voicecase\n"); + reset(jb); //reset hist because the timestamps are wakko. + result = JB_NOFRAME; + //- diff > 0, we may need to grow + } + else if ((diff > 0) && + (now > (jb->last_adjustment + jb->settings.wait_grow) + || (now + jb->current + interpl) < get_next_framets(jb) ) ) { //grow + /* first try to grow */ + if (diff<interpl/2) { + jb_dbg("ag"); + jb->current +=diff; } - } else { // no frame, thus interpolate - jb->cnt_successive_interp++; - /* assume silence instead of continuing to interpolate */ - if (jb->settings.max_successive_interp && jb->cnt_successive_interp >= jb->settings.max_successive_interp) { - jb->info.silence = 1; - jb->silence_begin_ts = jb->next_voice_time; + else { + jb_dbg("aG"); + /* grow by interp frame len */ + jb->current += interpl; } - jb_dbg("aI"); - jb->next_voice_time += interpl; - result = JB_INTERP; - } + jb->last_adjustment = now; + result = get_voice(jb, data, now, interpl); + //- diff < 0, we may need to shrink } - } - return result; + else if ( (diff < 0) + && (now > (jb->last_adjustment + jb->settings.wait_shrink)) + && ((-diff) > jb->settings.extra_delay) ) { + /* now try to shrink + * if there is a frame shrink by frame length + * otherwise shrink by interpl + */ + jb->last_adjustment = now; + + frame = get_frame(jb, now - jb->current); + if(frame) { + jb_dbg("as"); + /* shrink by frame size we're throwing out */ + jb->info.frames_dropped++; + jb->current -= frame->ms; + frame_free(frame); + } + else { + jb_dbg("aS"); + /* shrink by interpl */ + jb->current -= interpl; + } + result = get_voice(jb, data, now, interpl); + } + else { + /* if it is not the time to play a result = JB_NOFRAME + * else We try to play a frame if a frame is available + * and not late it is played otherwise + * if available it is dropped and the next is tried + * last option is interpolating + */ + if (now - jb->current < jb->next_voice_time) { + jb_dbg("aN"); + result = JB_NOFRAME; + } + else { + frame = get_frame(jb, now - jb->current); + if (frame) { //there is a frame + /* voice frame is late */ + if(frame->ts < jb->next_voice_time) { //late + jb_dbg("aL"); + jb->info.frames_late++; + frame_free(frame); + result = get_voice(jb, data, now, interpl); + } + else { + jb_dbg("aP"); + /* normal case; return the frame, increment stuff */ + *data = frame->data; + frame->data = NULL; + jb->next_voice_time = frame->ts + frame->ms; + jb->cnt_successive_interp = 0; + frame_free(frame); + result = JB_OK; + } + } + else { // no frame, thus interpolate + jb->cnt_successive_interp++; + /* assume silence instead of continuing to interpolate */ + if (jb->settings.max_successive_interp && jb->cnt_successive_interp >= jb->settings.max_successive_interp) { + jb->info.silence = 1; + jb->silence_begin_ts = jb->next_voice_time; + } + jb_dbg("aI"); + jb->next_voice_time += interpl; + result = JB_INTERP; + } + } + } + return result; } /*********** - * if there are frames and next frame->ts is smaller or equal ts + * if there are frames and next frame->ts is smaller or equal ts * return type of next frame. * else return 0 */ -static int get_next_frametype(jitterbuffer *jb, long ts) +static int get_next_frametype(jitterbuffer *jb, long ts) { - jb_frame *frame; - int result; - - result = 0; - frame = jb->voiceframes; - if (frame && frame->ts <= ts) { - result = frame->type; - } - return result; + jb_frame *frame; + int result; + + result = 0; + frame = jb->voiceframes; + if (frame && frame->ts <= ts) { + result = frame->type; + } + return result; } @@ -974,62 +1006,64 @@ static int get_next_frametype(jitterbuffer *jb, long ts) * returns ts from next frame in jb->voiceframes * or returns LONG_MAX if there is no frame */ -static long get_next_framets(jitterbuffer *jb) +static long get_next_framets(jitterbuffer *jb) { - if (jb->voiceframes) { - return jb->voiceframes->ts; - } - return LONG_MAX; + if (jb->voiceframes) { + return jb->voiceframes->ts; + } + return LONG_MAX; } /*********** - * if there is a frame in jb->voiceframes and + * if there is a frame in jb->voiceframes and * has a timestamp smaller/equal to ts - * this frame will be returned and + * this frame will be returned and * removed from the queue */ -static jb_frame *get_frame(jitterbuffer *jb, long ts) +static jb_frame *get_frame(jitterbuffer *jb, long ts) { - jb_frame *frame; - - frame = jb->voiceframes; - if (frame && frame->ts <= ts) { - if(frame->next == frame) { - jb->voiceframes = NULL; - } else { - /* remove this frame */ - frame->prev->next = frame->next; - frame->next->prev = frame->prev; - jb->voiceframes = frame->next; - } - return frame; - } - return NULL; + jb_frame *frame; + + frame = jb->voiceframes; + if (frame && frame->ts <= ts) { + if(frame->next == frame) { + jb->voiceframes = NULL; + } + else { + /* remove this frame */ + frame->prev->next = frame->next; + frame->next->prev = frame->prev; + jb->voiceframes = frame->next; + } + return frame; + } + return NULL; } /*********** * if there is a frame in jb->voiceframes - * this frame will be unconditionally returned and + * this frame will be unconditionally returned and * removed from the queue */ -static jb_frame *get_all_frames(jitterbuffer *jb) +static jb_frame *get_all_frames(jitterbuffer *jb) { - jb_frame *frame; - - frame = jb->voiceframes; - if (frame) { - if(frame->next == frame) { - jb->voiceframes = NULL; - } else { - /* remove this frame */ - frame->prev->next = frame->next; - frame->next->prev = frame->prev; - jb->voiceframes = frame->next; - } - return frame; - } - return NULL; + jb_frame *frame; + + frame = jb->voiceframes; + if (frame) { + if(frame->next == frame) { + jb->voiceframes = NULL; + } + else { + /* remove this frame */ + frame->prev->next = frame->next; + frame->next->prev = frame->prev; + jb->voiceframes = frame->next; + } + return frame; + } + return NULL; } diff --git a/tinyDAV/src/audio/tdav_producer_audio.c b/tinyDAV/src/audio/tdav_producer_audio.c index 8c73c9f..1c7e779 100755 --- a/tinyDAV/src/audio/tdav_producer_audio.c +++ b/tinyDAV/src/audio/tdav_producer_audio.c @@ -2,19 +2,19 @@ * Copyright (C) 2010-2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -45,25 +45,25 @@ */ int tdav_producer_audio_init(tdav_producer_audio_t* self) { - int ret; - - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - /* base */ - if((ret = tmedia_producer_init(TMEDIA_PRODUCER(self)))){ - return ret; - } - - /* self (should be update by prepare() by using the codec's info)*/ - TMEDIA_PRODUCER(self)->audio.bits_per_sample = TDAV_PRODUCER_BITS_PER_SAMPLE_DEFAULT; - TMEDIA_PRODUCER(self)->audio.channels = TDAV_PRODUCER_CHANNELS_DEFAULT; - TMEDIA_PRODUCER(self)->audio.rate = TDAV_PRODUCER_RATE_DEFAULT; - TMEDIA_PRODUCER(self)->audio.ptime = TDAV_PRODUCER_PTIME_DEFAULT; - TMEDIA_PRODUCER(self)->audio.gain = TSK_MIN(tmedia_defaults_get_audio_producer_gain(), TDAV_PRODUCER_AUDIO_GAIN_MAX); - - return 0; + int ret; + + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + /* base */ + if((ret = tmedia_producer_init(TMEDIA_PRODUCER(self)))) { + return ret; + } + + /* self (should be update by prepare() by using the codec's info)*/ + TMEDIA_PRODUCER(self)->audio.bits_per_sample = TDAV_PRODUCER_BITS_PER_SAMPLE_DEFAULT; + TMEDIA_PRODUCER(self)->audio.channels = TDAV_PRODUCER_CHANNELS_DEFAULT; + TMEDIA_PRODUCER(self)->audio.rate = TDAV_PRODUCER_RATE_DEFAULT; + TMEDIA_PRODUCER(self)->audio.ptime = TDAV_PRODUCER_PTIME_DEFAULT; + TMEDIA_PRODUCER(self)->audio.gain = TSK_MIN(tmedia_defaults_get_audio_producer_gain(), TDAV_PRODUCER_AUDIO_GAIN_MAX); + + return 0; } /** @@ -77,57 +77,57 @@ int tdav_producer_audio_init(tdav_producer_audio_t* self) */ int tdav_producer_audio_cmp(const tsk_object_t* producer1, const tsk_object_t* producer2) { - int ret; - tsk_subsat_int32_ptr(producer1, producer2, &ret); - return ret; + int ret; + tsk_subsat_int32_ptr(producer1, producer2, &ret); + return ret; } int tdav_producer_audio_set(tdav_producer_audio_t* self, const tmedia_param_t* param) { - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(param->plugin_type == tmedia_ppt_producer){ - if(param->value_type == tmedia_pvt_int32){ - if(tsk_striequals(param->key, "gain")){ - int32_t gain = *((int32_t*)param->value); - if(gain<TDAV_PRODUCER_AUDIO_GAIN_MAX && gain>=0){ - TMEDIA_PRODUCER(self)->audio.gain = (uint8_t)gain; - TSK_DEBUG_INFO("audio producer gain=%u", gain); - } - else{ - TSK_DEBUG_ERROR("%u is invalid as gain value", gain); - return -2; - } - } - else if(tsk_striequals(param->key, "volume")){ - TMEDIA_PRODUCER(self)->audio.volume = TSK_TO_INT32((uint8_t*)param->value); - TMEDIA_PRODUCER(self)->audio.volume = TSK_CLAMP(0, TMEDIA_PRODUCER(self)->audio.volume, 100); - TSK_DEBUG_INFO("audio producer volume=%u", TMEDIA_PRODUCER(self)->audio.volume); - } - } - } - - return 0; + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(param->plugin_type == tmedia_ppt_producer) { + if(param->value_type == tmedia_pvt_int32) { + if(tsk_striequals(param->key, "gain")) { + int32_t gain = *((int32_t*)param->value); + if(gain<TDAV_PRODUCER_AUDIO_GAIN_MAX && gain>=0) { + TMEDIA_PRODUCER(self)->audio.gain = (uint8_t)gain; + TSK_DEBUG_INFO("audio producer gain=%u", gain); + } + else { + TSK_DEBUG_ERROR("%u is invalid as gain value", gain); + return -2; + } + } + else if(tsk_striequals(param->key, "volume")) { + TMEDIA_PRODUCER(self)->audio.volume = TSK_TO_INT32((uint8_t*)param->value); + TMEDIA_PRODUCER(self)->audio.volume = TSK_CLAMP(0, TMEDIA_PRODUCER(self)->audio.volume, 100); + TSK_DEBUG_INFO("audio producer volume=%u", TMEDIA_PRODUCER(self)->audio.volume); + } + } + } + + return 0; } /** Deinitialize a producer */ int tdav_producer_audio_deinit(tdav_producer_audio_t* self) { - int ret; + int ret; - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - /* base */ - if((ret = tmedia_producer_deinit(TMEDIA_PRODUCER(self)))){ - return ret; - } + /* base */ + if((ret = tmedia_producer_deinit(TMEDIA_PRODUCER(self)))) { + return ret; + } - return ret; + return ret; }
\ No newline at end of file diff --git a/tinyDAV/src/audio/tdav_session_audio.c b/tinyDAV/src/audio/tdav_session_audio.c index f12e801..31c9b34 100755 --- a/tinyDAV/src/audio/tdav_session_audio.c +++ b/tinyDAV/src/audio/tdav_session_audio.c @@ -51,14 +51,13 @@ static void _tdav_session_audio_apply_gain(void* buffer, int len, int bps, int g static tmedia_resampler_t* _tdav_session_audio_resampler_create(int32_t bytes_per_sample, uint32_t in_freq, uint32_t out_freq, uint32_t frame_duration, uint32_t in_channels, uint32_t out_channels, uint32_t quality, void** resampler_buffer, tsk_size_t *resampler_buffer_size); /* DTMF event object */ -typedef struct tdav_session_audio_dtmfe_s -{ - TSK_DECLARE_OBJECT; +typedef struct tdav_session_audio_dtmfe_s { + TSK_DECLARE_OBJECT; - tsk_timer_id_t timer_id; - trtp_rtp_packet_t* packet; + tsk_timer_id_t timer_id; + trtp_rtp_packet_t* packet; - const tdav_session_audio_t* session; + const tdav_session_audio_t* session; } tdav_session_audio_dtmfe_t; extern const tsk_object_def_t *tdav_session_audio_dtmfe_def_t; @@ -66,230 +65,230 @@ extern const tsk_object_def_t *tdav_session_audio_dtmfe_def_t; // RTP/RTCP callback (From the network to the consumer) static int tdav_session_audio_rtp_cb(const void* callback_data, const struct trtp_rtp_packet_s* packet) { - tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data; - tmedia_codec_t* codec = tsk_null; - tdav_session_av_t* base = (tdav_session_av_t*)callback_data; - int ret = -1; - - if (!audio || !packet || !packet->header) { - TSK_DEBUG_ERROR("Invalid parameter"); - goto bail; - } - - if (audio->is_started && base->consumer && base->consumer->is_started) { - tsk_size_t out_size = 0; - - // Find the codec to use to decode the RTP payload - if (!audio->decoder.codec || audio->decoder.payload_type != packet->header->payload_type) { - tsk_istr_t format; - TSK_OBJECT_SAFE_FREE(audio->decoder.codec); - tsk_itoa(packet->header->payload_type, &format); - if (!(audio->decoder.codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->neg_codecs, format)) || !audio->decoder.codec->plugin || !audio->decoder.codec->plugin->decode){ - TSK_DEBUG_ERROR("%s is not a valid payload for this session", format); - ret = -2; - goto bail; - } - audio->decoder.payload_type = packet->header->payload_type; - } - // ref() the codec to be able to use it short time after stop(SAFE_FREE(codec)) - if (!(codec = tsk_object_ref(TSK_OBJECT(audio->decoder.codec)))) { - TSK_DEBUG_ERROR("Failed to get decoder codec"); - goto bail; - } - - // Open codec if not already done - if (!TMEDIA_CODEC(codec)->opened) { - tsk_safeobj_lock(base); - if ((ret = tmedia_codec_open(codec))) { - tsk_safeobj_unlock(base); - TSK_DEBUG_ERROR("Failed to open [%s] codec", codec->plugin->desc); - TSK_OBJECT_SAFE_FREE(audio->decoder.codec); - goto bail; - } - tsk_safeobj_unlock(base); - } - // Decode data - out_size = codec->plugin->decode(codec, packet->payload.data, packet->payload.size, &audio->decoder.buffer, &audio->decoder.buffer_size, packet->header); - if (out_size && audio->is_started) { // check "is_started" again ...to be sure stop() not called by another thread - void* buffer = audio->decoder.buffer; - tsk_size_t size = out_size; - - // resample if needed - if ((base->consumer->audio.out.rate && base->consumer->audio.out.rate != codec->in.rate) || (base->consumer->audio.out.channels && base->consumer->audio.out.channels != TMEDIA_CODEC_AUDIO(codec)->in.channels)) { - tsk_size_t resampler_result_size = 0; - int bytesPerSample = (base->consumer->audio.bits_per_sample >> 3); - - if (!audio->decoder.resampler.instance) { - TSK_DEBUG_INFO("Create audio resampler(%s) for consumer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d", - codec->plugin->desc, - codec->in.rate, base->consumer->audio.out.rate, - TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels, - bytesPerSample); - audio->decoder.resampler.instance = _tdav_session_audio_resampler_create( - bytesPerSample, - codec->in.rate, base->consumer->audio.out.rate, - base->consumer->audio.ptime, - TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels, - TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY, - &audio->decoder.resampler.buffer, &audio->decoder.resampler.buffer_size - ); - } - if (!audio->decoder.resampler.instance) { - TSK_DEBUG_ERROR("No resampler to handle data"); - ret = -5; - goto bail; - } - if (!(resampler_result_size = tmedia_resampler_process(audio->decoder.resampler.instance, buffer, size / bytesPerSample, audio->decoder.resampler.buffer, audio->decoder.resampler.buffer_size / bytesPerSample))){ - TSK_DEBUG_ERROR("Failed to process audio resampler input buffer"); - ret = -6; - goto bail; - } - - buffer = audio->decoder.resampler.buffer; - size = audio->decoder.resampler.buffer_size; - } - - // adjust the gain - if (base->consumer->audio.gain) { - _tdav_session_audio_apply_gain(buffer, (int)size, base->consumer->audio.bits_per_sample, base->consumer->audio.gain); - } - // consume the frame - tmedia_consumer_consume(base->consumer, buffer, size, packet->header); - } - } - else { - TSK_DEBUG_INFO("Session audio not ready"); - } - - // everything is ok - ret = 0; + tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data; + tmedia_codec_t* codec = tsk_null; + tdav_session_av_t* base = (tdav_session_av_t*)callback_data; + int ret = -1; + + if (!audio || !packet || !packet->header) { + TSK_DEBUG_ERROR("Invalid parameter"); + goto bail; + } + + if (audio->is_started && base->consumer && base->consumer->is_started) { + tsk_size_t out_size = 0; + + // Find the codec to use to decode the RTP payload + if (!audio->decoder.codec || audio->decoder.payload_type != packet->header->payload_type) { + tsk_istr_t format; + TSK_OBJECT_SAFE_FREE(audio->decoder.codec); + tsk_itoa(packet->header->payload_type, &format); + if (!(audio->decoder.codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->neg_codecs, format)) || !audio->decoder.codec->plugin || !audio->decoder.codec->plugin->decode) { + TSK_DEBUG_ERROR("%s is not a valid payload for this session", format); + ret = -2; + goto bail; + } + audio->decoder.payload_type = packet->header->payload_type; + } + // ref() the codec to be able to use it short time after stop(SAFE_FREE(codec)) + if (!(codec = tsk_object_ref(TSK_OBJECT(audio->decoder.codec)))) { + TSK_DEBUG_ERROR("Failed to get decoder codec"); + goto bail; + } + + // Open codec if not already done + if (!TMEDIA_CODEC(codec)->opened) { + tsk_safeobj_lock(base); + if ((ret = tmedia_codec_open(codec))) { + tsk_safeobj_unlock(base); + TSK_DEBUG_ERROR("Failed to open [%s] codec", codec->plugin->desc); + TSK_OBJECT_SAFE_FREE(audio->decoder.codec); + goto bail; + } + tsk_safeobj_unlock(base); + } + // Decode data + out_size = codec->plugin->decode(codec, packet->payload.data, packet->payload.size, &audio->decoder.buffer, &audio->decoder.buffer_size, packet->header); + if (out_size && audio->is_started) { // check "is_started" again ...to be sure stop() not called by another thread + void* buffer = audio->decoder.buffer; + tsk_size_t size = out_size; + + // resample if needed + if ((base->consumer->audio.out.rate && base->consumer->audio.out.rate != codec->in.rate) || (base->consumer->audio.out.channels && base->consumer->audio.out.channels != TMEDIA_CODEC_AUDIO(codec)->in.channels)) { + tsk_size_t resampler_result_size = 0; + int bytesPerSample = (base->consumer->audio.bits_per_sample >> 3); + + if (!audio->decoder.resampler.instance) { + TSK_DEBUG_INFO("Create audio resampler(%s) for consumer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d", + codec->plugin->desc, + codec->in.rate, base->consumer->audio.out.rate, + TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels, + bytesPerSample); + audio->decoder.resampler.instance = _tdav_session_audio_resampler_create( + bytesPerSample, + codec->in.rate, base->consumer->audio.out.rate, + base->consumer->audio.ptime, + TMEDIA_CODEC_AUDIO(codec)->in.channels, base->consumer->audio.out.channels, + TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY, + &audio->decoder.resampler.buffer, &audio->decoder.resampler.buffer_size + ); + } + if (!audio->decoder.resampler.instance) { + TSK_DEBUG_ERROR("No resampler to handle data"); + ret = -5; + goto bail; + } + if (!(resampler_result_size = tmedia_resampler_process(audio->decoder.resampler.instance, buffer, size / bytesPerSample, audio->decoder.resampler.buffer, audio->decoder.resampler.buffer_size / bytesPerSample))) { + TSK_DEBUG_ERROR("Failed to process audio resampler input buffer"); + ret = -6; + goto bail; + } + + buffer = audio->decoder.resampler.buffer; + size = audio->decoder.resampler.buffer_size; + } + + // adjust the gain + if (base->consumer->audio.gain) { + _tdav_session_audio_apply_gain(buffer, (int)size, base->consumer->audio.bits_per_sample, base->consumer->audio.gain); + } + // consume the frame + tmedia_consumer_consume(base->consumer, buffer, size, packet->header); + } + } + else { + TSK_DEBUG_INFO("Session audio not ready"); + } + + // everything is ok + ret = 0; bail: - tsk_object_unref(TSK_OBJECT(codec)); - return ret; + tsk_object_unref(TSK_OBJECT(codec)); + return ret; } // Producer callback (From the producer to the network). Will encode() data before sending static int tdav_session_audio_producer_enc_cb(const void* callback_data, const void* buffer, tsk_size_t size) { - int ret = 0; - - tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data; - tdav_session_av_t* base = (tdav_session_av_t*)callback_data; - - if (!audio) { - TSK_DEBUG_ERROR("Null session"); - return 0; - } - - // do nothing if session is held - // when the session is held the end user will get feedback he also has possibilities to put the consumer and producer on pause - if (TMEDIA_SESSION(audio)->lo_held) { - return 0; - } - - // get best negotiated codec if not already done - // the encoder codec could be null when session is renegotiated without re-starting (e.g. hold/resume) - if (!audio->encoder.codec) { - const tmedia_codec_t* codec; - tsk_safeobj_lock(base); - if (!(codec = tdav_session_av_get_best_neg_codec(base))) { - TSK_DEBUG_ERROR("No codec matched"); - tsk_safeobj_unlock(base); - return -2; - } - audio->encoder.codec = tsk_object_ref(TSK_OBJECT(codec)); - tsk_safeobj_unlock(base); - } - - if (audio->is_started && base->rtp_manager && base->rtp_manager->is_started) { - /* encode */ - tsk_size_t out_size = 0; - - // Open codec if not already done - if (!audio->encoder.codec->opened) { - tsk_safeobj_lock(base); - if ((ret = tmedia_codec_open(audio->encoder.codec))) { - tsk_safeobj_unlock(base); - TSK_DEBUG_ERROR("Failed to open [%s] codec", audio->encoder.codec->plugin->desc); - return -4; - } - tsk_safeobj_unlock(base); - } - // check if we're sending DTMF or not - if (audio->is_sending_dtmf_events) { - if (base->rtp_manager) { - // increment the timestamp - base->rtp_manager->rtp.timestamp += TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec)/*duration*/; - } - TSK_DEBUG_INFO("Skiping audio frame as we're sending DTMF..."); - return 0; - } - - // resample if needed - if (base->producer->audio.rate != audio->encoder.codec->out.rate || base->producer->audio.channels != TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels){ - tsk_size_t resampler_result_size = 0; - int bytesPerSample = (base->producer->audio.bits_per_sample >> 3); - - if (!audio->encoder.resampler.instance){ - TSK_DEBUG_INFO("Create audio resampler(%s) for producer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d", - audio->encoder.codec->plugin->desc, - base->producer->audio.rate, audio->encoder.codec->out.rate, - base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels, - bytesPerSample); - audio->encoder.resampler.instance = _tdav_session_audio_resampler_create( - bytesPerSample, - base->producer->audio.rate, audio->encoder.codec->out.rate, - base->producer->audio.ptime, - base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels, - TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY, - &audio->encoder.resampler.buffer, &audio->encoder.resampler.buffer_size - ); - } - if (!audio->encoder.resampler.instance){ - TSK_DEBUG_ERROR("No resampler to handle data"); - ret = -1; - goto done; - } - if (!(resampler_result_size = tmedia_resampler_process(audio->encoder.resampler.instance, buffer, size / bytesPerSample, audio->encoder.resampler.buffer, audio->encoder.resampler.buffer_size / bytesPerSample))){ - TSK_DEBUG_ERROR("Failed to process audio resampler input buffer"); - ret = -1; - goto done; - } - - buffer = audio->encoder.resampler.buffer; - size = audio->encoder.resampler.buffer_size; - } - - // Denoise (VAD, AGC, Noise suppression, ...) - // Must be done after resampling - if (audio->denoise){ - tsk_bool_t silence_or_noise = tsk_false; - if (audio->denoise->echo_supp_enabled){ - ret = tmedia_denoise_process_record(TMEDIA_DENOISE(audio->denoise), (void*)buffer, (uint32_t)size, &silence_or_noise); - } - } - // adjust the gain - // Must be done after resampling - if (base->producer->audio.gain){ - _tdav_session_audio_apply_gain((void*)buffer, (int)size, base->producer->audio.bits_per_sample, base->producer->audio.gain); - } - - // Encode data - if ((audio->encoder.codec = tsk_object_ref(audio->encoder.codec))){ /* Thread safeness (SIP reINVITE or UPDATE could update the encoder) */ - out_size = audio->encoder.codec->plugin->encode(audio->encoder.codec, buffer, size, &audio->encoder.buffer, &audio->encoder.buffer_size); - if (out_size){ - trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size, TMEDIA_CODEC_FRAME_DURATION_AUDIO_ENCODING(audio->encoder.codec), tsk_false/*Marker*/, tsk_true/*lastPacket*/); - } - tsk_object_unref(audio->encoder.codec); - } - else{ - TSK_DEBUG_WARN("No encoder"); - } - } + int ret = 0; + + tdav_session_audio_t* audio = (tdav_session_audio_t*)callback_data; + tdav_session_av_t* base = (tdav_session_av_t*)callback_data; + + if (!audio) { + TSK_DEBUG_ERROR("Null session"); + return 0; + } + + // do nothing if session is held + // when the session is held the end user will get feedback he also has possibilities to put the consumer and producer on pause + if (TMEDIA_SESSION(audio)->lo_held) { + return 0; + } + + // get best negotiated codec if not already done + // the encoder codec could be null when session is renegotiated without re-starting (e.g. hold/resume) + if (!audio->encoder.codec) { + const tmedia_codec_t* codec; + tsk_safeobj_lock(base); + if (!(codec = tdav_session_av_get_best_neg_codec(base))) { + TSK_DEBUG_ERROR("No codec matched"); + tsk_safeobj_unlock(base); + return -2; + } + audio->encoder.codec = tsk_object_ref(TSK_OBJECT(codec)); + tsk_safeobj_unlock(base); + } + + if (audio->is_started && base->rtp_manager && base->rtp_manager->is_started) { + /* encode */ + tsk_size_t out_size = 0; + + // Open codec if not already done + if (!audio->encoder.codec->opened) { + tsk_safeobj_lock(base); + if ((ret = tmedia_codec_open(audio->encoder.codec))) { + tsk_safeobj_unlock(base); + TSK_DEBUG_ERROR("Failed to open [%s] codec", audio->encoder.codec->plugin->desc); + return -4; + } + tsk_safeobj_unlock(base); + } + // check if we're sending DTMF or not + if (audio->is_sending_dtmf_events) { + if (base->rtp_manager) { + // increment the timestamp + base->rtp_manager->rtp.timestamp += TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec)/*duration*/; + } + TSK_DEBUG_INFO("Skiping audio frame as we're sending DTMF..."); + return 0; + } + + // resample if needed + if (base->producer->audio.rate != audio->encoder.codec->out.rate || base->producer->audio.channels != TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels) { + tsk_size_t resampler_result_size = 0; + int bytesPerSample = (base->producer->audio.bits_per_sample >> 3); + + if (!audio->encoder.resampler.instance) { + TSK_DEBUG_INFO("Create audio resampler(%s) for producer: rate=%d->%d, channels=%d->%d, bytesPerSample=%d", + audio->encoder.codec->plugin->desc, + base->producer->audio.rate, audio->encoder.codec->out.rate, + base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels, + bytesPerSample); + audio->encoder.resampler.instance = _tdav_session_audio_resampler_create( + bytesPerSample, + base->producer->audio.rate, audio->encoder.codec->out.rate, + base->producer->audio.ptime, + base->producer->audio.channels, TMEDIA_CODEC_AUDIO(audio->encoder.codec)->out.channels, + TDAV_AUDIO_RESAMPLER_DEFAULT_QUALITY, + &audio->encoder.resampler.buffer, &audio->encoder.resampler.buffer_size + ); + } + if (!audio->encoder.resampler.instance) { + TSK_DEBUG_ERROR("No resampler to handle data"); + ret = -1; + goto done; + } + if (!(resampler_result_size = tmedia_resampler_process(audio->encoder.resampler.instance, buffer, size / bytesPerSample, audio->encoder.resampler.buffer, audio->encoder.resampler.buffer_size / bytesPerSample))) { + TSK_DEBUG_ERROR("Failed to process audio resampler input buffer"); + ret = -1; + goto done; + } + + buffer = audio->encoder.resampler.buffer; + size = audio->encoder.resampler.buffer_size; + } + + // Denoise (VAD, AGC, Noise suppression, ...) + // Must be done after resampling + if (audio->denoise) { + tsk_bool_t silence_or_noise = tsk_false; + if (audio->denoise->echo_supp_enabled) { + ret = tmedia_denoise_process_record(TMEDIA_DENOISE(audio->denoise), (void*)buffer, (uint32_t)size, &silence_or_noise); + } + } + // adjust the gain + // Must be done after resampling + if (base->producer->audio.gain) { + _tdav_session_audio_apply_gain((void*)buffer, (int)size, base->producer->audio.bits_per_sample, base->producer->audio.gain); + } + + // Encode data + if ((audio->encoder.codec = tsk_object_ref(audio->encoder.codec))) { /* Thread safeness (SIP reINVITE or UPDATE could update the encoder) */ + out_size = audio->encoder.codec->plugin->encode(audio->encoder.codec, buffer, size, &audio->encoder.buffer, &audio->encoder.buffer_size); + if (out_size) { + trtp_manager_send_rtp(base->rtp_manager, audio->encoder.buffer, out_size, TMEDIA_CODEC_FRAME_DURATION_AUDIO_ENCODING(audio->encoder.codec), tsk_false/*Marker*/, tsk_true/*lastPacket*/); + } + tsk_object_unref(audio->encoder.codec); + } + else { + TSK_DEBUG_WARN("No encoder"); + } + } done: - return ret; + return ret; } @@ -297,512 +296,514 @@ done: static int tdav_session_audio_set(tmedia_session_t* self, const tmedia_param_t* param) { - int ret = 0; - tdav_session_audio_t* audio; - - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if (tdav_session_av_set(TDAV_SESSION_AV(self), param) == tsk_true){ - return 0; - } - - audio = (tdav_session_audio_t*)self; - - if (param->plugin_type == tmedia_ppt_consumer){ - TSK_DEBUG_ERROR("Not expected consumer_set(%s)", param->key); - } - else if (param->plugin_type == tmedia_ppt_producer){ - TSK_DEBUG_ERROR("Not expected producer_set(%s)", param->key); - } - else{ - if (param->value_type == tmedia_pvt_int32){ - if (tsk_striequals(param->key, "echo-supp")){ - if (audio->denoise){ - audio->denoise->echo_supp_enabled = (TSK_TO_INT32((uint8_t*)param->value) != 0); - } - } - else if (tsk_striequals(param->key, "echo-tail")){ - if (audio->denoise){ - return tmedia_denoise_set(audio->denoise, param); - } - } - } - } - - return ret; + int ret = 0; + tdav_session_audio_t* audio; + + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if (tdav_session_av_set(TDAV_SESSION_AV(self), param) == tsk_true) { + return 0; + } + + audio = (tdav_session_audio_t*)self; + + if (param->plugin_type == tmedia_ppt_consumer) { + TSK_DEBUG_ERROR("Not expected consumer_set(%s)", param->key); + } + else if (param->plugin_type == tmedia_ppt_producer) { + TSK_DEBUG_ERROR("Not expected producer_set(%s)", param->key); + } + else { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "echo-supp")) { + if (audio->denoise) { + audio->denoise->echo_supp_enabled = (TSK_TO_INT32((uint8_t*)param->value) != 0); + } + } + else if (tsk_striequals(param->key, "echo-tail")) { + if (audio->denoise) { + return tmedia_denoise_set(audio->denoise, param); + } + } + } + } + + return ret; } static int tdav_session_audio_get(tmedia_session_t* self, tmedia_param_t* param) { - if (!self || !param){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - // try with the base class to see if this option is supported or not - if (tdav_session_av_get(TDAV_SESSION_AV(self), param) == tsk_true){ - return 0; - } - else { - // the codec information is held by the session even if the user is authorized to request it for the consumer/producer - if (param->value_type == tmedia_pvt_pobject){ - if (param->plugin_type == tmedia_ppt_consumer){ - TSK_DEBUG_ERROR("Not implemented"); - return -4; - } - else if (param->plugin_type == tmedia_ppt_producer){ - if (tsk_striequals("codec", param->key)) { - const tmedia_codec_t* codec; - if (!(codec = TDAV_SESSION_AUDIO(self)->encoder.codec)){ - codec = tdav_session_av_get_best_neg_codec((const tdav_session_av_t*)self); // up to the caller to release the object - } - *((tsk_object_t**)param->value) = tsk_object_ref(TSK_OBJECT(codec)); - return 0; - } - } - else if (param->plugin_type == tmedia_ppt_session) { - if (tsk_striequals(param->key, "codec-encoder")) { - *((tsk_object_t**)param->value) = tsk_object_ref(TDAV_SESSION_AUDIO(self)->encoder.codec); // up to the caller to release the object - return 0; - } - } - } - } - - TSK_DEBUG_WARN("This session doesn't support get(%s)", param->key); - return -2; + if (!self || !param) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + // try with the base class to see if this option is supported or not + if (tdav_session_av_get(TDAV_SESSION_AV(self), param) == tsk_true) { + return 0; + } + else { + // the codec information is held by the session even if the user is authorized to request it for the consumer/producer + if (param->value_type == tmedia_pvt_pobject) { + if (param->plugin_type == tmedia_ppt_consumer) { + TSK_DEBUG_ERROR("Not implemented"); + return -4; + } + else if (param->plugin_type == tmedia_ppt_producer) { + if (tsk_striequals("codec", param->key)) { + const tmedia_codec_t* codec; + if (!(codec = TDAV_SESSION_AUDIO(self)->encoder.codec)) { + codec = tdav_session_av_get_best_neg_codec((const tdav_session_av_t*)self); // up to the caller to release the object + } + *((tsk_object_t**)param->value) = tsk_object_ref(TSK_OBJECT(codec)); + return 0; + } + } + else if (param->plugin_type == tmedia_ppt_session) { + if (tsk_striequals(param->key, "codec-encoder")) { + *((tsk_object_t**)param->value) = tsk_object_ref(TDAV_SESSION_AUDIO(self)->encoder.codec); // up to the caller to release the object + return 0; + } + } + } + } + + TSK_DEBUG_WARN("This session doesn't support get(%s)", param->key); + return -2; } static int tdav_session_audio_prepare(tmedia_session_t* self) { - tdav_session_av_t* base = (tdav_session_av_t*)(self); - int ret; + tdav_session_av_t* base = (tdav_session_av_t*)(self); + int ret; - if ((ret = tdav_session_av_prepare(base))){ - TSK_DEBUG_ERROR("tdav_session_av_prepare(audio) failed"); - return ret; - } + if ((ret = tdav_session_av_prepare(base))) { + TSK_DEBUG_ERROR("tdav_session_av_prepare(audio) failed"); + return ret; + } - if (base->rtp_manager){ - ret = trtp_manager_set_rtp_callback(base->rtp_manager, tdav_session_audio_rtp_cb, base); - } + if (base->rtp_manager) { + ret = trtp_manager_set_rtp_callback(base->rtp_manager, tdav_session_audio_rtp_cb, base); + } - return ret; + return ret; } static int tdav_session_audio_start(tmedia_session_t* self) { - int ret; - tdav_session_audio_t* audio; - const tmedia_codec_t* codec; - tdav_session_av_t* base; - - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - audio = (tdav_session_audio_t*)self; - base = (tdav_session_av_t*)self; - - if (audio->is_started) { - TSK_DEBUG_INFO("Audio session already started"); - return 0; - } - - if (!(codec = tdav_session_av_get_best_neg_codec(base))){ - TSK_DEBUG_ERROR("No codec matched"); - return -2; - } - - TSK_OBJECT_SAFE_FREE(audio->encoder.codec); - audio->encoder.codec = tsk_object_ref((tsk_object_t*)codec); - - if ((ret = tdav_session_av_start(base, codec))){ - TSK_DEBUG_ERROR("tdav_session_av_start(audio) failed"); - return ret; - } - - if (base->rtp_manager){ - /* Denoise (AEC, Noise Suppression, AGC) - * tmedia_denoise_process_record() is called after resampling and before encoding which means sampling rate is equal to codec's rate - * tmedia_denoise_echo_playback() is called before playback which means sampling rate is equal to consumer's rate - */ - if (audio->denoise){ - uint32_t record_frame_size_samples = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec); - uint32_t record_sampling_rate = TMEDIA_CODEC_RATE_ENCODING(audio->encoder.codec); - uint32_t record_channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(audio->encoder.codec); - - uint32_t playback_frame_size_samples = (base->consumer && base->consumer->audio.ptime && base->consumer->audio.out.rate && base->consumer->audio.out.channels) - ? ((base->consumer->audio.ptime * base->consumer->audio.out.rate) / 1000) * base->consumer->audio.out.channels - : TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_DECODING(audio->encoder.codec); - uint32_t playback_sampling_rate = (base->consumer && base->consumer->audio.out.rate) - ? base->consumer->audio.out.rate - : TMEDIA_CODEC_RATE_DECODING(audio->encoder.codec); - uint32_t playback_channels = (base->consumer && base->consumer->audio.out.channels) - ? base->consumer->audio.out.channels - : TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(audio->encoder.codec); - - TSK_DEBUG_INFO("Audio denoiser to be opened(record_frame_size_samples=%u, record_sampling_rate=%u, record_channels=%u, playback_frame_size_samples=%u, playback_sampling_rate=%u, playback_channels=%u)", - record_frame_size_samples, record_sampling_rate, record_channels, playback_frame_size_samples, playback_sampling_rate, playback_channels); - - // close() - tmedia_denoise_close(audio->denoise); - // open() with new values - tmedia_denoise_open(audio->denoise, - record_frame_size_samples, record_sampling_rate, TSK_CLAMP(1, record_channels, 2), - playback_frame_size_samples, playback_sampling_rate, TSK_CLAMP(1, playback_channels, 2)); - } - } - - audio->is_started = (ret == 0); - - return ret; + int ret; + tdav_session_audio_t* audio; + const tmedia_codec_t* codec; + tdav_session_av_t* base; + + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + audio = (tdav_session_audio_t*)self; + base = (tdav_session_av_t*)self; + + if (audio->is_started) { + TSK_DEBUG_INFO("Audio session already started"); + return 0; + } + + if (!(codec = tdav_session_av_get_best_neg_codec(base))) { + TSK_DEBUG_ERROR("No codec matched"); + return -2; + } + + TSK_OBJECT_SAFE_FREE(audio->encoder.codec); + audio->encoder.codec = tsk_object_ref((tsk_object_t*)codec); + + if ((ret = tdav_session_av_start(base, codec))) { + TSK_DEBUG_ERROR("tdav_session_av_start(audio) failed"); + return ret; + } + + if (base->rtp_manager) { + /* Denoise (AEC, Noise Suppression, AGC) + * tmedia_denoise_process_record() is called after resampling and before encoding which means sampling rate is equal to codec's rate + * tmedia_denoise_echo_playback() is called before playback which means sampling rate is equal to consumer's rate + */ + if (audio->denoise) { + uint32_t record_frame_size_samples = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec); + uint32_t record_sampling_rate = TMEDIA_CODEC_RATE_ENCODING(audio->encoder.codec); + uint32_t record_channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(audio->encoder.codec); + + uint32_t playback_frame_size_samples = (base->consumer && base->consumer->audio.ptime && base->consumer->audio.out.rate && base->consumer->audio.out.channels) + ? ((base->consumer->audio.ptime * base->consumer->audio.out.rate) / 1000) * base->consumer->audio.out.channels + : TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_DECODING(audio->encoder.codec); + uint32_t playback_sampling_rate = (base->consumer && base->consumer->audio.out.rate) + ? base->consumer->audio.out.rate + : TMEDIA_CODEC_RATE_DECODING(audio->encoder.codec); + uint32_t playback_channels = (base->consumer && base->consumer->audio.out.channels) + ? base->consumer->audio.out.channels + : TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(audio->encoder.codec); + + TSK_DEBUG_INFO("Audio denoiser to be opened(record_frame_size_samples=%u, record_sampling_rate=%u, record_channels=%u, playback_frame_size_samples=%u, playback_sampling_rate=%u, playback_channels=%u)", + record_frame_size_samples, record_sampling_rate, record_channels, playback_frame_size_samples, playback_sampling_rate, playback_channels); + + // close() + tmedia_denoise_close(audio->denoise); + // open() with new values + tmedia_denoise_open(audio->denoise, + record_frame_size_samples, record_sampling_rate, TSK_CLAMP(1, record_channels, 2), + playback_frame_size_samples, playback_sampling_rate, TSK_CLAMP(1, playback_channels, 2)); + } + } + + audio->is_started = (ret == 0); + + return ret; } static int tdav_session_audio_stop(tmedia_session_t* self) { - tdav_session_audio_t* audio = TDAV_SESSION_AUDIO(self); - tdav_session_av_t* base = TDAV_SESSION_AV(self); - int ret = tdav_session_av_stop(base); - audio->is_started = tsk_false; - TSK_OBJECT_SAFE_FREE(audio->encoder.codec); - TSK_OBJECT_SAFE_FREE(audio->decoder.codec); - - // close the jitter buffer and denoiser to be sure it will be reopened and reinitialized if reINVITE or UPDATE - // this is a "must" when the initial and updated sessions use codecs with different rate - if (audio->jitterbuffer && audio->jitterbuffer->opened) { - ret = tmedia_jitterbuffer_close(audio->jitterbuffer); - } - if (audio->denoise && audio->denoise->opened) { - ret = tmedia_denoise_close(audio->denoise); - } - return ret; + tdav_session_audio_t* audio = TDAV_SESSION_AUDIO(self); + tdav_session_av_t* base = TDAV_SESSION_AV(self); + int ret = tdav_session_av_stop(base); + audio->is_started = tsk_false; + TSK_OBJECT_SAFE_FREE(audio->encoder.codec); + TSK_OBJECT_SAFE_FREE(audio->decoder.codec); + + // close the jitter buffer and denoiser to be sure it will be reopened and reinitialized if reINVITE or UPDATE + // this is a "must" when the initial and updated sessions use codecs with different rate + if (audio->jitterbuffer && audio->jitterbuffer->opened) { + ret = tmedia_jitterbuffer_close(audio->jitterbuffer); + } + if (audio->denoise && audio->denoise->opened) { + ret = tmedia_denoise_close(audio->denoise); + } + return ret; } static int tdav_session_audio_send_dtmf(tmedia_session_t* self, uint8_t event) { - tdav_session_audio_t* audio; - tdav_session_av_t* base; - tmedia_codec_t* codec; - int ret, rate = 8000, ptime = 20; - uint16_t duration; - tdav_session_audio_dtmfe_t *dtmfe, *copy; - int format = 101; - - if (!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - audio = (tdav_session_audio_t*)self; - base = (tdav_session_av_t*)self; - - // Find the DTMF codec to use to use the RTP payload - if ((codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->codecs, TMEDIA_CODEC_FORMAT_DTMF))){ - rate = (int)codec->out.rate; - format = atoi(codec->neg_format ? codec->neg_format : codec->format); - TSK_OBJECT_SAFE_FREE(codec); - } - - /* do we have an RTP manager? */ - if (!base->rtp_manager){ - TSK_DEBUG_ERROR("No RTP manager associated to this session"); - return -2; - } - - /* Create Events list */ - if (!audio->dtmf_events){ - audio->dtmf_events = tsk_list_create(); - } - - /* Create global reference to the timer manager */ - if (!audio->timer.handle_mgr_global){ - if (!(audio->timer.handle_mgr_global = tsk_timer_mgr_global_ref())){ - TSK_DEBUG_ERROR("Failed to create Global Timer Manager"); - return -3; - } - } - - /* Start the timer manager */ - if (!audio->timer.started){ - if ((ret = tsk_timer_manager_start(audio->timer.handle_mgr_global))){ - TSK_DEBUG_ERROR("Failed to start Global Timer Manager"); - return ret; - } - audio->timer.started = tsk_true; - } - - - /* RFC 4733 - 5. Examples - - +-------+-----------+------+--------+------+--------+--------+------+ - | Time | Event | M | Time- | Seq | Event | Dura- | E | - | (ms) | | bit | stamp | No | Code | tion | bit | - +-------+-----------+------+--------+------+--------+--------+------+ - | 0 | "9" | | | | | | | - | | starts | | | | | | | - | 50 | RTP | "1" | 0 | 1 | 9 | 400 | "0" | - | | packet 1 | | | | | | | - | | sent | | | | | | | - | 100 | RTP | "0" | 0 | 2 | 9 | 800 | "0" | - | | packet 2 | | | | | | | - | | sent | | | | | | | - | 150 | RTP | "0" | 0 | 3 | 9 | 1200 | "0" | - | | packet 3 | | | | | | | - | | sent | | | | | | | - | 200 | RTP | "0" | 0 | 4 | 9 | 1600 | "0" | - | | packet 4 | | | | | | | - | | sent | | | | | | | - | 200 | "9" ends | | | | | | | - | 250 | RTP | "0" | 0 | 5 | 9 | 1600 | "1" | - | | packet 4 | | | | | | | - | | first | | | | | | | - | | retrans- | | | | | | | - | | mission | | | | | | | - | 300 | RTP | "0" | 0 | 6 | 9 | 1600 | "1" | - | | packet 4 | | | | | | | - | | second | | | | | | | - | | retrans- | | | | | | | - | | mission | | | | | | | - ===================================================================== - | 880 | First "1" | | | | | | | - | | starts | | | | | | | - | 930 | RTP | "1" | 7040 | 7 | 1 | 400 | "0" | - | | packet 5 | | | | | | | - | | sent | | | | | | | - */ - - // ref()(thread safeness) - audio = tsk_object_ref(audio); - - // says we're sending DTMF digits to avoid mixing with audio (SRTP won't let this happen because of senquence numbers) - // flag will be turned OFF when the list is empty - audio->is_sending_dtmf_events = tsk_true; - - duration = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec); - - // lock() list - tsk_list_lock(audio->dtmf_events); - - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 1, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_true, tsk_false); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 0, _tdav_session_audio_dtmfe_timercb, copy); - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 2, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 1, _tdav_session_audio_dtmfe_timercb, copy); - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 3, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 2, _tdav_session_audio_dtmfe_timercb, copy); - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 3, _tdav_session_audio_dtmfe_timercb, copy); - - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 4, _tdav_session_audio_dtmfe_timercb, copy); - copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true); - tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); - tsk_timer_mgr_global_schedule(ptime * 5, _tdav_session_audio_dtmfe_timercb, copy); - - // unlock() list - tsk_list_unlock(audio->dtmf_events); - - // increment timestamp - base->rtp_manager->rtp.timestamp += duration; - - // unref()(thread safeness) - audio = tsk_object_unref(audio); - - return 0; + tdav_session_audio_t* audio; + tdav_session_av_t* base; + tmedia_codec_t* codec; + int ret, rate = 8000, ptime = 20; + uint16_t duration; + tdav_session_audio_dtmfe_t *dtmfe, *copy; + int format = 101; + + if (!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + audio = (tdav_session_audio_t*)self; + base = (tdav_session_av_t*)self; + + // Find the DTMF codec to use to use the RTP payload + if ((codec = tmedia_codec_find_by_format(TMEDIA_SESSION(audio)->codecs, TMEDIA_CODEC_FORMAT_DTMF))) { + rate = (int)codec->out.rate; + format = atoi(codec->neg_format ? codec->neg_format : codec->format); + TSK_OBJECT_SAFE_FREE(codec); + } + + /* do we have an RTP manager? */ + if (!base->rtp_manager) { + TSK_DEBUG_ERROR("No RTP manager associated to this session"); + return -2; + } + + /* Create Events list */ + if (!audio->dtmf_events) { + audio->dtmf_events = tsk_list_create(); + } + + /* Create global reference to the timer manager */ + if (!audio->timer.handle_mgr_global) { + if (!(audio->timer.handle_mgr_global = tsk_timer_mgr_global_ref())) { + TSK_DEBUG_ERROR("Failed to create Global Timer Manager"); + return -3; + } + } + + /* Start the timer manager */ + if (!audio->timer.started) { + if ((ret = tsk_timer_manager_start(audio->timer.handle_mgr_global))) { + TSK_DEBUG_ERROR("Failed to start Global Timer Manager"); + return ret; + } + audio->timer.started = tsk_true; + } + + + /* RFC 4733 - 5. Examples + + +-------+-----------+------+--------+------+--------+--------+------+ + | Time | Event | M | Time- | Seq | Event | Dura- | E | + | (ms) | | bit | stamp | No | Code | tion | bit | + +-------+-----------+------+--------+------+--------+--------+------+ + | 0 | "9" | | | | | | | + | | starts | | | | | | | + | 50 | RTP | "1" | 0 | 1 | 9 | 400 | "0" | + | | packet 1 | | | | | | | + | | sent | | | | | | | + | 100 | RTP | "0" | 0 | 2 | 9 | 800 | "0" | + | | packet 2 | | | | | | | + | | sent | | | | | | | + | 150 | RTP | "0" | 0 | 3 | 9 | 1200 | "0" | + | | packet 3 | | | | | | | + | | sent | | | | | | | + | 200 | RTP | "0" | 0 | 4 | 9 | 1600 | "0" | + | | packet 4 | | | | | | | + | | sent | | | | | | | + | 200 | "9" ends | | | | | | | + | 250 | RTP | "0" | 0 | 5 | 9 | 1600 | "1" | + | | packet 4 | | | | | | | + | | first | | | | | | | + | | retrans- | | | | | | | + | | mission | | | | | | | + | 300 | RTP | "0" | 0 | 6 | 9 | 1600 | "1" | + | | packet 4 | | | | | | | + | | second | | | | | | | + | | retrans- | | | | | | | + | | mission | | | | | | | + ===================================================================== + | 880 | First "1" | | | | | | | + | | starts | | | | | | | + | 930 | RTP | "1" | 7040 | 7 | 1 | 400 | "0" | + | | packet 5 | | | | | | | + | | sent | | | | | | | + */ + + // ref()(thread safeness) + audio = tsk_object_ref(audio); + + // says we're sending DTMF digits to avoid mixing with audio (SRTP won't let this happen because of senquence numbers) + // flag will be turned OFF when the list is empty + audio->is_sending_dtmf_events = tsk_true; + + duration = TMEDIA_CODEC_PCM_FRAME_SIZE_AUDIO_ENCODING(audio->encoder.codec); + + // lock() list + tsk_list_lock(audio->dtmf_events); + + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 1, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_true, tsk_false); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 0, _tdav_session_audio_dtmfe_timercb, copy); + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 2, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 1, _tdav_session_audio_dtmfe_timercb, copy); + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 3, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 2, _tdav_session_audio_dtmfe_timercb, copy); + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_false); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 3, _tdav_session_audio_dtmfe_timercb, copy); + + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 4, _tdav_session_audio_dtmfe_timercb, copy); + copy = dtmfe = _tdav_session_audio_dtmfe_create(audio, event, duration * 4, ++base->rtp_manager->rtp.seq_num, base->rtp_manager->rtp.timestamp, (uint8_t)format, tsk_false, tsk_true); + tsk_list_push_back_data(audio->dtmf_events, (void**)&dtmfe); + tsk_timer_mgr_global_schedule(ptime * 5, _tdav_session_audio_dtmfe_timercb, copy); + + // unlock() list + tsk_list_unlock(audio->dtmf_events); + + // increment timestamp + base->rtp_manager->rtp.timestamp += duration; + + // unref()(thread safeness) + audio = tsk_object_unref(audio); + + return 0; } static int tdav_session_audio_pause(tmedia_session_t* self) { - return tdav_session_av_pause(TDAV_SESSION_AV(self)); + return tdav_session_av_pause(TDAV_SESSION_AV(self)); } static const tsdp_header_M_t* tdav_session_audio_get_lo(tmedia_session_t* self) { - tsk_bool_t updated = tsk_false; - const tsdp_header_M_t* ret; - tdav_session_av_t* base = TDAV_SESSION_AV(self); + tsk_bool_t updated = tsk_false; + const tsdp_header_M_t* ret; + tdav_session_av_t* base = TDAV_SESSION_AV(self); - if (!(ret = tdav_session_av_get_lo(base, &updated))){ - TSK_DEBUG_ERROR("tdav_session_av_get_lo(audio) failed"); - return tsk_null; - } + if (!(ret = tdav_session_av_get_lo(base, &updated))) { + TSK_DEBUG_ERROR("tdav_session_av_get_lo(audio) failed"); + return tsk_null; + } - if (updated){ - tsk_safeobj_lock(base); - TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec); - tsk_safeobj_unlock(base); - } + if (updated) { + tsk_safeobj_lock(base); + TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec); + tsk_safeobj_unlock(base); + } - return ret; + return ret; } static int tdav_session_audio_set_ro(tmedia_session_t* self, const tsdp_header_M_t* m) { - int ret; - tsk_bool_t updated = tsk_false; - tdav_session_av_t* base = TDAV_SESSION_AV(self); - - if ((ret = tdav_session_av_set_ro(base, m, &updated))){ - TSK_DEBUG_ERROR("tdav_session_av_set_ro(audio) failed"); - return ret; - } - - if (updated) { - tsk_safeobj_lock(base); - // reset audio jitter buffer (new Offer probably comes with new seq_nums or timestamps) - if (base->consumer) { - ret = tdav_consumer_audio_reset(TDAV_CONSUMER_AUDIO(base->consumer)); - } - // destroy encoder to force requesting new one - TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec); - tsk_safeobj_unlock(base); - } - - return ret; + int ret; + tsk_bool_t updated = tsk_false; + tdav_session_av_t* base = TDAV_SESSION_AV(self); + + if ((ret = tdav_session_av_set_ro(base, m, &updated))) { + TSK_DEBUG_ERROR("tdav_session_av_set_ro(audio) failed"); + return ret; + } + + if (updated) { + tsk_safeobj_lock(base); + // reset audio jitter buffer (new Offer probably comes with new seq_nums or timestamps) + if (base->consumer) { + ret = tdav_consumer_audio_reset(TDAV_CONSUMER_AUDIO(base->consumer)); + } + // destroy encoder to force requesting new one + TSK_OBJECT_SAFE_FREE(TDAV_SESSION_AUDIO(self)->encoder.codec); + tsk_safeobj_unlock(base); + } + + return ret; } /* apply gain */ static void _tdav_session_audio_apply_gain(void* buffer, int len, int bps, int gain) { - register int i; - int max_val; - - max_val = (1 << (bps - 1 - gain)) - 1; - - if (bps == 8) { - int8_t *buff = buffer; - for (i = 0; i < len; i++) { - if (buff[i] > -max_val && buff[i] < max_val) - buff[i] = buff[i] << gain; - } - } - else if (bps == 16) { - int16_t *buff = buffer; - for (i = 0; i < len / 2; i++) { - if (buff[i] > -max_val && buff[i] < max_val) - buff[i] = buff[i] << gain; - } - } + register int i; + int max_val; + + max_val = (1 << (bps - 1 - gain)) - 1; + + if (bps == 8) { + int8_t *buff = buffer; + for (i = 0; i < len; i++) { + if (buff[i] > -max_val && buff[i] < max_val) { + buff[i] = buff[i] << gain; + } + } + } + else if (bps == 16) { + int16_t *buff = buffer; + for (i = 0; i < len / 2; i++) { + if (buff[i] > -max_val && buff[i] < max_val) { + buff[i] = buff[i] << gain; + } + } + } } /* Internal function used to create new DTMF event */ static tdav_session_audio_dtmfe_t* _tdav_session_audio_dtmfe_create(const tdav_session_audio_t* session, uint8_t event, uint16_t duration, uint32_t seq, uint32_t timestamp, uint8_t format, tsk_bool_t M, tsk_bool_t E) { - tdav_session_audio_dtmfe_t* dtmfe; - const tdav_session_av_t* base = (const tdav_session_av_t*)session; - static uint8_t volume = 10; - static uint32_t ssrc = 0x5234A8; - - uint8_t pay[4] = { 0 }; - - /* RFC 4733 - 2.3. Payload Format - 0 1 2 3 - 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - | event |E|R| volume | duration | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ - - if (!(dtmfe = tsk_object_new(tdav_session_audio_dtmfe_def_t))){ - TSK_DEBUG_ERROR("Failed to create new DTMF event"); - return tsk_null; - } - dtmfe->session = session; - - if (!(dtmfe->packet = trtp_rtp_packet_create((session && base->rtp_manager) ? base->rtp_manager->rtp.ssrc.local : ssrc, seq, timestamp, format, M))){ - TSK_DEBUG_ERROR("Failed to create DTMF RTP packet"); - TSK_OBJECT_SAFE_FREE(dtmfe); - return tsk_null; - } - - pay[0] = event; - pay[1] |= ((E << 7) | (volume & 0x3F)); - pay[2] = (duration >> 8); - pay[3] = (duration & 0xFF); - - /* set data */ - if ((dtmfe->packet->payload.data = tsk_calloc(sizeof(pay), sizeof(uint8_t)))){ - memcpy(dtmfe->packet->payload.data, pay, sizeof(pay)); - dtmfe->packet->payload.size = sizeof(pay); - } - - return dtmfe; + tdav_session_audio_dtmfe_t* dtmfe; + const tdav_session_av_t* base = (const tdav_session_av_t*)session; + static uint8_t volume = 10; + static uint32_t ssrc = 0x5234A8; + + uint8_t pay[4] = { 0 }; + + /* RFC 4733 - 2.3. Payload Format + 0 1 2 3 + 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | event |E|R| volume | duration | + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + + if (!(dtmfe = tsk_object_new(tdav_session_audio_dtmfe_def_t))) { + TSK_DEBUG_ERROR("Failed to create new DTMF event"); + return tsk_null; + } + dtmfe->session = session; + + if (!(dtmfe->packet = trtp_rtp_packet_create((session && base->rtp_manager) ? base->rtp_manager->rtp.ssrc.local : ssrc, seq, timestamp, format, M))) { + TSK_DEBUG_ERROR("Failed to create DTMF RTP packet"); + TSK_OBJECT_SAFE_FREE(dtmfe); + return tsk_null; + } + + pay[0] = event; + pay[1] |= ((E << 7) | (volume & 0x3F)); + pay[2] = (duration >> 8); + pay[3] = (duration & 0xFF); + + /* set data */ + if ((dtmfe->packet->payload.data = tsk_calloc(sizeof(pay), sizeof(uint8_t)))) { + memcpy(dtmfe->packet->payload.data, pay, sizeof(pay)); + dtmfe->packet->payload.size = sizeof(pay); + } + + return dtmfe; } static int _tdav_session_audio_dtmfe_timercb(const void* arg, tsk_timer_id_t timer_id) { - tdav_session_audio_dtmfe_t* dtmfe = (tdav_session_audio_dtmfe_t*)arg; - tdav_session_audio_t *audio; + tdav_session_audio_dtmfe_t* dtmfe = (tdav_session_audio_dtmfe_t*)arg; + tdav_session_audio_t *audio; - if (!dtmfe || !dtmfe->session || !dtmfe->session->dtmf_events){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!dtmfe || !dtmfe->session || !dtmfe->session->dtmf_events) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - /* Send the data */ - TSK_DEBUG_INFO("Sending DTMF event..."); - trtp_manager_send_rtp_packet(TDAV_SESSION_AV(dtmfe->session)->rtp_manager, dtmfe->packet, tsk_false); + /* Send the data */ + TSK_DEBUG_INFO("Sending DTMF event..."); + trtp_manager_send_rtp_packet(TDAV_SESSION_AV(dtmfe->session)->rtp_manager, dtmfe->packet, tsk_false); - audio = tsk_object_ref(TSK_OBJECT(dtmfe->session)); - tsk_list_lock(audio->dtmf_events); - /* Remove and delete the event from the queue */ - tsk_list_remove_item_by_data(audio->dtmf_events, dtmfe); - /* Check if there are pending events */ - audio->is_sending_dtmf_events = !TSK_LIST_IS_EMPTY(audio->dtmf_events); - tsk_list_unlock(audio->dtmf_events); - tsk_object_unref(audio); + audio = tsk_object_ref(TSK_OBJECT(dtmfe->session)); + tsk_list_lock(audio->dtmf_events); + /* Remove and delete the event from the queue */ + tsk_list_remove_item_by_data(audio->dtmf_events, dtmfe); + /* Check if there are pending events */ + audio->is_sending_dtmf_events = !TSK_LIST_IS_EMPTY(audio->dtmf_events); + tsk_list_unlock(audio->dtmf_events); + tsk_object_unref(audio); - return 0; + return 0; } static tmedia_resampler_t* _tdav_session_audio_resampler_create(int32_t bytes_per_sample, uint32_t in_freq, uint32_t out_freq, uint32_t frame_duration, uint32_t in_channels, uint32_t out_channels, uint32_t quality, void** resampler_buffer, tsk_size_t *resampler_buffer_size) { - uint32_t resampler_buff_size; - tmedia_resampler_t* resampler; - int ret; - - if (out_channels > 2 || in_channels > 2) { - TSK_DEBUG_ERROR("Invalid parameter: out_channels=%u, in_channels=%u", out_channels, in_channels); - return tsk_null; - } - - resampler_buff_size = (((out_freq * frame_duration) / 1000) * bytes_per_sample) << (out_channels == 2 ? 1 : 0); - - if (!(resampler = tmedia_resampler_create())) { - TSK_DEBUG_ERROR("Failed to create audio resampler"); - return tsk_null; - } - else { - if ((ret = tmedia_resampler_open(resampler, in_freq, out_freq, frame_duration, in_channels, out_channels, quality, 16))) { - TSK_DEBUG_ERROR("Failed to open audio resampler (%d, %d, %d, %d, %d,%d) with retcode=%d", in_freq, out_freq, frame_duration, in_channels, out_channels, quality, ret); - TSK_OBJECT_SAFE_FREE(resampler); - goto done; - } - } - // create temp resampler buffer - if ((*resampler_buffer = tsk_realloc(*resampler_buffer, resampler_buff_size))) { - *resampler_buffer_size = resampler_buff_size; - } - else { - *resampler_buffer_size = 0; - TSK_DEBUG_ERROR("Failed to allocate resampler buffer with size = %d", resampler_buff_size); - TSK_OBJECT_SAFE_FREE(resampler); - goto done; - } + uint32_t resampler_buff_size; + tmedia_resampler_t* resampler; + int ret; + + if (out_channels > 2 || in_channels > 2) { + TSK_DEBUG_ERROR("Invalid parameter: out_channels=%u, in_channels=%u", out_channels, in_channels); + return tsk_null; + } + + resampler_buff_size = (((out_freq * frame_duration) / 1000) * bytes_per_sample) << (out_channels == 2 ? 1 : 0); + + if (!(resampler = tmedia_resampler_create())) { + TSK_DEBUG_ERROR("Failed to create audio resampler"); + return tsk_null; + } + else { + if ((ret = tmedia_resampler_open(resampler, in_freq, out_freq, frame_duration, in_channels, out_channels, quality, 16))) { + TSK_DEBUG_ERROR("Failed to open audio resampler (%d, %d, %d, %d, %d,%d) with retcode=%d", in_freq, out_freq, frame_duration, in_channels, out_channels, quality, ret); + TSK_OBJECT_SAFE_FREE(resampler); + goto done; + } + } + // create temp resampler buffer + if ((*resampler_buffer = tsk_realloc(*resampler_buffer, resampler_buff_size))) { + *resampler_buffer_size = resampler_buff_size; + } + else { + *resampler_buffer_size = 0; + TSK_DEBUG_ERROR("Failed to allocate resampler buffer with size = %d", resampler_buff_size); + TSK_OBJECT_SAFE_FREE(resampler); + goto done; + } done: - return resampler; + return resampler; } //================================================================================================= @@ -811,142 +812,139 @@ done: /* constructor */ static tsk_object_t* tdav_session_audio_ctor(tsk_object_t * self, va_list * app) { - tdav_session_audio_t *audio = self; - if (audio){ - int ret; - tdav_session_av_t *base = TDAV_SESSION_AV(self); - - /* init() base */ - if ((ret = tdav_session_av_init(base, tmedia_audio)) != 0){ - TSK_DEBUG_ERROR("tdav_session_av_init(audio) failed"); - return tsk_null; - } - - /* init() self */ - if (base->producer){ - tmedia_producer_set_enc_callback(base->producer, tdav_session_audio_producer_enc_cb, audio); - } - if (base->consumer){ - // It's important to create the denoiser and jitter buffer here as dynamic plugins (from shared libs) don't have access to the registry - if (!(audio->denoise = tmedia_denoise_create())){ - TSK_DEBUG_WARN("No Audio denoiser found"); - } - else{ - // IMPORTANT: This means that the consumer must be child of "tdav_consumer_audio_t" object - tdav_consumer_audio_set_denoise(TDAV_CONSUMER_AUDIO(base->consumer), audio->denoise); - } - - if (!(audio->jitterbuffer = tmedia_jitterbuffer_create(tmedia_audio))){ - TSK_DEBUG_ERROR("Failed to create jitter buffer"); - } - else{ - ret = tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(audio->jitterbuffer)); - tdav_consumer_audio_set_jitterbuffer(TDAV_CONSUMER_AUDIO(base->consumer), audio->jitterbuffer); - } - } - } - return self; + tdav_session_audio_t *audio = self; + if (audio) { + int ret; + tdav_session_av_t *base = TDAV_SESSION_AV(self); + + /* init() base */ + if ((ret = tdav_session_av_init(base, tmedia_audio)) != 0) { + TSK_DEBUG_ERROR("tdav_session_av_init(audio) failed"); + return tsk_null; + } + + /* init() self */ + if (base->producer) { + tmedia_producer_set_enc_callback(base->producer, tdav_session_audio_producer_enc_cb, audio); + } + if (base->consumer) { + // It's important to create the denoiser and jitter buffer here as dynamic plugins (from shared libs) don't have access to the registry + if (!(audio->denoise = tmedia_denoise_create())) { + TSK_DEBUG_WARN("No Audio denoiser found"); + } + else { + // IMPORTANT: This means that the consumer must be child of "tdav_consumer_audio_t" object + tdav_consumer_audio_set_denoise(TDAV_CONSUMER_AUDIO(base->consumer), audio->denoise); + } + + if (!(audio->jitterbuffer = tmedia_jitterbuffer_create(tmedia_audio))) { + TSK_DEBUG_ERROR("Failed to create jitter buffer"); + } + else { + ret = tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(audio->jitterbuffer)); + tdav_consumer_audio_set_jitterbuffer(TDAV_CONSUMER_AUDIO(base->consumer), audio->jitterbuffer); + } + } + } + return self; } /* destructor */ static tsk_object_t* tdav_session_audio_dtor(tsk_object_t * self) { - tdav_session_audio_t *audio = self; - TSK_DEBUG_INFO("*** tdav_session_audio_t destroyed ***"); - if (audio){ - tdav_session_audio_stop((tmedia_session_t*)audio); - // Do it in this order (deinit self first) - - /* Timer manager */ - if (audio->timer.started){ - if (audio->dtmf_events){ - /* Cancel all events */ - tsk_list_item_t* item; - tsk_list_foreach(item, audio->dtmf_events){ - tsk_timer_mgr_global_cancel(((tdav_session_audio_dtmfe_t*)item->data)->timer_id); - } - } - } - - tsk_timer_mgr_global_unref(&audio->timer.handle_mgr_global); - - /* CleanUp the DTMF events */ - TSK_OBJECT_SAFE_FREE(audio->dtmf_events); - - TSK_OBJECT_SAFE_FREE(audio->denoise); - TSK_OBJECT_SAFE_FREE(audio->jitterbuffer); - - TSK_OBJECT_SAFE_FREE(audio->encoder.codec); - TSK_FREE(audio->encoder.buffer); - TSK_OBJECT_SAFE_FREE(audio->decoder.codec); - TSK_FREE(audio->decoder.buffer); - - // free resamplers - TSK_FREE(audio->encoder.resampler.buffer); - TSK_OBJECT_SAFE_FREE(audio->encoder.resampler.instance); - TSK_FREE(audio->decoder.resampler.buffer); - TSK_OBJECT_SAFE_FREE(audio->decoder.resampler.instance); - - /* deinit base */ - tdav_session_av_deinit(TDAV_SESSION_AV(self)); - - TSK_DEBUG_INFO("*** Audio session destroyed ***"); - } - - return self; + tdav_session_audio_t *audio = self; + TSK_DEBUG_INFO("*** tdav_session_audio_t destroyed ***"); + if (audio) { + tdav_session_audio_stop((tmedia_session_t*)audio); + // Do it in this order (deinit self first) + + /* Timer manager */ + if (audio->timer.started) { + if (audio->dtmf_events) { + /* Cancel all events */ + tsk_list_item_t* item; + tsk_list_foreach(item, audio->dtmf_events) { + tsk_timer_mgr_global_cancel(((tdav_session_audio_dtmfe_t*)item->data)->timer_id); + } + } + } + + tsk_timer_mgr_global_unref(&audio->timer.handle_mgr_global); + + /* CleanUp the DTMF events */ + TSK_OBJECT_SAFE_FREE(audio->dtmf_events); + + TSK_OBJECT_SAFE_FREE(audio->denoise); + TSK_OBJECT_SAFE_FREE(audio->jitterbuffer); + + TSK_OBJECT_SAFE_FREE(audio->encoder.codec); + TSK_FREE(audio->encoder.buffer); + TSK_OBJECT_SAFE_FREE(audio->decoder.codec); + TSK_FREE(audio->decoder.buffer); + + // free resamplers + TSK_FREE(audio->encoder.resampler.buffer); + TSK_OBJECT_SAFE_FREE(audio->encoder.resampler.instance); + TSK_FREE(audio->decoder.resampler.buffer); + TSK_OBJECT_SAFE_FREE(audio->decoder.resampler.instance); + + /* deinit base */ + tdav_session_av_deinit(TDAV_SESSION_AV(self)); + + TSK_DEBUG_INFO("*** Audio session destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_session_audio_def_s = -{ - sizeof(tdav_session_audio_t), - tdav_session_audio_ctor, - tdav_session_audio_dtor, - tmedia_session_cmp, +static const tsk_object_def_t tdav_session_audio_def_s = { + sizeof(tdav_session_audio_t), + tdav_session_audio_ctor, + tdav_session_audio_dtor, + tmedia_session_cmp, }; /* plugin definition*/ -static const tmedia_session_plugin_def_t tdav_session_audio_plugin_def_s = -{ - &tdav_session_audio_def_s, - - tmedia_audio, - "audio", - - tdav_session_audio_set, - tdav_session_audio_get, - tdav_session_audio_prepare, - tdav_session_audio_start, - tdav_session_audio_pause, - tdav_session_audio_stop, - - /* Audio part */ - { - tdav_session_audio_send_dtmf - }, - - tdav_session_audio_get_lo, - tdav_session_audio_set_ro +static const tmedia_session_plugin_def_t tdav_session_audio_plugin_def_s = { + &tdav_session_audio_def_s, + + tmedia_audio, + "audio", + + tdav_session_audio_set, + tdav_session_audio_get, + tdav_session_audio_prepare, + tdav_session_audio_start, + tdav_session_audio_pause, + tdav_session_audio_stop, + + /* Audio part */ + { + tdav_session_audio_send_dtmf + }, + + tdav_session_audio_get_lo, + tdav_session_audio_set_ro }; const tmedia_session_plugin_def_t *tdav_session_audio_plugin_def_t = &tdav_session_audio_plugin_def_s; -static const tmedia_session_plugin_def_t tdav_session_bfcpaudio_plugin_def_s = -{ - &tdav_session_audio_def_s, - - tmedia_bfcp_audio, - "audio", - - tdav_session_audio_set, - tdav_session_audio_get, - tdav_session_audio_prepare, - tdav_session_audio_start, - tdav_session_audio_pause, - tdav_session_audio_stop, - - /* Audio part */ - { - tdav_session_audio_send_dtmf - }, - - tdav_session_audio_get_lo, - tdav_session_audio_set_ro +static const tmedia_session_plugin_def_t tdav_session_bfcpaudio_plugin_def_s = { + &tdav_session_audio_def_s, + + tmedia_bfcp_audio, + "audio", + + tdav_session_audio_set, + tdav_session_audio_get, + tdav_session_audio_prepare, + tdav_session_audio_start, + tdav_session_audio_pause, + tdav_session_audio_stop, + + /* Audio part */ + { + tdav_session_audio_send_dtmf + }, + + tdav_session_audio_get_lo, + tdav_session_audio_set_ro }; const tmedia_session_plugin_def_t *tdav_session_bfcpaudio_plugin_def_t = &tdav_session_bfcpaudio_plugin_def_s; @@ -957,35 +955,34 @@ const tmedia_session_plugin_def_t *tdav_session_bfcpaudio_plugin_def_t = &tdav_s // static tsk_object_t* tdav_session_audio_dtmfe_ctor(tsk_object_t * self, va_list * app) { - tdav_session_audio_dtmfe_t *event = self; - if (event){ - event->timer_id = TSK_INVALID_TIMER_ID; - } - return self; + tdav_session_audio_dtmfe_t *event = self; + if (event) { + event->timer_id = TSK_INVALID_TIMER_ID; + } + return self; } static tsk_object_t* tdav_session_audio_dtmfe_dtor(tsk_object_t * self) { - tdav_session_audio_dtmfe_t *event = self; - if (event){ - TSK_OBJECT_SAFE_FREE(event->packet); - } + tdav_session_audio_dtmfe_t *event = self; + if (event) { + TSK_OBJECT_SAFE_FREE(event->packet); + } - return self; + return self; } static int tdav_session_audio_dtmfe_cmp(const tsk_object_t *_e1, const tsk_object_t *_e2) { - int ret; - tsk_subsat_int32_ptr(_e1, _e2, &ret); - return ret; + int ret; + tsk_subsat_int32_ptr(_e1, _e2, &ret); + return ret; } -static const tsk_object_def_t tdav_session_audio_dtmfe_def_s = -{ - sizeof(tdav_session_audio_dtmfe_t), - tdav_session_audio_dtmfe_ctor, - tdav_session_audio_dtmfe_dtor, - tdav_session_audio_dtmfe_cmp, +static const tsk_object_def_t tdav_session_audio_dtmfe_def_s = { + sizeof(tdav_session_audio_dtmfe_t), + tdav_session_audio_dtmfe_ctor, + tdav_session_audio_dtmfe_dtor, + tdav_session_audio_dtmfe_cmp, }; const tsk_object_def_t *tdav_session_audio_dtmfe_def_t = &tdav_session_audio_dtmfe_def_s; diff --git a/tinyDAV/src/audio/tdav_speakup_jitterbuffer.c b/tinyDAV/src/audio/tdav_speakup_jitterbuffer.c index cccc235..0c21e41 100755 --- a/tinyDAV/src/audio/tdav_speakup_jitterbuffer.c +++ b/tinyDAV/src/audio/tdav_speakup_jitterbuffer.c @@ -2,19 +2,19 @@ * Copyright (C) 2011 Mamadou Diop. * * Contact: Mamadou Diop <diopmamadou(at)doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -41,7 +41,7 @@ #if TSK_UNDER_WINDOWS # include <Winsock2.h> // timeval #elif defined(__SYMBIAN32__) -# include <_timeval.h> +# include <_timeval.h> #else # include <sys/time.h> #endif @@ -52,171 +52,170 @@ static int tdav_speakup_jitterbuffer_set(tmedia_jitterbuffer_t *self, const tmedia_param_t* param) { - TSK_DEBUG_ERROR("Not implemented"); - return -2; + TSK_DEBUG_ERROR("Not implemented"); + return -2; } static int tdav_speakup_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate, uint32_t channels) { - tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; - if(!jitterbuffer->jbuffer){ - if(!(jitterbuffer->jbuffer = jb_new())){ - TSK_DEBUG_ERROR("Failed to create new buffer"); - return -1; - } - jitterbuffer->jcodec = JB_CODEC_OTHER; - } - jitterbuffer->ref_timestamp = 0; - jitterbuffer->frame_duration = frame_duration; - jitterbuffer->rate = rate; - jitterbuffer->channels = channels; - jitterbuffer->_10ms_size_bytes = 160 * (rate/8000); - - return 0; + tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; + if(!jitterbuffer->jbuffer) { + if(!(jitterbuffer->jbuffer = jb_new())) { + TSK_DEBUG_ERROR("Failed to create new buffer"); + return -1; + } + jitterbuffer->jcodec = JB_CODEC_OTHER; + } + jitterbuffer->ref_timestamp = 0; + jitterbuffer->frame_duration = frame_duration; + jitterbuffer->rate = rate; + jitterbuffer->channels = channels; + jitterbuffer->_10ms_size_bytes = 160 * (rate/8000); + + return 0; } static int tdav_speakup_jitterbuffer_tick(tmedia_jitterbuffer_t* self) { - return 0; + return 0; } static int tdav_speakup_jitterbuffer_put(tmedia_jitterbuffer_t* self, void* data, tsk_size_t data_size, const tsk_object_t* proto_hdr) { - tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; - const trtp_rtp_header_t* rtp_hdr = (const trtp_rtp_header_t*)proto_hdr; + tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; + const trtp_rtp_header_t* rtp_hdr = (const trtp_rtp_header_t*)proto_hdr; int i; long now, ts; void* _10ms_buf; - uint8_t* pdata; - - if(!self || !data || !data_size || !jitterbuffer->jbuffer || !rtp_hdr){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - /* synchronize the reference timestamp */ - if(!jitterbuffer->ref_timestamp){ - uint64_t now = tsk_time_now(); - struct timeval tv; - long ts = (rtp_hdr->timestamp/(jitterbuffer->rate/1000)); - //=> Do not use (see clock_gettime() on linux): tsk_gettimeofday(&tv, tsk_null); - tv.tv_sec = (long)(now)/1000; - tv.tv_usec = (long)(now - (tv.tv_sec*1000))*1000; - - tv.tv_sec -= (ts / jitterbuffer->rate); - tv.tv_usec -= (ts % jitterbuffer->rate) * 125; - if((tv.tv_usec -= (tv.tv_usec % (TDAV_SPEAKUP_10MS * 10000))) <0){ - tv.tv_usec += 1000000; - tv.tv_sec -= 1; - } - jitterbuffer->ref_timestamp = tsk_time_get_ms(&tv); - - switch(rtp_hdr->payload_type){ - case 8: /*TMEDIA_CODEC_FORMAT_G711a*/ - case 0: /* TMEDIA_CODEC_FORMAT_G711u */ - jitterbuffer->jcodec = JB_CODEC_G711x; - break; - case 18: /* TMEDIA_CODEC_FORMAT_G729 */ - jitterbuffer->jcodec = JB_CODEC_G729A; - break; - case 3: /* TMEDIA_CODEC_FORMAT_GSM */ - jitterbuffer->jcodec = JB_CODEC_GSM_EFR; - break; - - default: - jitterbuffer->jcodec = JB_CODEC_OTHER; - break; - } - } - - // split as several 10ms frames - now = (long) (tsk_time_now()-jitterbuffer->ref_timestamp); - ts = (long)(rtp_hdr->timestamp/(jitterbuffer->rate/1000)); - pdata = (uint8_t*)data; - for(i=0; i<(int)(data_size/jitterbuffer->_10ms_size_bytes);i++){ - if((_10ms_buf = tsk_calloc(jitterbuffer->_10ms_size_bytes, 1))){ - memcpy(_10ms_buf, &pdata[i*jitterbuffer->_10ms_size_bytes], jitterbuffer->_10ms_size_bytes); - jb_put(jitterbuffer->jbuffer, _10ms_buf, JB_TYPE_VOICE, TDAV_SPEAKUP_10MS, ts, now, jitterbuffer->jcodec); - _10ms_buf = tsk_null; - } - ts += TDAV_SPEAKUP_10MS; - } - - return 0; + uint8_t* pdata; + + if(!self || !data || !data_size || !jitterbuffer->jbuffer || !rtp_hdr) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + /* synchronize the reference timestamp */ + if(!jitterbuffer->ref_timestamp) { + uint64_t now = tsk_time_now(); + struct timeval tv; + long ts = (rtp_hdr->timestamp/(jitterbuffer->rate/1000)); + //=> Do not use (see clock_gettime() on linux): tsk_gettimeofday(&tv, tsk_null); + tv.tv_sec = (long)(now)/1000; + tv.tv_usec = (long)(now - (tv.tv_sec*1000))*1000; + + tv.tv_sec -= (ts / jitterbuffer->rate); + tv.tv_usec -= (ts % jitterbuffer->rate) * 125; + if((tv.tv_usec -= (tv.tv_usec % (TDAV_SPEAKUP_10MS * 10000))) <0) { + tv.tv_usec += 1000000; + tv.tv_sec -= 1; + } + jitterbuffer->ref_timestamp = tsk_time_get_ms(&tv); + + switch(rtp_hdr->payload_type) { + case 8: /*TMEDIA_CODEC_FORMAT_G711a*/ + case 0: /* TMEDIA_CODEC_FORMAT_G711u */ + jitterbuffer->jcodec = JB_CODEC_G711x; + break; + case 18: /* TMEDIA_CODEC_FORMAT_G729 */ + jitterbuffer->jcodec = JB_CODEC_G729A; + break; + case 3: /* TMEDIA_CODEC_FORMAT_GSM */ + jitterbuffer->jcodec = JB_CODEC_GSM_EFR; + break; + + default: + jitterbuffer->jcodec = JB_CODEC_OTHER; + break; + } + } + + // split as several 10ms frames + now = (long) (tsk_time_now()-jitterbuffer->ref_timestamp); + ts = (long)(rtp_hdr->timestamp/(jitterbuffer->rate/1000)); + pdata = (uint8_t*)data; + for(i=0; i<(int)(data_size/jitterbuffer->_10ms_size_bytes); i++) { + if((_10ms_buf = tsk_calloc(jitterbuffer->_10ms_size_bytes, 1))) { + memcpy(_10ms_buf, &pdata[i*jitterbuffer->_10ms_size_bytes], jitterbuffer->_10ms_size_bytes); + jb_put(jitterbuffer->jbuffer, _10ms_buf, JB_TYPE_VOICE, TDAV_SPEAKUP_10MS, ts, now, jitterbuffer->jcodec); + _10ms_buf = tsk_null; + } + ts += TDAV_SPEAKUP_10MS; + } + + return 0; } static tsk_size_t tdav_speakup_jitterbuffer_get(tmedia_jitterbuffer_t* self, void* out_data, tsk_size_t out_size) { - tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; - int jret; - - int i, _10ms_count; - long now; - short* _10ms_buf = tsk_null; - uint8_t* pout_data = (uint8_t*)out_data; - - if(!out_data || (out_size % jitterbuffer->_10ms_size_bytes)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return 0; - } - - _10ms_count = (out_size/jitterbuffer->_10ms_size_bytes); - now = (long) (tsk_time_now() - jitterbuffer->ref_timestamp); - for(i=0; i<_10ms_count; i++){ - - jret = jb_get(jitterbuffer->jbuffer, (void**)&_10ms_buf, now, TDAV_SPEAKUP_10MS); - switch(jret){ - case JB_INTERP: - TSK_DEBUG_INFO("JB_INTERP"); - jb_reset_all(jitterbuffer->jbuffer); - memset(&pout_data[i*jitterbuffer->_10ms_size_bytes], 0, (_10ms_count*jitterbuffer->_10ms_size_bytes)-(i*jitterbuffer->_10ms_size_bytes)); - i = _10ms_count; // for exit - break; - case JB_OK: - case JB_EMPTY: - case JB_NOFRAME: - case JB_NOJB: - { - if(_10ms_buf && (jret == JB_OK)){ - /* copy data */ - memcpy(&pout_data[i*jitterbuffer->_10ms_size_bytes], _10ms_buf, jitterbuffer->_10ms_size_bytes); - } - else{ - /* copy silence */ - memset(&pout_data[i*jitterbuffer->_10ms_size_bytes], 0, jitterbuffer->_10ms_size_bytes); - } - } - - default: - break; - } - TSK_FREE(_10ms_buf); - } - - return (_10ms_count * jitterbuffer->_10ms_size_bytes); + tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; + int jret; + + int i, _10ms_count; + long now; + short* _10ms_buf = tsk_null; + uint8_t* pout_data = (uint8_t*)out_data; + + if(!out_data || (out_size % jitterbuffer->_10ms_size_bytes)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return 0; + } + + _10ms_count = (out_size/jitterbuffer->_10ms_size_bytes); + now = (long) (tsk_time_now() - jitterbuffer->ref_timestamp); + for(i=0; i<_10ms_count; i++) { + + jret = jb_get(jitterbuffer->jbuffer, (void**)&_10ms_buf, now, TDAV_SPEAKUP_10MS); + switch(jret) { + case JB_INTERP: + TSK_DEBUG_INFO("JB_INTERP"); + jb_reset_all(jitterbuffer->jbuffer); + memset(&pout_data[i*jitterbuffer->_10ms_size_bytes], 0, (_10ms_count*jitterbuffer->_10ms_size_bytes)-(i*jitterbuffer->_10ms_size_bytes)); + i = _10ms_count; // for exit + break; + case JB_OK: + case JB_EMPTY: + case JB_NOFRAME: + case JB_NOJB: { + if(_10ms_buf && (jret == JB_OK)) { + /* copy data */ + memcpy(&pout_data[i*jitterbuffer->_10ms_size_bytes], _10ms_buf, jitterbuffer->_10ms_size_bytes); + } + else { + /* copy silence */ + memset(&pout_data[i*jitterbuffer->_10ms_size_bytes], 0, jitterbuffer->_10ms_size_bytes); + } + } + + default: + break; + } + TSK_FREE(_10ms_buf); + } + + return (_10ms_count * jitterbuffer->_10ms_size_bytes); } static int tdav_speakup_jitterbuffer_reset(tmedia_jitterbuffer_t* self) { - tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; - if(jitterbuffer->jbuffer){ - jb_reset_all(jitterbuffer->jbuffer); - return 0; - } - else{ - TSK_DEBUG_ERROR("invalid parameter"); - return -1; - } + tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; + if(jitterbuffer->jbuffer) { + jb_reset_all(jitterbuffer->jbuffer); + return 0; + } + else { + TSK_DEBUG_ERROR("invalid parameter"); + return -1; + } } static int tdav_speakup_jitterbuffer_close(tmedia_jitterbuffer_t* self) { - tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; - if(jitterbuffer->jbuffer){ - jb_destroy(jitterbuffer->jbuffer); - jitterbuffer->jbuffer = tsk_null; - } - return 0; + tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self; + if(jitterbuffer->jbuffer) { + jb_destroy(jitterbuffer->jbuffer); + jitterbuffer->jbuffer = tsk_null; + } + return 0; } @@ -228,53 +227,51 @@ static int tdav_speakup_jitterbuffer_close(tmedia_jitterbuffer_t* self) /* constructor */ static tsk_object_t* tdav_speakup_jitterbuffer_ctor(tsk_object_t * self, va_list * app) { - tdav_speakup_jitterbuffer_t *jitterbuffer = self; - TSK_DEBUG_INFO("Create speekup jitter buffer"); - if(jitterbuffer){ - /* init base */ - tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(jitterbuffer)); - /* init self */ - } - return self; + tdav_speakup_jitterbuffer_t *jitterbuffer = self; + TSK_DEBUG_INFO("Create speekup jitter buffer"); + if(jitterbuffer) { + /* init base */ + tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(jitterbuffer)); + /* init self */ + } + return self; } /* destructor */ static tsk_object_t* tdav_speakup_jitterbuffer_dtor(tsk_object_t * self) -{ - tdav_speakup_jitterbuffer_t *jitterbuffer = self; - if(jitterbuffer){ - /* deinit base */ - tmedia_jitterbuffer_deinit(TMEDIA_JITTER_BUFFER(jitterbuffer)); - /* deinit self */ - if(jitterbuffer->jbuffer){ - jb_destroy(jitterbuffer->jbuffer); - jitterbuffer->jbuffer = tsk_null; - } - } - - return self; +{ + tdav_speakup_jitterbuffer_t *jitterbuffer = self; + if(jitterbuffer) { + /* deinit base */ + tmedia_jitterbuffer_deinit(TMEDIA_JITTER_BUFFER(jitterbuffer)); + /* deinit self */ + if(jitterbuffer->jbuffer) { + jb_destroy(jitterbuffer->jbuffer); + jitterbuffer->jbuffer = tsk_null; + } + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_speakup_jitterbuffer_def_s = -{ - sizeof(tdav_speakup_jitterbuffer_t), - tdav_speakup_jitterbuffer_ctor, - tdav_speakup_jitterbuffer_dtor, - tsk_null, +static const tsk_object_def_t tdav_speakup_jitterbuffer_def_s = { + sizeof(tdav_speakup_jitterbuffer_t), + tdav_speakup_jitterbuffer_ctor, + tdav_speakup_jitterbuffer_dtor, + tsk_null, }; /* plugin definition*/ -static const tmedia_jitterbuffer_plugin_def_t tdav_speakup_jitterbuffer_plugin_def_s = -{ - &tdav_speakup_jitterbuffer_def_s, - tmedia_audio, - "Audio/video JitterBuffer based on Speakup", - - tdav_speakup_jitterbuffer_set, - tdav_speakup_jitterbuffer_open, - tdav_speakup_jitterbuffer_tick, - tdav_speakup_jitterbuffer_put, - tdav_speakup_jitterbuffer_get, - tdav_speakup_jitterbuffer_reset, - tdav_speakup_jitterbuffer_close, +static const tmedia_jitterbuffer_plugin_def_t tdav_speakup_jitterbuffer_plugin_def_s = { + &tdav_speakup_jitterbuffer_def_s, + tmedia_audio, + "Audio/video JitterBuffer based on Speakup", + + tdav_speakup_jitterbuffer_set, + tdav_speakup_jitterbuffer_open, + tdav_speakup_jitterbuffer_tick, + tdav_speakup_jitterbuffer_put, + tdav_speakup_jitterbuffer_get, + tdav_speakup_jitterbuffer_reset, + tdav_speakup_jitterbuffer_close, }; const tmedia_jitterbuffer_plugin_def_t *tdav_speakup_jitterbuffer_plugin_def_t = &tdav_speakup_jitterbuffer_plugin_def_s; diff --git a/tinyDAV/src/audio/tdav_speex_denoise.c b/tinyDAV/src/audio/tdav_speex_denoise.c index 4f344dd..ee2733e 100755 --- a/tinyDAV/src/audio/tdav_speex_denoise.c +++ b/tinyDAV/src/audio/tdav_speex_denoise.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2010-2011 Mamadou Diop. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -37,205 +37,204 @@ #include <speex/speex_echo.h> /** Speex denoiser*/ -typedef struct tdav_speex_denoise_s -{ - TMEDIA_DECLARE_DENOISE; +typedef struct tdav_speex_denoise_s { + TMEDIA_DECLARE_DENOISE; - SpeexPreprocessState *preprocess_state_record; - SpeexPreprocessState *preprocess_state_playback; - SpeexEchoState *echo_state; + SpeexPreprocessState *preprocess_state_record; + SpeexPreprocessState *preprocess_state_playback; + SpeexEchoState *echo_state; - spx_int16_t* echo_output_frame; - uint32_t record_frame_size_samples, record_frame_size_bytes; - uint32_t playback_frame_size_samples, playback_frame_size_bytes; + spx_int16_t* echo_output_frame; + uint32_t record_frame_size_samples, record_frame_size_bytes; + uint32_t playback_frame_size_samples, playback_frame_size_bytes; } tdav_speex_denoise_t; static int tdav_speex_denoise_set(tmedia_denoise_t* _self, const tmedia_param_t* param) { - tdav_speex_denoise_t *self = (tdav_speex_denoise_t *)_self; - if(!self || !param){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(param->value_type == tmedia_pvt_int32){ - if(tsk_striequals(param->key, "echo-tail")){ - int32_t echo_tail = *((int32_t*)param->value); - TSK_DEBUG_INFO("speex_set_echo_tail(%d) ignore", echo_tail); // because Speex AEC just do not work (use WebRTC) - return 0; - } - } - return -1; + tdav_speex_denoise_t *self = (tdav_speex_denoise_t *)_self; + if(!self || !param) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(param->value_type == tmedia_pvt_int32) { + if(tsk_striequals(param->key, "echo-tail")) { + int32_t echo_tail = *((int32_t*)param->value); + TSK_DEBUG_INFO("speex_set_echo_tail(%d) ignore", echo_tail); // because Speex AEC just do not work (use WebRTC) + return 0; + } + } + return -1; } static int tdav_speex_denoise_open(tmedia_denoise_t* self, uint32_t record_frame_size_samples, uint32_t record_sampling_rate, uint32_t record_channels, uint32_t playback_frame_size_samples, uint32_t playback_sampling_rate, uint32_t playback_channels) { - tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; - float f; - int i; - - if (!denoiser->echo_state && TMEDIA_DENOISE(denoiser)->echo_supp_enabled) { - TSK_DEBUG_INFO("Init Aec frame_size[%u] filter_length[%u] SampleRate[%u]", - (uint32_t)(record_frame_size_samples),TMEDIA_DENOISE(denoiser)->echo_tail*record_frame_size_samples, record_sampling_rate); - if((denoiser->echo_state = speex_echo_state_init(record_frame_size_samples, TMEDIA_DENOISE(denoiser)->echo_tail))){ - speex_echo_ctl(denoiser->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE, &record_sampling_rate); - } - } - - if (!denoiser->preprocess_state_record && !denoiser->preprocess_state_playback) { - denoiser->record_frame_size_samples = record_frame_size_samples; - denoiser->record_frame_size_bytes = (record_frame_size_samples << 1); - denoiser->playback_frame_size_samples = playback_frame_size_samples; - denoiser->playback_frame_size_bytes = (playback_frame_size_samples << 1); - - if((denoiser->preprocess_state_record = speex_preprocess_state_init(record_frame_size_samples, record_sampling_rate)) - && (denoiser->preprocess_state_playback = speex_preprocess_state_init(playback_frame_size_samples, playback_sampling_rate)) - ){ - - // Echo suppression - if(denoiser->echo_state){ - int echo_supp , echo_supp_active = 0; - - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_STATE, denoiser->echo_state); - - TSK_FREE(denoiser->echo_output_frame); - denoiser->echo_output_frame = tsk_calloc(denoiser->record_frame_size_samples, sizeof(spx_int16_t)); - - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS , &echo_supp ); - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); - TSK_DEBUG_INFO("AEC echo_supp level [%d] echo_supp_active level[%d] ", echo_supp , echo_supp_active); - echo_supp = -60 ; - echo_supp_active = -60 ; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS , &echo_supp ); - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); - // TRACES - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS , &echo_supp ); - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); - TSK_DEBUG_INFO("New aec echo_supp level [%d] echo_supp_active level[%d] ", echo_supp , echo_supp_active); - } - - // Noise suppression - if(TMEDIA_DENOISE(denoiser)->noise_supp_enabled){ - TSK_DEBUG_INFO("SpeexDSP: Noise supp enabled"); - i = 1; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_DENOISE, &i); - speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_DENOISE, &i); - i = TMEDIA_DENOISE(denoiser)->noise_supp_level; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); - speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); - } - else{ - i = 0; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_DENOISE, &i); - speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_DENOISE, &i); - } - - // Automatic gain control - if(TMEDIA_DENOISE(denoiser)->agc_enabled){ - float agc_level = TMEDIA_DENOISE(denoiser)->agc_level; - TSK_DEBUG_INFO("SpeexDSP: AGC enabled"); - - i = 1; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC, &i); - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC_LEVEL, &agc_level); - } - else{ - i = 0, f = 8000.0f; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC, &i); - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC_LEVEL, &f); - } - - // Voice Activity detection - i = TMEDIA_DENOISE(denoiser)->vad_enabled ? 1 : 0; - speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_VAD, &i); - - return 0; - } - else{ - TSK_DEBUG_ERROR("Failed to create Speex preprocessor state"); - return -2; - } - } - - return 0; + tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; + float f; + int i; + + if (!denoiser->echo_state && TMEDIA_DENOISE(denoiser)->echo_supp_enabled) { + TSK_DEBUG_INFO("Init Aec frame_size[%u] filter_length[%u] SampleRate[%u]", + (uint32_t)(record_frame_size_samples),TMEDIA_DENOISE(denoiser)->echo_tail*record_frame_size_samples, record_sampling_rate); + if((denoiser->echo_state = speex_echo_state_init(record_frame_size_samples, TMEDIA_DENOISE(denoiser)->echo_tail))) { + speex_echo_ctl(denoiser->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE, &record_sampling_rate); + } + } + + if (!denoiser->preprocess_state_record && !denoiser->preprocess_state_playback) { + denoiser->record_frame_size_samples = record_frame_size_samples; + denoiser->record_frame_size_bytes = (record_frame_size_samples << 1); + denoiser->playback_frame_size_samples = playback_frame_size_samples; + denoiser->playback_frame_size_bytes = (playback_frame_size_samples << 1); + + if((denoiser->preprocess_state_record = speex_preprocess_state_init(record_frame_size_samples, record_sampling_rate)) + && (denoiser->preprocess_state_playback = speex_preprocess_state_init(playback_frame_size_samples, playback_sampling_rate)) + ) { + + // Echo suppression + if(denoiser->echo_state) { + int echo_supp , echo_supp_active = 0; + + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_STATE, denoiser->echo_state); + + TSK_FREE(denoiser->echo_output_frame); + denoiser->echo_output_frame = tsk_calloc(denoiser->record_frame_size_samples, sizeof(spx_int16_t)); + + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS , &echo_supp ); + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); + TSK_DEBUG_INFO("AEC echo_supp level [%d] echo_supp_active level[%d] ", echo_supp , echo_supp_active); + echo_supp = -60 ; + echo_supp_active = -60 ; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS , &echo_supp ); + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); + // TRACES + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS , &echo_supp ); + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE , &echo_supp_active ); + TSK_DEBUG_INFO("New aec echo_supp level [%d] echo_supp_active level[%d] ", echo_supp , echo_supp_active); + } + + // Noise suppression + if(TMEDIA_DENOISE(denoiser)->noise_supp_enabled) { + TSK_DEBUG_INFO("SpeexDSP: Noise supp enabled"); + i = 1; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_DENOISE, &i); + speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_DENOISE, &i); + i = TMEDIA_DENOISE(denoiser)->noise_supp_level; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); + speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); + } + else { + i = 0; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_DENOISE, &i); + speex_preprocess_ctl(denoiser->preprocess_state_playback, SPEEX_PREPROCESS_SET_DENOISE, &i); + } + + // Automatic gain control + if(TMEDIA_DENOISE(denoiser)->agc_enabled) { + float agc_level = TMEDIA_DENOISE(denoiser)->agc_level; + TSK_DEBUG_INFO("SpeexDSP: AGC enabled"); + + i = 1; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC, &i); + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC_LEVEL, &agc_level); + } + else { + i = 0, f = 8000.0f; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC, &i); + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_AGC_LEVEL, &f); + } + + // Voice Activity detection + i = TMEDIA_DENOISE(denoiser)->vad_enabled ? 1 : 0; + speex_preprocess_ctl(denoiser->preprocess_state_record, SPEEX_PREPROCESS_SET_VAD, &i); + + return 0; + } + else { + TSK_DEBUG_ERROR("Failed to create Speex preprocessor state"); + return -2; + } + } + + return 0; } static int tdav_speex_denoise_echo_playback(tmedia_denoise_t* self, const void* echo_frame, uint32_t echo_frame_size_bytes) { - tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; + tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; - if(denoiser->record_frame_size_bytes != echo_frame_size_bytes){ - TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->record_frame_size_bytes, echo_frame_size_bytes); - return -1; - } + if(denoiser->record_frame_size_bytes != echo_frame_size_bytes) { + TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->record_frame_size_bytes, echo_frame_size_bytes); + return -1; + } - if(denoiser->echo_state){ - speex_echo_playback(denoiser->echo_state, echo_frame); - } - return 0; + if(denoiser->echo_state) { + speex_echo_playback(denoiser->echo_state, echo_frame); + } + return 0; } static int tdav_speex_denoise_process_record(tmedia_denoise_t* self, void* audio_frame, uint32_t audio_frame_size_bytes, tsk_bool_t* silence_or_noise) { - tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; - int vad; - - if(denoiser->record_frame_size_bytes != audio_frame_size_bytes){ - TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->record_frame_size_bytes, audio_frame_size_bytes); - return -1; - } - - if(denoiser->preprocess_state_record){ - if(denoiser->echo_state && denoiser->echo_output_frame){ - speex_echo_capture(denoiser->echo_state, audio_frame, denoiser->echo_output_frame); - memcpy(audio_frame, denoiser->echo_output_frame, denoiser->record_frame_size_bytes); - } - vad = speex_preprocess_run(denoiser->preprocess_state_record, audio_frame); - if(!vad && TMEDIA_DENOISE(denoiser)->vad_enabled){ - *silence_or_noise = tsk_true; - } - } - - return 0; + tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; + int vad; + + if(denoiser->record_frame_size_bytes != audio_frame_size_bytes) { + TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->record_frame_size_bytes, audio_frame_size_bytes); + return -1; + } + + if(denoiser->preprocess_state_record) { + if(denoiser->echo_state && denoiser->echo_output_frame) { + speex_echo_capture(denoiser->echo_state, audio_frame, denoiser->echo_output_frame); + memcpy(audio_frame, denoiser->echo_output_frame, denoiser->record_frame_size_bytes); + } + vad = speex_preprocess_run(denoiser->preprocess_state_record, audio_frame); + if(!vad && TMEDIA_DENOISE(denoiser)->vad_enabled) { + *silence_or_noise = tsk_true; + } + } + + return 0; } static int tdav_speex_denoise_process_playback(tmedia_denoise_t* self, void* audio_frame, uint32_t audio_frame_size_bytes) { - tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; + tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; - if(denoiser->playback_frame_size_bytes != audio_frame_size_bytes){ - TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->playback_frame_size_bytes, audio_frame_size_bytes); - return -1; - } + if(denoiser->playback_frame_size_bytes != audio_frame_size_bytes) { + TSK_DEBUG_ERROR("Size mismatch: %u<>%u", denoiser->playback_frame_size_bytes, audio_frame_size_bytes); + return -1; + } - if(denoiser->preprocess_state_playback){ - speex_preprocess_run(denoiser->preprocess_state_playback, audio_frame); - } - return 0; + if(denoiser->preprocess_state_playback) { + speex_preprocess_run(denoiser->preprocess_state_playback, audio_frame); + } + return 0; } static int tdav_speex_denoise_close(tmedia_denoise_t* self) { - tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; - - if(denoiser->preprocess_state_record){ - speex_preprocess_state_destroy(denoiser->preprocess_state_record); - denoiser->preprocess_state_record = tsk_null; - } - if(denoiser->preprocess_state_playback){ - speex_preprocess_state_destroy(denoiser->preprocess_state_playback); - denoiser->preprocess_state_playback = tsk_null; - } - if(denoiser->echo_state){ - speex_echo_state_destroy(denoiser->echo_state); - denoiser->echo_state = tsk_null; - } - TSK_FREE(denoiser->echo_output_frame); - - return 0; + tdav_speex_denoise_t *denoiser = (tdav_speex_denoise_t *)self; + + if(denoiser->preprocess_state_record) { + speex_preprocess_state_destroy(denoiser->preprocess_state_record); + denoiser->preprocess_state_record = tsk_null; + } + if(denoiser->preprocess_state_playback) { + speex_preprocess_state_destroy(denoiser->preprocess_state_playback); + denoiser->preprocess_state_playback = tsk_null; + } + if(denoiser->echo_state) { + speex_echo_state_destroy(denoiser->echo_state); + denoiser->echo_state = tsk_null; + } + TSK_FREE(denoiser->echo_output_frame); + + return 0; } @@ -247,64 +246,62 @@ static int tdav_speex_denoise_close(tmedia_denoise_t* self) /* constructor */ static tsk_object_t* tdav_speex_denoise_ctor(tsk_object_t * self, va_list * app) { - tdav_speex_denoise_t *denoise = self; - if(denoise){ - /* init base */ - tmedia_denoise_init(TMEDIA_DENOISE(denoise)); - /* init self */ - - TSK_DEBUG_INFO("Create SpeexDSP denoiser"); - } - return self; + tdav_speex_denoise_t *denoise = self; + if(denoise) { + /* init base */ + tmedia_denoise_init(TMEDIA_DENOISE(denoise)); + /* init self */ + + TSK_DEBUG_INFO("Create SpeexDSP denoiser"); + } + return self; } /* destructor */ static tsk_object_t* tdav_speex_denoise_dtor(tsk_object_t * self) -{ - tdav_speex_denoise_t *denoise = self; - if(denoise){ - /* deinit base */ - tmedia_denoise_deinit(TMEDIA_DENOISE(denoise)); - /* deinit self */ - if(denoise->preprocess_state_record){ - speex_preprocess_state_destroy(denoise->preprocess_state_record); - denoise->preprocess_state_record = tsk_null; - } - if(denoise->preprocess_state_playback){ - speex_preprocess_state_destroy(denoise->preprocess_state_playback); - denoise->preprocess_state_playback = tsk_null; - } - if(denoise->echo_state){ - speex_echo_state_destroy(denoise->echo_state); - denoise->echo_state = tsk_null; - } - TSK_FREE(denoise->echo_output_frame); - - TSK_DEBUG_INFO("*** SpeexDSP denoiser destroyed ***"); - } - - return self; +{ + tdav_speex_denoise_t *denoise = self; + if(denoise) { + /* deinit base */ + tmedia_denoise_deinit(TMEDIA_DENOISE(denoise)); + /* deinit self */ + if(denoise->preprocess_state_record) { + speex_preprocess_state_destroy(denoise->preprocess_state_record); + denoise->preprocess_state_record = tsk_null; + } + if(denoise->preprocess_state_playback) { + speex_preprocess_state_destroy(denoise->preprocess_state_playback); + denoise->preprocess_state_playback = tsk_null; + } + if(denoise->echo_state) { + speex_echo_state_destroy(denoise->echo_state); + denoise->echo_state = tsk_null; + } + TSK_FREE(denoise->echo_output_frame); + + TSK_DEBUG_INFO("*** SpeexDSP denoiser destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_speex_denoise_def_s = -{ - sizeof(tdav_speex_denoise_t), - tdav_speex_denoise_ctor, - tdav_speex_denoise_dtor, - tsk_null, +static const tsk_object_def_t tdav_speex_denoise_def_s = { + sizeof(tdav_speex_denoise_t), + tdav_speex_denoise_ctor, + tdav_speex_denoise_dtor, + tsk_null, }; /* plugin definition*/ -static const tmedia_denoise_plugin_def_t tdav_speex_denoise_plugin_def_s = -{ - &tdav_speex_denoise_def_s, +static const tmedia_denoise_plugin_def_t tdav_speex_denoise_plugin_def_s = { + &tdav_speex_denoise_def_s, - "Audio Denoiser based on SpeexDSP", + "Audio Denoiser based on SpeexDSP", - tdav_speex_denoise_set, - tdav_speex_denoise_open, - tdav_speex_denoise_echo_playback, - tdav_speex_denoise_process_record, - tdav_speex_denoise_process_playback, - tdav_speex_denoise_close, + tdav_speex_denoise_set, + tdav_speex_denoise_open, + tdav_speex_denoise_echo_playback, + tdav_speex_denoise_process_record, + tdav_speex_denoise_process_playback, + tdav_speex_denoise_close, }; const tmedia_denoise_plugin_def_t *tdav_speex_denoise_plugin_def_t = &tdav_speex_denoise_plugin_def_s; diff --git a/tinyDAV/src/audio/tdav_speex_jitterbuffer.c b/tinyDAV/src/audio/tdav_speex_jitterbuffer.c index d4639b9..a83f49b 100755 --- a/tinyDAV/src/audio/tdav_speex_jitterbuffer.c +++ b/tinyDAV/src/audio/tdav_speex_jitterbuffer.c @@ -36,221 +36,220 @@ #include <speex/speex_jitter.h> /** Speex JitterBuffer*/ -typedef struct tdav_speex_jitterBuffer_s -{ - TMEDIA_DECLARE_JITTER_BUFFER; - - JitterBuffer* state; - uint32_t rate; - uint32_t frame_duration; - uint32_t channels; - uint32_t x_data_size; // expected data size - uint16_t fake_seqnum; // if ptime mismatch then, reassembled pkt will have invalid seqnum - struct { - uint8_t* ptr; - tsk_size_t size; - tsk_size_t index; - } buff; - - uint64_t num_pkt_in; // Number of incoming pkts since the last reset - uint64_t num_pkt_miss; // Number of times we got consecutive "JITTER_BUFFER_MISSING" results - uint64_t num_pkt_miss_max; // Max value for "num_pkt_miss" before reset()ing the jitter buffer +typedef struct tdav_speex_jitterBuffer_s { + TMEDIA_DECLARE_JITTER_BUFFER; + + JitterBuffer* state; + uint32_t rate; + uint32_t frame_duration; + uint32_t channels; + uint32_t x_data_size; // expected data size + uint16_t fake_seqnum; // if ptime mismatch then, reassembled pkt will have invalid seqnum + struct { + uint8_t* ptr; + tsk_size_t size; + tsk_size_t index; + } buff; + + uint64_t num_pkt_in; // Number of incoming pkts since the last reset + uint64_t num_pkt_miss; // Number of times we got consecutive "JITTER_BUFFER_MISSING" results + uint64_t num_pkt_miss_max; // Max value for "num_pkt_miss" before reset()ing the jitter buffer } tdav_speex_jitterbuffer_t; static int tdav_speex_jitterbuffer_set(tmedia_jitterbuffer_t *self, const tmedia_param_t* param) { - TSK_DEBUG_ERROR("Not implemented"); - return -2; + TSK_DEBUG_ERROR("Not implemented"); + return -2; } static int tdav_speex_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate, uint32_t channels) { - tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; - spx_int32_t tmp; - - TSK_DEBUG_INFO("Open speex jb (ptime=%u, rate=%u)", frame_duration, rate); - - if (!(jitterbuffer->state = jitter_buffer_init((int)frame_duration))) { - TSK_DEBUG_ERROR("jitter_buffer_init() failed"); - return -2; - } - jitterbuffer->rate = rate; - jitterbuffer->frame_duration = frame_duration; - jitterbuffer->channels = channels; - jitterbuffer->x_data_size = ((frame_duration * jitterbuffer->rate) / 500) << (channels == 2 ? 1 : 0); - - jitterbuffer->num_pkt_in = 0; - jitterbuffer->num_pkt_miss = 0; - jitterbuffer->num_pkt_miss_max = (1000 / frame_duration) * 2; // 2 seconds missing --> "Houston, we have a problem" - - jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_GET_MARGIN, &tmp); - TSK_DEBUG_INFO("Default Jitter buffer margin=%d", tmp); - jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_GET_MAX_LATE_RATE, &tmp); - TSK_DEBUG_INFO("Default Jitter max late rate=%d", tmp); - - if ((tmp = tmedia_defaults_get_jb_margin()) >= 0) { - jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_SET_MARGIN, &tmp); - TSK_DEBUG_INFO("New Jitter buffer margin=%d", tmp); - } - if ((tmp = tmedia_defaults_get_jb_max_late_rate()) >= 0) { - jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_SET_MAX_LATE_RATE, &tmp); - TSK_DEBUG_INFO("New Jitter buffer max late rate=%d", tmp); - } - - return 0; + tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; + spx_int32_t tmp; + + TSK_DEBUG_INFO("Open speex jb (ptime=%u, rate=%u)", frame_duration, rate); + + if (!(jitterbuffer->state = jitter_buffer_init((int)frame_duration))) { + TSK_DEBUG_ERROR("jitter_buffer_init() failed"); + return -2; + } + jitterbuffer->rate = rate; + jitterbuffer->frame_duration = frame_duration; + jitterbuffer->channels = channels; + jitterbuffer->x_data_size = ((frame_duration * jitterbuffer->rate) / 500) << (channels == 2 ? 1 : 0); + + jitterbuffer->num_pkt_in = 0; + jitterbuffer->num_pkt_miss = 0; + jitterbuffer->num_pkt_miss_max = (1000 / frame_duration) * 2; // 2 seconds missing --> "Houston, we have a problem" + + jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_GET_MARGIN, &tmp); + TSK_DEBUG_INFO("Default Jitter buffer margin=%d", tmp); + jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_GET_MAX_LATE_RATE, &tmp); + TSK_DEBUG_INFO("Default Jitter max late rate=%d", tmp); + + if ((tmp = tmedia_defaults_get_jb_margin()) >= 0) { + jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_SET_MARGIN, &tmp); + TSK_DEBUG_INFO("New Jitter buffer margin=%d", tmp); + } + if ((tmp = tmedia_defaults_get_jb_max_late_rate()) >= 0) { + jitter_buffer_ctl(jitterbuffer->state, JITTER_BUFFER_SET_MAX_LATE_RATE, &tmp); + TSK_DEBUG_INFO("New Jitter buffer max late rate=%d", tmp); + } + + return 0; } static int tdav_speex_jitterbuffer_tick(tmedia_jitterbuffer_t* self) { - tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; - if (!jitterbuffer->state) { - TSK_DEBUG_ERROR("Invalid state"); - return -1; - } - jitter_buffer_tick(jitterbuffer->state); - return 0; + tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; + if (!jitterbuffer->state) { + TSK_DEBUG_ERROR("Invalid state"); + return -1; + } + jitter_buffer_tick(jitterbuffer->state); + return 0; } static int tdav_speex_jitterbuffer_put(tmedia_jitterbuffer_t* self, void* data, tsk_size_t data_size, const tsk_object_t* proto_hdr) { - tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; - const trtp_rtp_header_t* rtp_hdr; - JitterBufferPacket jb_packet; - static uint16_t seq_num = 0; - - if (!data || !data_size || !proto_hdr) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if (!jb->state) { - TSK_DEBUG_ERROR("Invalid state"); - return -2; - } - - rtp_hdr = TRTP_RTP_HEADER(proto_hdr); - - jb_packet.user_data = 0; - jb_packet.span = jb->frame_duration; - jb_packet.len = jb->x_data_size; - - if (jb->x_data_size == data_size) { /* ptime match */ - jb_packet.data = data; - jb_packet.sequence = rtp_hdr->seq_num; - jb_packet.timestamp = (rtp_hdr->seq_num * jb_packet.span); - jitter_buffer_put(jb->state, &jb_packet); - } - else { /* ptime mismatch */ - tsk_size_t i; - jb_packet.sequence = 0; // Ignore - if ((jb->buff.index + data_size) > jb->buff.size) { - if (!(jb->buff.ptr = tsk_realloc(jb->buff.ptr, (jb->buff.index + data_size)))) { - jb->buff.size = 0; - jb->buff.index = 0; - return 0; - } - jb->buff.size = (jb->buff.index + data_size); - } - - memcpy(&jb->buff.ptr[jb->buff.index], data, data_size); - jb->buff.index += data_size; - - if (jb->buff.index >= jb->x_data_size) { - tsk_size_t copied = 0; - for (i = 0; (i + jb->x_data_size) <= jb->buff.index; i += jb->x_data_size) { - jb_packet.data = (char*)&jb->buff.ptr[i]; - jb_packet.timestamp = (++jb->fake_seqnum * jb_packet.span);// reassembled pkt will have fake seqnum - jitter_buffer_put(jb->state, &jb_packet); - copied += jb->x_data_size; - } - if (copied == jb->buff.index) { - // all copied - jb->buff.index = 0; - } - else { - memmove(&jb->buff.ptr[0], &jb->buff.ptr[copied], (jb->buff.index - copied)); - jb->buff.index -= copied; - } - } - } - ++jb->num_pkt_in; - - return 0; + tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; + const trtp_rtp_header_t* rtp_hdr; + JitterBufferPacket jb_packet; + static uint16_t seq_num = 0; + + if (!data || !data_size || !proto_hdr) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if (!jb->state) { + TSK_DEBUG_ERROR("Invalid state"); + return -2; + } + + rtp_hdr = TRTP_RTP_HEADER(proto_hdr); + + jb_packet.user_data = 0; + jb_packet.span = jb->frame_duration; + jb_packet.len = jb->x_data_size; + + if (jb->x_data_size == data_size) { /* ptime match */ + jb_packet.data = data; + jb_packet.sequence = rtp_hdr->seq_num; + jb_packet.timestamp = (rtp_hdr->seq_num * jb_packet.span); + jitter_buffer_put(jb->state, &jb_packet); + } + else { /* ptime mismatch */ + tsk_size_t i; + jb_packet.sequence = 0; // Ignore + if ((jb->buff.index + data_size) > jb->buff.size) { + if (!(jb->buff.ptr = tsk_realloc(jb->buff.ptr, (jb->buff.index + data_size)))) { + jb->buff.size = 0; + jb->buff.index = 0; + return 0; + } + jb->buff.size = (jb->buff.index + data_size); + } + + memcpy(&jb->buff.ptr[jb->buff.index], data, data_size); + jb->buff.index += data_size; + + if (jb->buff.index >= jb->x_data_size) { + tsk_size_t copied = 0; + for (i = 0; (i + jb->x_data_size) <= jb->buff.index; i += jb->x_data_size) { + jb_packet.data = (char*)&jb->buff.ptr[i]; + jb_packet.timestamp = (++jb->fake_seqnum * jb_packet.span);// reassembled pkt will have fake seqnum + jitter_buffer_put(jb->state, &jb_packet); + copied += jb->x_data_size; + } + if (copied == jb->buff.index) { + // all copied + jb->buff.index = 0; + } + else { + memmove(&jb->buff.ptr[0], &jb->buff.ptr[copied], (jb->buff.index - copied)); + jb->buff.index -= copied; + } + } + } + ++jb->num_pkt_in; + + return 0; } static tsk_size_t tdav_speex_jitterbuffer_get(tmedia_jitterbuffer_t* self, void* out_data, tsk_size_t out_size) { - tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; - JitterBufferPacket jb_packet; - int ret, miss = 0; - tsk_size_t ret_size = 0; - - if (!out_data || !out_size) { - TSK_DEBUG_ERROR("Invalid parameter"); - return 0; - } - if (!jb->state) { - TSK_DEBUG_ERROR("Invalid state"); - return 0; - } - if (jb->x_data_size != out_size) { // consumer must request PTIME data - TSK_DEBUG_WARN("%d not expected as frame size. %u<>%u", out_size, jb->frame_duration, (out_size * 500) / jb->rate); - return 0; - } - - jb_packet.data = out_data; - jb_packet.len = (spx_uint32_t)out_size; - - if ((ret = jitter_buffer_get(jb->state, &jb_packet, jb->frame_duration/*(out_size * 500)/jb->rate*/, tsk_null)) != JITTER_BUFFER_OK) { - ++jb->num_pkt_miss; - switch (ret) { - case JITTER_BUFFER_MISSING: - /*TSK_DEBUG_INFO("JITTER_BUFFER_MISSING - %d", ret);*/ - if (jb->num_pkt_miss > jb->num_pkt_miss_max /*too much missing pkts*/ && jb->num_pkt_in > jb->num_pkt_miss_max/*we're really receiving pkts*/) { - jb->num_pkt_miss = 0; - self->plugin->reset(self); - TSK_DEBUG_WARN("Too much missing audio pkts"); - } - break; - case JITTER_BUFFER_INSERTION: - /*TSK_DEBUG_INFO("JITTER_BUFFER_INSERTION - %d", ret);*/ - break; - default: - TSK_DEBUG_INFO("jitter_buffer_get() failed - %d", ret); - break; - } - // jitter_buffer_update_delay(jb->state, &jb_packet, NULL); - //return 0; - } - else { - jb->num_pkt_miss = 0; // reset - ret_size = jb_packet.len; - } - //jitter_buffer_update_delay(jb->state, &jb_packet, NULL); - - return ret_size; + tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; + JitterBufferPacket jb_packet; + int ret, miss = 0; + tsk_size_t ret_size = 0; + + if (!out_data || !out_size) { + TSK_DEBUG_ERROR("Invalid parameter"); + return 0; + } + if (!jb->state) { + TSK_DEBUG_ERROR("Invalid state"); + return 0; + } + if (jb->x_data_size != out_size) { // consumer must request PTIME data + TSK_DEBUG_WARN("%d not expected as frame size. %u<>%u", out_size, jb->frame_duration, (out_size * 500) / jb->rate); + return 0; + } + + jb_packet.data = out_data; + jb_packet.len = (spx_uint32_t)out_size; + + if ((ret = jitter_buffer_get(jb->state, &jb_packet, jb->frame_duration/*(out_size * 500)/jb->rate*/, tsk_null)) != JITTER_BUFFER_OK) { + ++jb->num_pkt_miss; + switch (ret) { + case JITTER_BUFFER_MISSING: + /*TSK_DEBUG_INFO("JITTER_BUFFER_MISSING - %d", ret);*/ + if (jb->num_pkt_miss > jb->num_pkt_miss_max /*too much missing pkts*/ && jb->num_pkt_in > jb->num_pkt_miss_max/*we're really receiving pkts*/) { + jb->num_pkt_miss = 0; + self->plugin->reset(self); + TSK_DEBUG_WARN("Too much missing audio pkts"); + } + break; + case JITTER_BUFFER_INSERTION: + /*TSK_DEBUG_INFO("JITTER_BUFFER_INSERTION - %d", ret);*/ + break; + default: + TSK_DEBUG_INFO("jitter_buffer_get() failed - %d", ret); + break; + } + // jitter_buffer_update_delay(jb->state, &jb_packet, NULL); + //return 0; + } + else { + jb->num_pkt_miss = 0; // reset + ret_size = jb_packet.len; + } + //jitter_buffer_update_delay(jb->state, &jb_packet, NULL); + + return ret_size; } static int tdav_speex_jitterbuffer_reset(tmedia_jitterbuffer_t* self) { - tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; - if (jb->state) { - jitter_buffer_reset(jb->state); - } - jb->num_pkt_in = 0; - jb->num_pkt_miss = 0; - return 0; + tdav_speex_jitterbuffer_t *jb = (tdav_speex_jitterbuffer_t *)self; + if (jb->state) { + jitter_buffer_reset(jb->state); + } + jb->num_pkt_in = 0; + jb->num_pkt_miss = 0; + return 0; } static int tdav_speex_jitterbuffer_close(tmedia_jitterbuffer_t* self) { - tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; - if (jitterbuffer->state) { - jitter_buffer_destroy(jitterbuffer->state); - jitterbuffer->state = tsk_null; - } - return 0; + tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self; + if (jitterbuffer->state) { + jitter_buffer_destroy(jitterbuffer->state); + jitterbuffer->state = tsk_null; + } + return 0; } @@ -262,56 +261,54 @@ static int tdav_speex_jitterbuffer_close(tmedia_jitterbuffer_t* self) /* constructor */ static tsk_object_t* tdav_speex_jitterbuffer_ctor(tsk_object_t * self, va_list * app) { - tdav_speex_jitterbuffer_t *jitterbuffer = self; - TSK_DEBUG_INFO("Create SpeexDSP jitter buffer"); - if (jitterbuffer){ - /* init base */ - tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(jitterbuffer)); - /* init self */ - } - return self; + tdav_speex_jitterbuffer_t *jitterbuffer = self; + TSK_DEBUG_INFO("Create SpeexDSP jitter buffer"); + if (jitterbuffer) { + /* init base */ + tmedia_jitterbuffer_init(TMEDIA_JITTER_BUFFER(jitterbuffer)); + /* init self */ + } + return self; } /* destructor */ static tsk_object_t* tdav_speex_jitterbuffer_dtor(tsk_object_t * self) { - tdav_speex_jitterbuffer_t *jb = self; - if (jb){ - /* deinit base */ - tmedia_jitterbuffer_deinit(TMEDIA_JITTER_BUFFER(jb)); - /* deinit self */ - if (jb->state){ - jitter_buffer_destroy(jb->state); - jb->state = tsk_null; - } - TSK_FREE(jb->buff.ptr); - - TSK_DEBUG_INFO("*** SpeexDSP jb destroyed ***"); - } - - return self; + tdav_speex_jitterbuffer_t *jb = self; + if (jb) { + /* deinit base */ + tmedia_jitterbuffer_deinit(TMEDIA_JITTER_BUFFER(jb)); + /* deinit self */ + if (jb->state) { + jitter_buffer_destroy(jb->state); + jb->state = tsk_null; + } + TSK_FREE(jb->buff.ptr); + + TSK_DEBUG_INFO("*** SpeexDSP jb destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_speex_jitterbuffer_def_s = -{ - sizeof(tdav_speex_jitterbuffer_t), - tdav_speex_jitterbuffer_ctor, - tdav_speex_jitterbuffer_dtor, - tsk_null, +static const tsk_object_def_t tdav_speex_jitterbuffer_def_s = { + sizeof(tdav_speex_jitterbuffer_t), + tdav_speex_jitterbuffer_ctor, + tdav_speex_jitterbuffer_dtor, + tsk_null, }; /* plugin definition*/ -static const tmedia_jitterbuffer_plugin_def_t tdav_speex_jitterbuffer_plugin_def_s = -{ - &tdav_speex_jitterbuffer_def_s, - tmedia_audio, - "Audio JitterBuffer based on Speex", - - tdav_speex_jitterbuffer_set, - tdav_speex_jitterbuffer_open, - tdav_speex_jitterbuffer_tick, - tdav_speex_jitterbuffer_put, - tdav_speex_jitterbuffer_get, - tdav_speex_jitterbuffer_reset, - tdav_speex_jitterbuffer_close, +static const tmedia_jitterbuffer_plugin_def_t tdav_speex_jitterbuffer_plugin_def_s = { + &tdav_speex_jitterbuffer_def_s, + tmedia_audio, + "Audio JitterBuffer based on Speex", + + tdav_speex_jitterbuffer_set, + tdav_speex_jitterbuffer_open, + tdav_speex_jitterbuffer_tick, + tdav_speex_jitterbuffer_put, + tdav_speex_jitterbuffer_get, + tdav_speex_jitterbuffer_reset, + tdav_speex_jitterbuffer_close, }; const tmedia_jitterbuffer_plugin_def_t *tdav_speex_jitterbuffer_plugin_def_t = &tdav_speex_jitterbuffer_plugin_def_s; diff --git a/tinyDAV/src/audio/tdav_speex_resampler.c b/tinyDAV/src/audio/tdav_speex_resampler.c index f71ddd2..75c51e5 100755 --- a/tinyDAV/src/audio/tdav_speex_resampler.c +++ b/tinyDAV/src/audio/tdav_speex_resampler.c @@ -29,168 +29,167 @@ #define TDAV_SPEEX_RESAMPLER_MAX_QUALITY 10 /** Speex resampler*/ -typedef struct tdav_speex_resampler_s -{ - TMEDIA_DECLARE_RESAMPLER; +typedef struct tdav_speex_resampler_s { + TMEDIA_DECLARE_RESAMPLER; - tsk_size_t in_size; - tsk_size_t out_size; - uint32_t in_channels; - uint32_t out_channels; - uint32_t bytes_per_sample; + tsk_size_t in_size; + tsk_size_t out_size; + uint32_t in_channels; + uint32_t out_channels; + uint32_t bytes_per_sample; - struct{ - void* ptr; - tsk_size_t size_in_samples; - } tmp_buffer; + struct { + void* ptr; + tsk_size_t size_in_samples; + } tmp_buffer; - SpeexResamplerState *state; + SpeexResamplerState *state; } tdav_speex_resampler_t; static int tdav_speex_resampler_open(tmedia_resampler_t* self, uint32_t in_freq, uint32_t out_freq, uint32_t frame_duration, uint32_t in_channels, uint32_t out_channels, uint32_t quality, uint32_t bits_per_sample) { - tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; - int ret = 0; - uint32_t bytes_per_sample = (bits_per_sample >> 3); - - if (in_channels != 1 && in_channels != 2) { - TSK_DEBUG_ERROR("%d not valid as input channel", in_channels); - return -1; - } - if (out_channels != 1 && out_channels != 2) { - TSK_DEBUG_ERROR("%d not valid as output channel", out_channels); - return -1; - } - if (bytes_per_sample != sizeof(spx_int16_t) && bytes_per_sample != sizeof(float)) { - TSK_DEBUG_ERROR("%d not valid as bits_per_sample", bits_per_sample); - return -1; - } - - if (!(resampler->state = speex_resampler_init(in_channels, in_freq, out_freq, TSK_CLAMP(0, quality, TDAV_SPEEX_RESAMPLER_MAX_QUALITY), &ret))) { - TSK_DEBUG_ERROR("speex_resampler_init() returned %d", ret); - return -2; - } - - resampler->bytes_per_sample = bytes_per_sample; - resampler->in_size = ((in_freq * frame_duration) / 1000) << (in_channels == 2 ? 1 : 0); - resampler->out_size = ((out_freq * frame_duration) / 1000) << (out_channels == 2 ? 1 : 0); - resampler->in_channels = in_channels; - resampler->out_channels = out_channels; - - if (in_channels != out_channels) { - resampler->tmp_buffer.size_in_samples = ((TSK_MAX(in_freq, out_freq) * frame_duration) / 1000) << (TSK_MAX(in_channels, out_channels) == 2 ? 1 : 0); - if (!(resampler->tmp_buffer.ptr = tsk_realloc(resampler->tmp_buffer.ptr, resampler->tmp_buffer.size_in_samples * resampler->bytes_per_sample))) { - resampler->tmp_buffer.size_in_samples = 0; - return -2; - } - } - - return 0; + tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; + int ret = 0; + uint32_t bytes_per_sample = (bits_per_sample >> 3); + + if (in_channels != 1 && in_channels != 2) { + TSK_DEBUG_ERROR("%d not valid as input channel", in_channels); + return -1; + } + if (out_channels != 1 && out_channels != 2) { + TSK_DEBUG_ERROR("%d not valid as output channel", out_channels); + return -1; + } + if (bytes_per_sample != sizeof(spx_int16_t) && bytes_per_sample != sizeof(float)) { + TSK_DEBUG_ERROR("%d not valid as bits_per_sample", bits_per_sample); + return -1; + } + + if (!(resampler->state = speex_resampler_init(in_channels, in_freq, out_freq, TSK_CLAMP(0, quality, TDAV_SPEEX_RESAMPLER_MAX_QUALITY), &ret))) { + TSK_DEBUG_ERROR("speex_resampler_init() returned %d", ret); + return -2; + } + + resampler->bytes_per_sample = bytes_per_sample; + resampler->in_size = ((in_freq * frame_duration) / 1000) << (in_channels == 2 ? 1 : 0); + resampler->out_size = ((out_freq * frame_duration) / 1000) << (out_channels == 2 ? 1 : 0); + resampler->in_channels = in_channels; + resampler->out_channels = out_channels; + + if (in_channels != out_channels) { + resampler->tmp_buffer.size_in_samples = ((TSK_MAX(in_freq, out_freq) * frame_duration) / 1000) << (TSK_MAX(in_channels, out_channels) == 2 ? 1 : 0); + if (!(resampler->tmp_buffer.ptr = tsk_realloc(resampler->tmp_buffer.ptr, resampler->tmp_buffer.size_in_samples * resampler->bytes_per_sample))) { + resampler->tmp_buffer.size_in_samples = 0; + return -2; + } + } + + return 0; } static tsk_size_t tdav_speex_resampler_process(tmedia_resampler_t* self, const void* in_data, tsk_size_t in_size_in_sample, void* out_data, tsk_size_t out_size_in_sample) { - tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; - int err = RESAMPLER_ERR_SUCCESS; - spx_uint32_t _out_size_in_sample = (spx_uint32_t)out_size_in_sample; - if (!resampler->state || !out_data) { - TSK_DEBUG_ERROR("Invalid parameter"); - return 0; - } - - if (in_size_in_sample != resampler->in_size) { - TSK_DEBUG_ERROR("Input data has wrong size"); - return 0; - } - - if (out_size_in_sample < resampler->out_size) { - TSK_DEBUG_ERROR("Output data is too short"); - return 0; - } - - if (resampler->in_channels == resampler->out_channels) { - if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { - err = speex_resampler_process_int(resampler->state, 0, - (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, - (spx_int16_t *)out_data, &_out_size_in_sample); - } - else { - err = speex_resampler_process_float(resampler->state, 0, - (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, - (float *)out_data, &_out_size_in_sample); - } - } - else { - spx_uint32_t i, j; - // in_channels = 1, out_channels = 2 - if (resampler->in_channels == 1) { - if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { - err = speex_resampler_process_int(resampler->state, 0, (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, resampler->tmp_buffer.ptr, &_out_size_in_sample); - if (err == RESAMPLER_ERR_SUCCESS) { - spx_int16_t* pout_data = (spx_int16_t*)(out_data); - for (i = 0, j = 0; i < _out_size_in_sample; ++i, j += 2) { - pout_data[j] = pout_data[j + 1] = *(((const spx_int16_t*)resampler->tmp_buffer.ptr) + i); - } - } - } - else { - err = speex_resampler_process_float(resampler->state, 0, (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, resampler->tmp_buffer.ptr, &_out_size_in_sample); - if (err == RESAMPLER_ERR_SUCCESS) { - float* pout_data = (float*)(out_data); - for (i = 0, j = 0; i < _out_size_in_sample; ++i, j += 2) { - pout_data[j] = pout_data[j + 1] = *(((const float*)resampler->tmp_buffer.ptr) + i); - } - } - } - - } - else { - // in_channels = 2, out_channels = 1 - spx_uint32_t _out_size2_in_sample = (_out_size_in_sample << 1); - if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { - err = speex_resampler_process_int(resampler->state, 0, - (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, - (spx_int16_t *)resampler->tmp_buffer.ptr, &_out_size2_in_sample); - if (err == RESAMPLER_ERR_SUCCESS) { - spx_int16_t* pout_data = (spx_int16_t*)(out_data); - _out_size_in_sample = (spx_uint32_t)resampler->out_size; - for (i = 0, j = 0; j < _out_size2_in_sample; ++i, j += 2) { - pout_data[i] = *(((const spx_int16_t*)resampler->tmp_buffer.ptr) + j); - } - } - } - else { - err = speex_resampler_process_float(resampler->state, 0, - (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, - (float *)resampler->tmp_buffer.ptr, &_out_size2_in_sample); - if (err == RESAMPLER_ERR_SUCCESS) { - float* pout_data = (float*)(out_data); - for (i = 0, j = 0; j < _out_size2_in_sample; ++i, j += 2) { - pout_data[i] = *(((const float*)resampler->tmp_buffer.ptr) + j); - } - } - } - } - } - - if (err != RESAMPLER_ERR_SUCCESS) { - TSK_DEBUG_ERROR("speex_resampler_process_int() failed with error code %d", err); - return 0; - } - return (tsk_size_t)_out_size_in_sample; + tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; + int err = RESAMPLER_ERR_SUCCESS; + spx_uint32_t _out_size_in_sample = (spx_uint32_t)out_size_in_sample; + if (!resampler->state || !out_data) { + TSK_DEBUG_ERROR("Invalid parameter"); + return 0; + } + + if (in_size_in_sample != resampler->in_size) { + TSK_DEBUG_ERROR("Input data has wrong size"); + return 0; + } + + if (out_size_in_sample < resampler->out_size) { + TSK_DEBUG_ERROR("Output data is too short"); + return 0; + } + + if (resampler->in_channels == resampler->out_channels) { + if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { + err = speex_resampler_process_int(resampler->state, 0, + (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, + (spx_int16_t *)out_data, &_out_size_in_sample); + } + else { + err = speex_resampler_process_float(resampler->state, 0, + (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, + (float *)out_data, &_out_size_in_sample); + } + } + else { + spx_uint32_t i, j; + // in_channels = 1, out_channels = 2 + if (resampler->in_channels == 1) { + if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { + err = speex_resampler_process_int(resampler->state, 0, (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, resampler->tmp_buffer.ptr, &_out_size_in_sample); + if (err == RESAMPLER_ERR_SUCCESS) { + spx_int16_t* pout_data = (spx_int16_t*)(out_data); + for (i = 0, j = 0; i < _out_size_in_sample; ++i, j += 2) { + pout_data[j] = pout_data[j + 1] = *(((const spx_int16_t*)resampler->tmp_buffer.ptr) + i); + } + } + } + else { + err = speex_resampler_process_float(resampler->state, 0, (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, resampler->tmp_buffer.ptr, &_out_size_in_sample); + if (err == RESAMPLER_ERR_SUCCESS) { + float* pout_data = (float*)(out_data); + for (i = 0, j = 0; i < _out_size_in_sample; ++i, j += 2) { + pout_data[j] = pout_data[j + 1] = *(((const float*)resampler->tmp_buffer.ptr) + i); + } + } + } + + } + else { + // in_channels = 2, out_channels = 1 + spx_uint32_t _out_size2_in_sample = (_out_size_in_sample << 1); + if (resampler->bytes_per_sample == sizeof(spx_int16_t)) { + err = speex_resampler_process_int(resampler->state, 0, + (const spx_int16_t *)in_data, (spx_uint32_t *)&in_size_in_sample, + (spx_int16_t *)resampler->tmp_buffer.ptr, &_out_size2_in_sample); + if (err == RESAMPLER_ERR_SUCCESS) { + spx_int16_t* pout_data = (spx_int16_t*)(out_data); + _out_size_in_sample = (spx_uint32_t)resampler->out_size; + for (i = 0, j = 0; j < _out_size2_in_sample; ++i, j += 2) { + pout_data[i] = *(((const spx_int16_t*)resampler->tmp_buffer.ptr) + j); + } + } + } + else { + err = speex_resampler_process_float(resampler->state, 0, + (const float *)in_data, (spx_uint32_t *)&in_size_in_sample, + (float *)resampler->tmp_buffer.ptr, &_out_size2_in_sample); + if (err == RESAMPLER_ERR_SUCCESS) { + float* pout_data = (float*)(out_data); + for (i = 0, j = 0; j < _out_size2_in_sample; ++i, j += 2) { + pout_data[i] = *(((const float*)resampler->tmp_buffer.ptr) + j); + } + } + } + } + } + + if (err != RESAMPLER_ERR_SUCCESS) { + TSK_DEBUG_ERROR("speex_resampler_process_int() failed with error code %d", err); + return 0; + } + return (tsk_size_t)_out_size_in_sample; } static int tdav_speex_resampler_close(tmedia_resampler_t* self) { - tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; + tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; - if (resampler->state) { - speex_resampler_destroy(resampler->state); - resampler->state = tsk_null; - } - return 0; + if (resampler->state) { + speex_resampler_destroy(resampler->state); + resampler->state = tsk_null; + } + return 0; } @@ -202,51 +201,49 @@ static int tdav_speex_resampler_close(tmedia_resampler_t* self) /* constructor */ static tsk_object_t* tdav_speex_resampler_ctor(tsk_object_t * self, va_list * app) { - tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; - if (resampler){ - /* init base */ - tmedia_resampler_init(TMEDIA_RESAMPLER(resampler)); - /* init self */ - } - return self; + tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; + if (resampler) { + /* init base */ + tmedia_resampler_init(TMEDIA_RESAMPLER(resampler)); + /* init self */ + } + return self; } /* destructor */ static tsk_object_t* tdav_speex_resampler_dtor(tsk_object_t * self) { - tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; - if (resampler){ - /* deinit base */ - tmedia_resampler_deinit(TMEDIA_RESAMPLER(resampler)); - /* deinit self */ - if (resampler->state) { - speex_resampler_destroy(resampler->state); - resampler->state = tsk_null; - } - TSK_FREE(resampler->tmp_buffer.ptr); - - TSK_DEBUG_INFO("*** SpeexDSP resampler (plugin) destroyed ***"); - } - - return self; + tdav_speex_resampler_t *resampler = (tdav_speex_resampler_t *)self; + if (resampler) { + /* deinit base */ + tmedia_resampler_deinit(TMEDIA_RESAMPLER(resampler)); + /* deinit self */ + if (resampler->state) { + speex_resampler_destroy(resampler->state); + resampler->state = tsk_null; + } + TSK_FREE(resampler->tmp_buffer.ptr); + + TSK_DEBUG_INFO("*** SpeexDSP resampler (plugin) destroyed ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_speex_resampler_def_s = -{ - sizeof(tdav_speex_resampler_t), - tdav_speex_resampler_ctor, - tdav_speex_resampler_dtor, - tsk_null, +static const tsk_object_def_t tdav_speex_resampler_def_s = { + sizeof(tdav_speex_resampler_t), + tdav_speex_resampler_ctor, + tdav_speex_resampler_dtor, + tsk_null, }; /* plugin definition*/ -static const tmedia_resampler_plugin_def_t tdav_speex_resampler_plugin_def_s = -{ - &tdav_speex_resampler_def_s, +static const tmedia_resampler_plugin_def_t tdav_speex_resampler_plugin_def_s = { + &tdav_speex_resampler_def_s, - "Audio Resampler based on Speex", + "Audio Resampler based on Speex", - tdav_speex_resampler_open, - tdav_speex_resampler_process, - tdav_speex_resampler_close, + tdav_speex_resampler_open, + tdav_speex_resampler_process, + tdav_speex_resampler_close, }; const tmedia_resampler_plugin_def_t *tdav_speex_resampler_plugin_def_t = &tdav_speex_resampler_plugin_def_s; diff --git a/tinyDAV/src/audio/tdav_webrtc_denoise.c b/tinyDAV/src/audio/tdav_webrtc_denoise.c index 598470a..c69ab9e 100755 --- a/tinyDAV/src/audio/tdav_webrtc_denoise.c +++ b/tinyDAV/src/audio/tdav_webrtc_denoise.c @@ -51,34 +51,32 @@ typedef int16_t sample_t; typedef float sample_t; #endif -typedef struct tdav_webrtc_pin_xs -{ - uint32_t n_duration; - uint32_t n_rate; - uint32_t n_channels; - uint32_t n_sample_size; +typedef struct tdav_webrtc_pin_xs { + uint32_t n_duration; + uint32_t n_rate; + uint32_t n_channels; + uint32_t n_sample_size; } tdav_webrtc_pin_xt; -typedef struct tdav_webrtc_resampler_s -{ - TSK_DECLARE_OBJECT; - - tmedia_resampler_t* p_resampler; - void* p_bufftmp_ptr; // used to convert float <->int16 - tsk_size_t n_bufftmp_size_in_bytes; - - struct { - tdav_webrtc_pin_xt x_pin; - tsk_size_t n_buff_size_in_bytes; - tsk_size_t n_buff_size_in_samples; - } in; - struct { - tdav_webrtc_pin_xt x_pin; - void* p_buff_ptr; - tsk_size_t n_buff_size_in_bytes; - tsk_size_t n_buff_size_in_samples; - } out; +typedef struct tdav_webrtc_resampler_s { + TSK_DECLARE_OBJECT; + + tmedia_resampler_t* p_resampler; + void* p_bufftmp_ptr; // used to convert float <->int16 + tsk_size_t n_bufftmp_size_in_bytes; + + struct { + tdav_webrtc_pin_xt x_pin; + tsk_size_t n_buff_size_in_bytes; + tsk_size_t n_buff_size_in_samples; + } in; + struct { + tdav_webrtc_pin_xt x_pin; + void* p_buff_ptr; + tsk_size_t n_buff_size_in_bytes; + tsk_size_t n_buff_size_in_samples; + } out; } tdav_webrtc_resampler_t; @@ -86,447 +84,446 @@ static int _tdav_webrtc_resampler_create(const tdav_webrtc_pin_xt* p_pin_in, con static int _tdav_webrtc_resampler_process(tdav_webrtc_resampler_t* p_self, const void* p_buff_ptr, tsk_size_t n_buff_size_in_bytes); /** WebRTC denoiser (AEC, NS, AGC...) */ -typedef struct tdav_webrtc_denoise_s -{ - TMEDIA_DECLARE_DENOISE; +typedef struct tdav_webrtc_denoise_s { + TMEDIA_DECLARE_DENOISE; - void *AEC_inst; + void *AEC_inst; #if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER - SpeexPreprocessState *SpeexDenoiser_proc; + SpeexPreprocessState *SpeexDenoiser_proc; #else - TDAV_NsHandle *NS_inst; + TDAV_NsHandle *NS_inst; #endif - uint32_t echo_tail; - uint32_t echo_skew; - - struct { - tdav_webrtc_resampler_t* p_rpl_in2den; // input -> denoiser - tdav_webrtc_resampler_t* p_rpl_den2in; // denoiser -> input - } record; - struct { - tdav_webrtc_resampler_t* p_rpl_in2den; // input -> denoiser - tdav_webrtc_resampler_t* p_rpl_den2in; // denoiser -> input - } playback; - - struct { - uint32_t nb_samples_per_process; - uint32_t sampling_rate; - uint32_t channels; // always "1" - } neg; - - TSK_DECLARE_SAFEOBJ; + uint32_t echo_tail; + uint32_t echo_skew; + + struct { + tdav_webrtc_resampler_t* p_rpl_in2den; // input -> denoiser + tdav_webrtc_resampler_t* p_rpl_den2in; // denoiser -> input + } record; + struct { + tdav_webrtc_resampler_t* p_rpl_in2den; // input -> denoiser + tdav_webrtc_resampler_t* p_rpl_den2in; // denoiser -> input + } playback; + + struct { + uint32_t nb_samples_per_process; + uint32_t sampling_rate; + uint32_t channels; // always "1" + } neg; + + TSK_DECLARE_SAFEOBJ; } tdav_webrtc_denoise_t; static int tdav_webrtc_denoise_set(tmedia_denoise_t* _self, const tmedia_param_t* param) { - tdav_webrtc_denoise_t *self = (tdav_webrtc_denoise_t *)_self; - if (!self || !param) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "echo-tail")) { - int32_t echo_tail = *((int32_t*)param->value); - self->echo_tail = TSK_CLAMP(WEBRTC_MIN_ECHO_TAIL, echo_tail, WEBRTC_MAX_ECHO_TAIL); - TSK_DEBUG_INFO("set_echo_tail (%d->%d)", echo_tail, self->echo_tail); - return 0; - } - } - return -1; + tdav_webrtc_denoise_t *self = (tdav_webrtc_denoise_t *)_self; + if (!self || !param) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "echo-tail")) { + int32_t echo_tail = *((int32_t*)param->value); + self->echo_tail = TSK_CLAMP(WEBRTC_MIN_ECHO_TAIL, echo_tail, WEBRTC_MAX_ECHO_TAIL); + TSK_DEBUG_INFO("set_echo_tail (%d->%d)", echo_tail, self->echo_tail); + return 0; + } + } + return -1; } static int tdav_webrtc_denoise_open(tmedia_denoise_t* self, uint32_t record_frame_size_samples, uint32_t record_sampling_rate, uint32_t record_channels, uint32_t playback_frame_size_samples, uint32_t playback_sampling_rate, uint32_t playback_channels) { - tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; - int ret; - tdav_webrtc_pin_xt pin_record_in = { 0 }, pin_record_den = { 0 }, pin_playback_in = { 0 }, pin_playback_den = { 0 }; + tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; + int ret; + tdav_webrtc_pin_xt pin_record_in = { 0 }, pin_record_den = { 0 }, pin_playback_in = { 0 }, pin_playback_den = { 0 }; - if (!denoiser) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!denoiser) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if (denoiser->AEC_inst || + if (denoiser->AEC_inst || #if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER - denoiser->SpeexDenoiser_proc + denoiser->SpeexDenoiser_proc #else - denoiser->NS_inst + denoiser->NS_inst #endif - ){ - TSK_DEBUG_ERROR("Denoiser already initialized"); - return -2; - } - - denoiser->echo_tail = TSK_CLAMP(WEBRTC_MIN_ECHO_TAIL, TMEDIA_DENOISE(denoiser)->echo_tail, WEBRTC_MAX_ECHO_TAIL); - denoiser->echo_skew = TMEDIA_DENOISE(denoiser)->echo_skew; - TSK_DEBUG_INFO("echo_tail=%d, echo_skew=%d, echo_supp_enabled=%d, noise_supp_enabled=%d", denoiser->echo_tail, denoiser->echo_skew, self->echo_supp_enabled, self->noise_supp_enabled); - - // - // DENOISER - // + ) { + TSK_DEBUG_ERROR("Denoiser already initialized"); + return -2; + } + + denoiser->echo_tail = TSK_CLAMP(WEBRTC_MIN_ECHO_TAIL, TMEDIA_DENOISE(denoiser)->echo_tail, WEBRTC_MAX_ECHO_TAIL); + denoiser->echo_skew = TMEDIA_DENOISE(denoiser)->echo_skew; + TSK_DEBUG_INFO("echo_tail=%d, echo_skew=%d, echo_supp_enabled=%d, noise_supp_enabled=%d", denoiser->echo_tail, denoiser->echo_skew, self->echo_supp_enabled, self->noise_supp_enabled); + + // + // DENOISER + // #if TDAV_UNDER_MOBILE // AECM= [8-16]k, AEC=[8-32]k - denoiser->neg.sampling_rate = TSK_MIN(TSK_MAX(record_sampling_rate, playback_sampling_rate), 16000); + denoiser->neg.sampling_rate = TSK_MIN(TSK_MAX(record_sampling_rate, playback_sampling_rate), 16000); #else - denoiser->neg.sampling_rate = TSK_MIN(TSK_MAX(record_sampling_rate, playback_sampling_rate), 16000); // FIXME: 32000 accepted by echo_process fails + denoiser->neg.sampling_rate = TSK_MIN(TSK_MAX(record_sampling_rate, playback_sampling_rate), 16000); // FIXME: 32000 accepted by echo_process fails #endif - denoiser->neg.nb_samples_per_process = /*TSK_CLAMP(80,*/ ((denoiser->neg.sampling_rate * 10) / 1000)/*, 160)*/; // Supported by the module: "80"(10ms) and "160"(20ms) - denoiser->neg.channels = 1; - - // - // RECORD - // - TSK_OBJECT_SAFE_FREE(denoiser->record.p_rpl_den2in); - TSK_OBJECT_SAFE_FREE(denoiser->record.p_rpl_in2den); - pin_record_in.n_sample_size = sizeof(int16_t); - pin_record_in.n_rate = record_sampling_rate; - pin_record_in.n_channels = record_channels; - pin_record_in.n_duration = (((record_frame_size_samples * 1000) / record_sampling_rate)) / record_channels; - pin_record_den.n_sample_size = sizeof(sample_t); - pin_record_den.n_rate = denoiser->neg.sampling_rate; - - pin_record_den.n_channels = 1; - pin_record_den.n_duration = pin_record_in.n_duration; - if (pin_record_in.n_sample_size != pin_record_den.n_sample_size || pin_record_in.n_rate != pin_record_den.n_rate || pin_record_in.n_channels != pin_record_den.n_channels) { - if ((ret = _tdav_webrtc_resampler_create(&pin_record_in, &pin_record_den, &denoiser->record.p_rpl_in2den))) { - return ret; - } - if ((ret = _tdav_webrtc_resampler_create(&pin_record_den, &pin_record_in, &denoiser->record.p_rpl_den2in))) { - return ret; - } - } - // - // PLAYBACK - // - TSK_OBJECT_SAFE_FREE(denoiser->playback.p_rpl_den2in); - TSK_OBJECT_SAFE_FREE(denoiser->playback.p_rpl_in2den); - pin_playback_in.n_sample_size = sizeof(int16_t); - pin_playback_in.n_rate = playback_sampling_rate; - pin_playback_in.n_channels = playback_channels; - pin_playback_in.n_duration = (((playback_frame_size_samples * 1000) / playback_sampling_rate)) / playback_channels; - pin_playback_den.n_sample_size = sizeof(sample_t); - pin_playback_den.n_rate = denoiser->neg.sampling_rate; - pin_playback_den.n_channels = 1; - pin_playback_den.n_duration = pin_playback_in.n_duration; - if (pin_playback_in.n_sample_size != pin_playback_den.n_sample_size || pin_playback_in.n_rate != pin_playback_den.n_rate || pin_playback_in.n_channels != pin_playback_den.n_channels) { - if ((ret = _tdav_webrtc_resampler_create(&pin_playback_in, &pin_playback_den, &denoiser->playback.p_rpl_in2den))) { - return ret; - } - if ((ret = _tdav_webrtc_resampler_create(&pin_playback_den, &pin_playback_in, &denoiser->playback.p_rpl_den2in))) { - return ret; - } - } - - // - // AEC instance - // - if ((ret = TDAV_WebRtcAec_Create(&denoiser->AEC_inst))) { - TSK_DEBUG_ERROR("WebRtcAec_Create failed with error code = %d", ret); - return ret; - } - if ((ret = TDAV_WebRtcAec_Init(denoiser->AEC_inst, denoiser->neg.sampling_rate, denoiser->neg.sampling_rate))) { - TSK_DEBUG_ERROR("WebRtcAec_Init failed with error code = %d", ret); - return ret; - } + denoiser->neg.nb_samples_per_process = /*TSK_CLAMP(80,*/ ((denoiser->neg.sampling_rate * 10) / 1000)/*, 160)*/; // Supported by the module: "80"(10ms) and "160"(20ms) + denoiser->neg.channels = 1; + + // + // RECORD + // + TSK_OBJECT_SAFE_FREE(denoiser->record.p_rpl_den2in); + TSK_OBJECT_SAFE_FREE(denoiser->record.p_rpl_in2den); + pin_record_in.n_sample_size = sizeof(int16_t); + pin_record_in.n_rate = record_sampling_rate; + pin_record_in.n_channels = record_channels; + pin_record_in.n_duration = (((record_frame_size_samples * 1000) / record_sampling_rate)) / record_channels; + pin_record_den.n_sample_size = sizeof(sample_t); + pin_record_den.n_rate = denoiser->neg.sampling_rate; + + pin_record_den.n_channels = 1; + pin_record_den.n_duration = pin_record_in.n_duration; + if (pin_record_in.n_sample_size != pin_record_den.n_sample_size || pin_record_in.n_rate != pin_record_den.n_rate || pin_record_in.n_channels != pin_record_den.n_channels) { + if ((ret = _tdav_webrtc_resampler_create(&pin_record_in, &pin_record_den, &denoiser->record.p_rpl_in2den))) { + return ret; + } + if ((ret = _tdav_webrtc_resampler_create(&pin_record_den, &pin_record_in, &denoiser->record.p_rpl_den2in))) { + return ret; + } + } + // + // PLAYBACK + // + TSK_OBJECT_SAFE_FREE(denoiser->playback.p_rpl_den2in); + TSK_OBJECT_SAFE_FREE(denoiser->playback.p_rpl_in2den); + pin_playback_in.n_sample_size = sizeof(int16_t); + pin_playback_in.n_rate = playback_sampling_rate; + pin_playback_in.n_channels = playback_channels; + pin_playback_in.n_duration = (((playback_frame_size_samples * 1000) / playback_sampling_rate)) / playback_channels; + pin_playback_den.n_sample_size = sizeof(sample_t); + pin_playback_den.n_rate = denoiser->neg.sampling_rate; + pin_playback_den.n_channels = 1; + pin_playback_den.n_duration = pin_playback_in.n_duration; + if (pin_playback_in.n_sample_size != pin_playback_den.n_sample_size || pin_playback_in.n_rate != pin_playback_den.n_rate || pin_playback_in.n_channels != pin_playback_den.n_channels) { + if ((ret = _tdav_webrtc_resampler_create(&pin_playback_in, &pin_playback_den, &denoiser->playback.p_rpl_in2den))) { + return ret; + } + if ((ret = _tdav_webrtc_resampler_create(&pin_playback_den, &pin_playback_in, &denoiser->playback.p_rpl_den2in))) { + return ret; + } + } + + // + // AEC instance + // + if ((ret = TDAV_WebRtcAec_Create(&denoiser->AEC_inst))) { + TSK_DEBUG_ERROR("WebRtcAec_Create failed with error code = %d", ret); + return ret; + } + if ((ret = TDAV_WebRtcAec_Init(denoiser->AEC_inst, denoiser->neg.sampling_rate, denoiser->neg.sampling_rate))) { + TSK_DEBUG_ERROR("WebRtcAec_Init failed with error code = %d", ret); + return ret; + } #if TDAV_UNDER_MOBILE #else - { - AecConfig aecConfig; + { + AecConfig aecConfig; #if WEBRTC_AEC_AGGRESSIVE - aecConfig.nlpMode = kAecNlpAggressive; + aecConfig.nlpMode = kAecNlpAggressive; #else - aecConfig.nlpMode = kAecNlpModerate; + aecConfig.nlpMode = kAecNlpModerate; #endif - aecConfig.skewMode = kAecFalse; - aecConfig.metricsMode = kAecTrue; - aecConfig.delay_logging = kAecFalse; - if ((ret = WebRtcAec_set_config(denoiser->AEC_inst, aecConfig))) { - TSK_DEBUG_ERROR("WebRtcAec_set_config failed with error code = %d", ret); - } - } + aecConfig.skewMode = kAecFalse; + aecConfig.metricsMode = kAecTrue; + aecConfig.delay_logging = kAecFalse; + if ((ret = WebRtcAec_set_config(denoiser->AEC_inst, aecConfig))) { + TSK_DEBUG_ERROR("WebRtcAec_set_config failed with error code = %d", ret); + } + } #endif - // - // Noise Suppression instance - // - if (TMEDIA_DENOISE(denoiser)->noise_supp_enabled) { + // + // Noise Suppression instance + // + if (TMEDIA_DENOISE(denoiser)->noise_supp_enabled) { #if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER - if ((denoiser->SpeexDenoiser_proc = speex_preprocess_state_init((pin_record_den.n_rate / 1000) * pin_record_den.n_duration, pin_record_den.n_rate))) { - int i = 1; - speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_DENOISE, &i); - i = TMEDIA_DENOISE(denoiser)->noise_supp_level; - speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); - } + if ((denoiser->SpeexDenoiser_proc = speex_preprocess_state_init((pin_record_den.n_rate / 1000) * pin_record_den.n_duration, pin_record_den.n_rate))) { + int i = 1; + speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_DENOISE, &i); + i = TMEDIA_DENOISE(denoiser)->noise_supp_level; + speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i); + } #else - if ((ret = TDAV_WebRtcNs_Create(&denoiser->NS_inst))) { - TSK_DEBUG_ERROR("WebRtcNs_Create failed with error code = %d", ret); - return ret; - } - if ((ret = TDAV_WebRtcNs_Init(denoiser->NS_inst, 80))) { - TSK_DEBUG_ERROR("WebRtcNs_Init failed with error code = %d", ret); - return ret; - } + if ((ret = TDAV_WebRtcNs_Create(&denoiser->NS_inst))) { + TSK_DEBUG_ERROR("WebRtcNs_Create failed with error code = %d", ret); + return ret; + } + if ((ret = TDAV_WebRtcNs_Init(denoiser->NS_inst, 80))) { + TSK_DEBUG_ERROR("WebRtcNs_Init failed with error code = %d", ret); + return ret; + } #endif - } + } - TSK_DEBUG_INFO("WebRTC denoiser opened: record:%uHz,%uchannels // playback:%uHz,%uchannels // neg:%uHz,%uchannels", - record_sampling_rate, record_channels, - playback_sampling_rate, playback_channels, - denoiser->neg.sampling_rate, denoiser->neg.channels); + TSK_DEBUG_INFO("WebRTC denoiser opened: record:%uHz,%uchannels // playback:%uHz,%uchannels // neg:%uHz,%uchannels", + record_sampling_rate, record_channels, + playback_sampling_rate, playback_channels, + denoiser->neg.sampling_rate, denoiser->neg.channels); - return ret; + return ret; } static int tdav_webrtc_denoise_echo_playback(tmedia_denoise_t* self, const void* echo_frame, uint32_t echo_frame_size_bytes) { - tdav_webrtc_denoise_t *p_self = (tdav_webrtc_denoise_t *)self; - int ret = 0; - - tsk_safeobj_lock(p_self); - if (p_self->AEC_inst && echo_frame && echo_frame_size_bytes) { - const sample_t* _echo_frame = (const sample_t*)echo_frame; - tsk_size_t _echo_frame_size_bytes = echo_frame_size_bytes; - tsk_size_t _echo_frame_size_samples = (_echo_frame_size_bytes / sizeof(int16_t)); - // IN -> DEN - if (p_self->playback.p_rpl_in2den) { - if ((ret = _tdav_webrtc_resampler_process(p_self->playback.p_rpl_in2den, _echo_frame, _echo_frame_size_bytes))) { - goto bail; - } - _echo_frame = p_self->playback.p_rpl_in2den->out.p_buff_ptr; - _echo_frame_size_bytes = p_self->playback.p_rpl_in2den->out.n_buff_size_in_bytes; - _echo_frame_size_samples = p_self->playback.p_rpl_in2den->out.n_buff_size_in_samples; - } - // PROCESS - if (_echo_frame_size_samples && _echo_frame) { - uint32_t _samples; - for (_samples = 0; _samples < _echo_frame_size_samples; _samples += p_self->neg.nb_samples_per_process) { - if ((ret = TDAV_WebRtcAec_BufferFarend(p_self->AEC_inst, &_echo_frame[_samples], p_self->neg.nb_samples_per_process))){ - TSK_DEBUG_ERROR("WebRtcAec_BufferFarend failed with error code = %d, nb_samples_per_process=%u", ret, p_self->neg.nb_samples_per_process); - goto bail; - } - } - } - } + tdav_webrtc_denoise_t *p_self = (tdav_webrtc_denoise_t *)self; + int ret = 0; + + tsk_safeobj_lock(p_self); + if (p_self->AEC_inst && echo_frame && echo_frame_size_bytes) { + const sample_t* _echo_frame = (const sample_t*)echo_frame; + tsk_size_t _echo_frame_size_bytes = echo_frame_size_bytes; + tsk_size_t _echo_frame_size_samples = (_echo_frame_size_bytes / sizeof(int16_t)); + // IN -> DEN + if (p_self->playback.p_rpl_in2den) { + if ((ret = _tdav_webrtc_resampler_process(p_self->playback.p_rpl_in2den, _echo_frame, _echo_frame_size_bytes))) { + goto bail; + } + _echo_frame = p_self->playback.p_rpl_in2den->out.p_buff_ptr; + _echo_frame_size_bytes = p_self->playback.p_rpl_in2den->out.n_buff_size_in_bytes; + _echo_frame_size_samples = p_self->playback.p_rpl_in2den->out.n_buff_size_in_samples; + } + // PROCESS + if (_echo_frame_size_samples && _echo_frame) { + uint32_t _samples; + for (_samples = 0; _samples < _echo_frame_size_samples; _samples += p_self->neg.nb_samples_per_process) { + if ((ret = TDAV_WebRtcAec_BufferFarend(p_self->AEC_inst, &_echo_frame[_samples], p_self->neg.nb_samples_per_process))) { + TSK_DEBUG_ERROR("WebRtcAec_BufferFarend failed with error code = %d, nb_samples_per_process=%u", ret, p_self->neg.nb_samples_per_process); + goto bail; + } + } + } + } bail: - tsk_safeobj_unlock(p_self); - return ret; + tsk_safeobj_unlock(p_self); + return ret; } static int tdav_webrtc_denoise_process_record(tmedia_denoise_t* self, void* audio_frame, uint32_t audio_frame_size_bytes, tsk_bool_t* silence_or_noise) { - tdav_webrtc_denoise_t *p_self = (tdav_webrtc_denoise_t *)self; - int ret = 0; - - *silence_or_noise = tsk_false; - - tsk_safeobj_lock(p_self); - - if (p_self->AEC_inst && audio_frame && audio_frame_size_bytes) { - tsk_size_t _samples; - const sample_t* _audio_frame = (const sample_t*)audio_frame; - tsk_size_t _audio_frame_size_bytes = audio_frame_size_bytes; - tsk_size_t _audio_frame_size_samples = (_audio_frame_size_bytes / sizeof(int16_t)); - // IN -> DEN - if (p_self->record.p_rpl_in2den) { - if ((ret = _tdav_webrtc_resampler_process(p_self->record.p_rpl_in2den, _audio_frame, _audio_frame_size_bytes))) { - goto bail; - } - _audio_frame = p_self->record.p_rpl_in2den->out.p_buff_ptr; - _audio_frame_size_bytes = p_self->record.p_rpl_in2den->out.n_buff_size_in_bytes; - _audio_frame_size_samples = p_self->record.p_rpl_in2den->out.n_buff_size_in_samples; - } - // NOISE SUPPRESSION + tdav_webrtc_denoise_t *p_self = (tdav_webrtc_denoise_t *)self; + int ret = 0; + + *silence_or_noise = tsk_false; + + tsk_safeobj_lock(p_self); + + if (p_self->AEC_inst && audio_frame && audio_frame_size_bytes) { + tsk_size_t _samples; + const sample_t* _audio_frame = (const sample_t*)audio_frame; + tsk_size_t _audio_frame_size_bytes = audio_frame_size_bytes; + tsk_size_t _audio_frame_size_samples = (_audio_frame_size_bytes / sizeof(int16_t)); + // IN -> DEN + if (p_self->record.p_rpl_in2den) { + if ((ret = _tdav_webrtc_resampler_process(p_self->record.p_rpl_in2den, _audio_frame, _audio_frame_size_bytes))) { + goto bail; + } + _audio_frame = p_self->record.p_rpl_in2den->out.p_buff_ptr; + _audio_frame_size_bytes = p_self->record.p_rpl_in2den->out.n_buff_size_in_bytes; + _audio_frame_size_samples = p_self->record.p_rpl_in2den->out.n_buff_size_in_samples; + } + // NOISE SUPPRESSION #if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER - if (p_self->SpeexDenoiser_proc) { - speex_preprocess_run(p_self->SpeexDenoiser_proc, (spx_int16_t*)_audio_frame); - } + if (p_self->SpeexDenoiser_proc) { + speex_preprocess_run(p_self->SpeexDenoiser_proc, (spx_int16_t*)_audio_frame); + } #else - // WebRTC NoiseSupp only accept 10ms frames - // Our encoder will always output 20ms frames ==> execute 2x noise_supp - if (p_self->NS_inst) { - for (_samples = 0; _samples < _audio_frame_size_samples; _samples+= p_self->neg.nb_samples_per_process) { - if ((ret = TDAV_WebRtcNs_Process(p_self->NS_inst, &_audio_frame[_samples], tsk_null, _audio_frame, tsk_null))) { - TSK_DEBUG_ERROR("WebRtcNs_Process with error code = %d", ret); - goto bail; - } - } - } + // WebRTC NoiseSupp only accept 10ms frames + // Our encoder will always output 20ms frames ==> execute 2x noise_supp + if (p_self->NS_inst) { + for (_samples = 0; _samples < _audio_frame_size_samples; _samples+= p_self->neg.nb_samples_per_process) { + if ((ret = TDAV_WebRtcNs_Process(p_self->NS_inst, &_audio_frame[_samples], tsk_null, _audio_frame, tsk_null))) { + TSK_DEBUG_ERROR("WebRtcNs_Process with error code = %d", ret); + goto bail; + } + } + } #endif - // PROCESS - if (_audio_frame_size_samples && _audio_frame) { - for (_samples = 0; _samples < _audio_frame_size_samples; _samples += p_self->neg.nb_samples_per_process) { - if ((ret = TDAV_WebRtcAec_Process(p_self->AEC_inst, &_audio_frame[_samples], tsk_null, (sample_t*)&_audio_frame[_samples], tsk_null, p_self->neg.nb_samples_per_process, p_self->echo_tail, p_self->echo_skew))){ - TSK_DEBUG_ERROR("WebRtcAec_Process with error code = %d, nb_samples_per_process=%u", ret, p_self->neg.nb_samples_per_process); - goto bail; - } - } - } - // DEN -> IN - if (p_self->record.p_rpl_den2in) { - if ((ret = _tdav_webrtc_resampler_process(p_self->record.p_rpl_den2in, _audio_frame, _audio_frame_size_bytes))) { - goto bail; - } - _audio_frame = p_self->record.p_rpl_den2in->out.p_buff_ptr; - _audio_frame_size_bytes = p_self->record.p_rpl_den2in->out.n_buff_size_in_bytes; - _audio_frame_size_samples = p_self->record.p_rpl_den2in->out.n_buff_size_in_samples; - } - // Sanity check - if (_audio_frame_size_bytes != audio_frame_size_bytes) { - TSK_DEBUG_ERROR("Size mismatch: %u <> %u", _audio_frame_size_bytes, audio_frame_size_bytes); - ret = -3; - goto bail; - } - if (audio_frame != (const void*)_audio_frame) { - memcpy(audio_frame, _audio_frame, _audio_frame_size_bytes); - } - } + // PROCESS + if (_audio_frame_size_samples && _audio_frame) { + for (_samples = 0; _samples < _audio_frame_size_samples; _samples += p_self->neg.nb_samples_per_process) { + if ((ret = TDAV_WebRtcAec_Process(p_self->AEC_inst, &_audio_frame[_samples], tsk_null, (sample_t*)&_audio_frame[_samples], tsk_null, p_self->neg.nb_samples_per_process, p_self->echo_tail, p_self->echo_skew))) { + TSK_DEBUG_ERROR("WebRtcAec_Process with error code = %d, nb_samples_per_process=%u", ret, p_self->neg.nb_samples_per_process); + goto bail; + } + } + } + // DEN -> IN + if (p_self->record.p_rpl_den2in) { + if ((ret = _tdav_webrtc_resampler_process(p_self->record.p_rpl_den2in, _audio_frame, _audio_frame_size_bytes))) { + goto bail; + } + _audio_frame = p_self->record.p_rpl_den2in->out.p_buff_ptr; + _audio_frame_size_bytes = p_self->record.p_rpl_den2in->out.n_buff_size_in_bytes; + _audio_frame_size_samples = p_self->record.p_rpl_den2in->out.n_buff_size_in_samples; + } + // Sanity check + if (_audio_frame_size_bytes != audio_frame_size_bytes) { + TSK_DEBUG_ERROR("Size mismatch: %u <> %u", _audio_frame_size_bytes, audio_frame_size_bytes); + ret = -3; + goto bail; + } + if (audio_frame != (const void*)_audio_frame) { + memcpy(audio_frame, _audio_frame, _audio_frame_size_bytes); + } + } bail: - tsk_safeobj_unlock(p_self); - return ret; + tsk_safeobj_unlock(p_self); + return ret; } static int tdav_webrtc_denoise_process_playback(tmedia_denoise_t* self, void* audio_frame, uint32_t audio_frame_size_bytes) { - tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; + tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; - (void)(denoiser); + (void)(denoiser); - // Not mandatory to denoise audio before playback. - // All Doubango clients support noise suppression. - return 0; + // Not mandatory to denoise audio before playback. + // All Doubango clients support noise suppression. + return 0; } static int tdav_webrtc_denoise_close(tmedia_denoise_t* self) { - tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; + tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; - tsk_safeobj_lock(denoiser); - if (denoiser->AEC_inst) { - TDAV_WebRtcAec_Free(denoiser->AEC_inst); - denoiser->AEC_inst = tsk_null; - } + tsk_safeobj_lock(denoiser); + if (denoiser->AEC_inst) { + TDAV_WebRtcAec_Free(denoiser->AEC_inst); + denoiser->AEC_inst = tsk_null; + } #if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER - if (denoiser->SpeexDenoiser_proc) { - speex_preprocess_state_destroy(denoiser->SpeexDenoiser_proc); - denoiser->SpeexDenoiser_proc = tsk_null; - } + if (denoiser->SpeexDenoiser_proc) { + speex_preprocess_state_destroy(denoiser->SpeexDenoiser_proc); + denoiser->SpeexDenoiser_proc = tsk_null; + } #else - if (denoiser->NS_inst) { - TDAV_WebRtcNs_Free(denoiser->NS_inst); - denoiser->NS_inst = tsk_null; - } + if (denoiser->NS_inst) { + TDAV_WebRtcNs_Free(denoiser->NS_inst); + denoiser->NS_inst = tsk_null; + } #endif - tsk_safeobj_unlock(denoiser); + tsk_safeobj_unlock(denoiser); - return 0; + return 0; } static int _tdav_webrtc_resampler_create(const tdav_webrtc_pin_xt* p_pin_in, const tdav_webrtc_pin_xt* p_pin_out, tdav_webrtc_resampler_t **pp_resampler) { - extern const tsk_object_def_t *tdav_webrtc_resampler_def_t; - int ret = 0; - if (!p_pin_in || !p_pin_out || !pp_resampler || *pp_resampler) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if (!(*pp_resampler = tsk_object_new(tdav_webrtc_resampler_def_t))) { - TSK_DEBUG_ERROR("Failed to create resampler object"); - ret = -3; - goto bail; - } - if (!((*pp_resampler)->p_resampler = tmedia_resampler_create())) { - ret = -3; - goto bail; - } - ret = tmedia_resampler_open((*pp_resampler)->p_resampler, - p_pin_in->n_rate, p_pin_out->n_rate, - p_pin_in->n_duration, - p_pin_in->n_channels, p_pin_out->n_channels, - TMEDIA_RESAMPLER_QUALITY, - (p_pin_out->n_sample_size << 3)); - if (ret) { - TSK_DEBUG_ERROR("Failed to open resampler: in_rate=%u,in_duration=%u,in_channels=%u /// out_rate=%u,out_duration=%u,out_channels=%u", - p_pin_in->n_rate, p_pin_in->n_duration, p_pin_in->n_channels, - p_pin_out->n_rate, p_pin_out->n_duration, p_pin_out->n_channels); - goto bail; - } - - (*pp_resampler)->out.n_buff_size_in_bytes = ((((p_pin_out->n_rate * p_pin_out->n_duration) / 1000)) * p_pin_out->n_channels) * p_pin_out->n_sample_size; - (*pp_resampler)->out.p_buff_ptr = tsk_malloc((*pp_resampler)->out.n_buff_size_in_bytes); - if (!(*pp_resampler)->out.p_buff_ptr) { - TSK_DEBUG_ERROR("Failed to allocate buffer with size=%u", (*pp_resampler)->out.n_buff_size_in_bytes); - ret = -3; - goto bail; - } - (*pp_resampler)->out.n_buff_size_in_samples = (*pp_resampler)->out.n_buff_size_in_bytes / p_pin_out->n_sample_size; - (*pp_resampler)->in.n_buff_size_in_bytes = ((((p_pin_in->n_rate * p_pin_in->n_duration) / 1000)) * p_pin_in->n_channels) * p_pin_in->n_sample_size; - (*pp_resampler)->in.n_buff_size_in_samples = (*pp_resampler)->in.n_buff_size_in_bytes / p_pin_in->n_sample_size; - - (*pp_resampler)->n_bufftmp_size_in_bytes = (((48000 * TSK_MAX(p_pin_in->n_duration, p_pin_out->n_duration)) / 1000) * 2/*channels*/) * sizeof(float); // Max - (*pp_resampler)->p_bufftmp_ptr = tsk_malloc((*pp_resampler)->n_bufftmp_size_in_bytes); - if (!(*pp_resampler)->p_bufftmp_ptr) { - TSK_DEBUG_ERROR("Failed to allocate buffer with size:%u", (*pp_resampler)->n_bufftmp_size_in_bytes); - ret = -3; - goto bail; - } - - memcpy(&(*pp_resampler)->in.x_pin, p_pin_in, sizeof(tdav_webrtc_pin_xt)); - memcpy(&(*pp_resampler)->out.x_pin, p_pin_out, sizeof(tdav_webrtc_pin_xt)); + extern const tsk_object_def_t *tdav_webrtc_resampler_def_t; + int ret = 0; + if (!p_pin_in || !p_pin_out || !pp_resampler || *pp_resampler) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if (!(*pp_resampler = tsk_object_new(tdav_webrtc_resampler_def_t))) { + TSK_DEBUG_ERROR("Failed to create resampler object"); + ret = -3; + goto bail; + } + if (!((*pp_resampler)->p_resampler = tmedia_resampler_create())) { + ret = -3; + goto bail; + } + ret = tmedia_resampler_open((*pp_resampler)->p_resampler, + p_pin_in->n_rate, p_pin_out->n_rate, + p_pin_in->n_duration, + p_pin_in->n_channels, p_pin_out->n_channels, + TMEDIA_RESAMPLER_QUALITY, + (p_pin_out->n_sample_size << 3)); + if (ret) { + TSK_DEBUG_ERROR("Failed to open resampler: in_rate=%u,in_duration=%u,in_channels=%u /// out_rate=%u,out_duration=%u,out_channels=%u", + p_pin_in->n_rate, p_pin_in->n_duration, p_pin_in->n_channels, + p_pin_out->n_rate, p_pin_out->n_duration, p_pin_out->n_channels); + goto bail; + } + + (*pp_resampler)->out.n_buff_size_in_bytes = ((((p_pin_out->n_rate * p_pin_out->n_duration) / 1000)) * p_pin_out->n_channels) * p_pin_out->n_sample_size; + (*pp_resampler)->out.p_buff_ptr = tsk_malloc((*pp_resampler)->out.n_buff_size_in_bytes); + if (!(*pp_resampler)->out.p_buff_ptr) { + TSK_DEBUG_ERROR("Failed to allocate buffer with size=%u", (*pp_resampler)->out.n_buff_size_in_bytes); + ret = -3; + goto bail; + } + (*pp_resampler)->out.n_buff_size_in_samples = (*pp_resampler)->out.n_buff_size_in_bytes / p_pin_out->n_sample_size; + (*pp_resampler)->in.n_buff_size_in_bytes = ((((p_pin_in->n_rate * p_pin_in->n_duration) / 1000)) * p_pin_in->n_channels) * p_pin_in->n_sample_size; + (*pp_resampler)->in.n_buff_size_in_samples = (*pp_resampler)->in.n_buff_size_in_bytes / p_pin_in->n_sample_size; + + (*pp_resampler)->n_bufftmp_size_in_bytes = (((48000 * TSK_MAX(p_pin_in->n_duration, p_pin_out->n_duration)) / 1000) * 2/*channels*/) * sizeof(float); // Max + (*pp_resampler)->p_bufftmp_ptr = tsk_malloc((*pp_resampler)->n_bufftmp_size_in_bytes); + if (!(*pp_resampler)->p_bufftmp_ptr) { + TSK_DEBUG_ERROR("Failed to allocate buffer with size:%u", (*pp_resampler)->n_bufftmp_size_in_bytes); + ret = -3; + goto bail; + } + + memcpy(&(*pp_resampler)->in.x_pin, p_pin_in, sizeof(tdav_webrtc_pin_xt)); + memcpy(&(*pp_resampler)->out.x_pin, p_pin_out, sizeof(tdav_webrtc_pin_xt)); bail: - if (ret) { - TSK_OBJECT_SAFE_FREE((*pp_resampler)); - } - return ret; + if (ret) { + TSK_OBJECT_SAFE_FREE((*pp_resampler)); + } + return ret; } static int _tdav_webrtc_resampler_process(tdav_webrtc_resampler_t *p_self, const void* p_buff_ptr, tsk_size_t n_buff_size_in_bytes) { - tsk_size_t n_out_size; - const void* _p_buff_ptr = p_buff_ptr; - tsk_size_t _n_buff_size_in_bytes = n_buff_size_in_bytes; - tsk_size_t _n_buff_size_in_samples; - - if (!p_self || !p_buff_ptr || !n_buff_size_in_bytes) { - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - if (p_self->in.n_buff_size_in_bytes != n_buff_size_in_bytes) { - TSK_DEBUG_ERROR("Invalid input size: %u <> %u", p_self->in.n_buff_size_in_bytes, n_buff_size_in_bytes); - return -2; - } - _n_buff_size_in_samples = p_self->in.n_buff_size_in_samples; - if (p_self->in.x_pin.n_sample_size != p_self->out.x_pin.n_sample_size) { - tsk_size_t index; - if (p_self->in.x_pin.n_sample_size == sizeof(int16_t)) { - // int16_t -> float - const int16_t* p_src = (const int16_t*)p_buff_ptr; - float* p_dst = (float*)p_self->p_bufftmp_ptr; - for (index = 0; index < _n_buff_size_in_samples; ++index) { - p_dst[index] = (float)p_src[index]; - } - } - else { - // float -> int16_t - const float* p_src = (const float*)p_buff_ptr; - int16_t* p_dst = (int16_t*)p_self->p_bufftmp_ptr; - for (index = 0; index < _n_buff_size_in_samples; ++index) { - p_dst[index] = (int16_t)p_src[index]; - } - } - _p_buff_ptr = p_self->p_bufftmp_ptr; - _n_buff_size_in_bytes = p_self->in.n_buff_size_in_bytes; - } - n_out_size = tmedia_resampler_process(p_self->p_resampler, _p_buff_ptr, _n_buff_size_in_samples, (int16_t*)p_self->out.p_buff_ptr, p_self->out.n_buff_size_in_samples); - if (n_out_size != p_self->out.n_buff_size_in_samples) { - TSK_DEBUG_ERROR("Invalid output size: %u <> %u", n_out_size, p_self->out.n_buff_size_in_bytes); - return -4; - } - return 0; + tsk_size_t n_out_size; + const void* _p_buff_ptr = p_buff_ptr; + tsk_size_t _n_buff_size_in_bytes = n_buff_size_in_bytes; + tsk_size_t _n_buff_size_in_samples; + + if (!p_self || !p_buff_ptr || !n_buff_size_in_bytes) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + if (p_self->in.n_buff_size_in_bytes != n_buff_size_in_bytes) { + TSK_DEBUG_ERROR("Invalid input size: %u <> %u", p_self->in.n_buff_size_in_bytes, n_buff_size_in_bytes); + return -2; + } + _n_buff_size_in_samples = p_self->in.n_buff_size_in_samples; + if (p_self->in.x_pin.n_sample_size != p_self->out.x_pin.n_sample_size) { + tsk_size_t index; + if (p_self->in.x_pin.n_sample_size == sizeof(int16_t)) { + // int16_t -> float + const int16_t* p_src = (const int16_t*)p_buff_ptr; + float* p_dst = (float*)p_self->p_bufftmp_ptr; + for (index = 0; index < _n_buff_size_in_samples; ++index) { + p_dst[index] = (float)p_src[index]; + } + } + else { + // float -> int16_t + const float* p_src = (const float*)p_buff_ptr; + int16_t* p_dst = (int16_t*)p_self->p_bufftmp_ptr; + for (index = 0; index < _n_buff_size_in_samples; ++index) { + p_dst[index] = (int16_t)p_src[index]; + } + } + _p_buff_ptr = p_self->p_bufftmp_ptr; + _n_buff_size_in_bytes = p_self->in.n_buff_size_in_bytes; + } + n_out_size = tmedia_resampler_process(p_self->p_resampler, _p_buff_ptr, _n_buff_size_in_samples, (int16_t*)p_self->out.p_buff_ptr, p_self->out.n_buff_size_in_samples); + if (n_out_size != p_self->out.n_buff_size_in_samples) { + TSK_DEBUG_ERROR("Invalid output size: %u <> %u", n_out_size, p_self->out.n_buff_size_in_bytes); + return -4; + } + return 0; } // @@ -534,28 +531,27 @@ static int _tdav_webrtc_resampler_process(tdav_webrtc_resampler_t *p_self, const // static tsk_object_t* tdav_webrtc_resampler_ctor(tsk_object_t * self, va_list * app) { - tdav_webrtc_resampler_t *p_resampler = (tdav_webrtc_resampler_t*)self; - if (p_resampler) { + tdav_webrtc_resampler_t *p_resampler = (tdav_webrtc_resampler_t*)self; + if (p_resampler) { - } - return self; + } + return self; } static tsk_object_t* tdav_webrtc_resampler_dtor(tsk_object_t * self) { - tdav_webrtc_resampler_t *p_resampler = (tdav_webrtc_resampler_t*)self; - if (p_resampler) { - TSK_OBJECT_SAFE_FREE(p_resampler->p_resampler); - TSK_FREE(p_resampler->out.p_buff_ptr); - TSK_FREE(p_resampler->p_bufftmp_ptr); - } - return self; + tdav_webrtc_resampler_t *p_resampler = (tdav_webrtc_resampler_t*)self; + if (p_resampler) { + TSK_OBJECT_SAFE_FREE(p_resampler->p_resampler); + TSK_FREE(p_resampler->out.p_buff_ptr); + TSK_FREE(p_resampler->p_bufftmp_ptr); + } + return self; } -static const tsk_object_def_t tdav_webrtc_resampler_def_s = -{ - sizeof(tdav_webrtc_resampler_t), - tdav_webrtc_resampler_ctor, - tdav_webrtc_resampler_dtor, - tsk_object_cmp, +static const tsk_object_def_t tdav_webrtc_resampler_def_s = { + sizeof(tdav_webrtc_resampler_t), + tdav_webrtc_resampler_ctor, + tdav_webrtc_resampler_dtor, + tsk_object_cmp, }; const tsk_object_def_t *tdav_webrtc_resampler_def_t = &tdav_webrtc_resampler_def_s; @@ -567,59 +563,57 @@ const tsk_object_def_t *tdav_webrtc_resampler_def_t = &tdav_webrtc_resampler_def /* constructor */ static tsk_object_t* tdav_webrtc_denoise_ctor(tsk_object_t * _self, va_list * app) { - tdav_webrtc_denoise_t *self = _self; - if (self){ - /* init base */ - tmedia_denoise_init(TMEDIA_DENOISE(self)); - /* init self */ - tsk_safeobj_init(self); - self->neg.channels = 1; - - TSK_DEBUG_INFO("Create WebRTC denoiser"); - } - return self; + tdav_webrtc_denoise_t *self = _self; + if (self) { + /* init base */ + tmedia_denoise_init(TMEDIA_DENOISE(self)); + /* init self */ + tsk_safeobj_init(self); + self->neg.channels = 1; + + TSK_DEBUG_INFO("Create WebRTC denoiser"); + } + return self; } /* destructor */ static tsk_object_t* tdav_webrtc_denoise_dtor(tsk_object_t * _self) { - tdav_webrtc_denoise_t *self = _self; - if (self){ - /* deinit base (will close the denoise if not done yet) */ - tmedia_denoise_deinit(TMEDIA_DENOISE(self)); - /* deinit self */ - tdav_webrtc_denoise_close(TMEDIA_DENOISE(self)); - TSK_OBJECT_SAFE_FREE(self->record.p_rpl_in2den); - TSK_OBJECT_SAFE_FREE(self->record.p_rpl_den2in); - TSK_OBJECT_SAFE_FREE(self->playback.p_rpl_in2den); - TSK_OBJECT_SAFE_FREE(self->playback.p_rpl_den2in); - tsk_safeobj_deinit(self); - - TSK_DEBUG_INFO("*** Destroy WebRTC denoiser ***"); - } - - return self; + tdav_webrtc_denoise_t *self = _self; + if (self) { + /* deinit base (will close the denoise if not done yet) */ + tmedia_denoise_deinit(TMEDIA_DENOISE(self)); + /* deinit self */ + tdav_webrtc_denoise_close(TMEDIA_DENOISE(self)); + TSK_OBJECT_SAFE_FREE(self->record.p_rpl_in2den); + TSK_OBJECT_SAFE_FREE(self->record.p_rpl_den2in); + TSK_OBJECT_SAFE_FREE(self->playback.p_rpl_in2den); + TSK_OBJECT_SAFE_FREE(self->playback.p_rpl_den2in); + tsk_safeobj_deinit(self); + + TSK_DEBUG_INFO("*** Destroy WebRTC denoiser ***"); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_webrtc_denoise_def_s = -{ - sizeof(tdav_webrtc_denoise_t), - tdav_webrtc_denoise_ctor, - tdav_webrtc_denoise_dtor, - tsk_null, +static const tsk_object_def_t tdav_webrtc_denoise_def_s = { + sizeof(tdav_webrtc_denoise_t), + tdav_webrtc_denoise_ctor, + tdav_webrtc_denoise_dtor, + tsk_null, }; /* plugin definition*/ -static const tmedia_denoise_plugin_def_t tdav_webrtc_denoise_plugin_def_s = -{ - &tdav_webrtc_denoise_def_s, +static const tmedia_denoise_plugin_def_t tdav_webrtc_denoise_plugin_def_s = { + &tdav_webrtc_denoise_def_s, - "Audio Denoiser based on Google WebRTC", + "Audio Denoiser based on Google WebRTC", - tdav_webrtc_denoise_set, - tdav_webrtc_denoise_open, - tdav_webrtc_denoise_echo_playback, - tdav_webrtc_denoise_process_record, - tdav_webrtc_denoise_process_playback, - tdav_webrtc_denoise_close, + tdav_webrtc_denoise_set, + tdav_webrtc_denoise_open, + tdav_webrtc_denoise_echo_playback, + tdav_webrtc_denoise_process_record, + tdav_webrtc_denoise_process_playback, + tdav_webrtc_denoise_close, }; const tmedia_denoise_plugin_def_t *tdav_webrtc_denoise_plugin_def_t = &tdav_webrtc_denoise_plugin_def_s; diff --git a/tinyDAV/src/audio/wasapi/tdav_consumer_wasapi.cxx b/tinyDAV/src/audio/wasapi/tdav_consumer_wasapi.cxx index c3a88e3..c4bd37f 100755 --- a/tinyDAV/src/audio/wasapi/tdav_consumer_wasapi.cxx +++ b/tinyDAV/src/audio/wasapi/tdav_consumer_wasapi.cxx @@ -1,18 +1,18 @@ /*Copyright (C) 2013 Mamadou DIOP * Copyright (C) 2013-2014 Doubango Telecom <http://www.doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -24,7 +24,7 @@ #if HAVE_WASAPI -#include "tinydav/audio/tdav_consumer_audio.h" +#include "tinydav/audio/tdav_consumer_audio.h" #include "tsk_thread.h" #include "tsk_memory.h" @@ -53,59 +53,58 @@ struct tdav_consumer_wasapi_s; namespace Doubango { - namespace VoIP - { - ref class AudioRender sealed - { - public: - virtual ~AudioRender(); - internal: - AudioRender(); - - int Prepare(struct tdav_consumer_wasapi_s* wasapi, const tmedia_codec_t* codec); - int UnPrepare(); - int Start(); - int Stop(); - int Pause(); - int Consume(const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr); - private: - tsk_size_t Read(void* data, tsk_size_t size); - void AsyncThread(Windows::Foundation::IAsyncAction^ operation); - - private: - tsk_mutex_handle_t* m_hMutex; - const struct tdav_consumer_wasapi_s* m_pWrappedConsumer; // Must not take ref() otherwise dtor() will be never called (circular reference) - IAudioClient2* m_pDevice; - IAudioRenderClient* m_pClient; - HANDLE m_hEvent; - Windows::Foundation::IAsyncAction^ m_pAsyncThread; - INT32 m_nBytesPerNotif; - INT32 m_nSourceFrameSizeInBytes; - UINT32 m_nMaxFrameCount; - UINT32 m_nPtime; - - struct { - struct { - void* buffer; - tsk_size_t size; - } chunck; - tsk_ssize_t leftBytes; - SpeexBuffer* buffer; - tsk_size_t size; - } m_ring; - - bool m_bStarted; - bool m_bPrepared; - bool m_bPaused; - }; - } +namespace VoIP +{ +ref class AudioRender sealed +{ +public: + virtual ~AudioRender(); +internal: + AudioRender(); + + int Prepare(struct tdav_consumer_wasapi_s* wasapi, const tmedia_codec_t* codec); + int UnPrepare(); + int Start(); + int Stop(); + int Pause(); + int Consume(const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr); +private: + tsk_size_t Read(void* data, tsk_size_t size); + void AsyncThread(Windows::Foundation::IAsyncAction^ operation); + +private: + tsk_mutex_handle_t* m_hMutex; + const struct tdav_consumer_wasapi_s* m_pWrappedConsumer; // Must not take ref() otherwise dtor() will be never called (circular reference) + IAudioClient2* m_pDevice; + IAudioRenderClient* m_pClient; + HANDLE m_hEvent; + Windows::Foundation::IAsyncAction^ m_pAsyncThread; + INT32 m_nBytesPerNotif; + INT32 m_nSourceFrameSizeInBytes; + UINT32 m_nMaxFrameCount; + UINT32 m_nPtime; + + struct { + struct { + void* buffer; + tsk_size_t size; + } chunck; + tsk_ssize_t leftBytes; + SpeexBuffer* buffer; + tsk_size_t size; + } m_ring; + + bool m_bStarted; + bool m_bPrepared; + bool m_bPaused; +}; +} } -typedef struct tdav_consumer_wasapi_s -{ - TDAV_DECLARE_CONSUMER_AUDIO; +typedef struct tdav_consumer_wasapi_s { + TDAV_DECLARE_CONSUMER_AUDIO; - Doubango::VoIP::AudioRender ^AudioRender; + Doubango::VoIP::AudioRender ^AudioRender; } tdav_consumer_wasapi_t; @@ -115,83 +114,83 @@ extern "C" void tdav_win32_print_error(const char* func, HRESULT hr); /* ============ Media consumer Interface ================= */ static int tdav_consumer_wasapi_set(tmedia_consumer_t* self, const tmedia_param_t* param) -{ - return tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); +{ + return tdav_consumer_audio_set(TDAV_CONSUMER_AUDIO(self), param); } static int tdav_consumer_wasapi_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; + tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; - if (!wasapi || !codec || !wasapi->AudioRender) { - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!wasapi || !codec || !wasapi->AudioRender) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - TMEDIA_CONSUMER(wasapi)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(wasapi)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); TMEDIA_CONSUMER(wasapi)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); TMEDIA_CONSUMER(wasapi)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - + WASAPI_DEBUG_INFO("in.channels=%d, out.channles=%d, in.rate=%d, out.rate=%d, ptime=%d", - TMEDIA_CONSUMER(wasapi)->audio.in.channels, - TMEDIA_CONSUMER(wasapi)->audio.out.channels, - TMEDIA_CONSUMER(wasapi)->audio.in.rate, - TMEDIA_CONSUMER(wasapi)->audio.out.rate, - TMEDIA_CONSUMER(wasapi)->audio.ptime); + TMEDIA_CONSUMER(wasapi)->audio.in.channels, + TMEDIA_CONSUMER(wasapi)->audio.out.channels, + TMEDIA_CONSUMER(wasapi)->audio.in.rate, + TMEDIA_CONSUMER(wasapi)->audio.out.rate, + TMEDIA_CONSUMER(wasapi)->audio.ptime); - return wasapi->AudioRender->Prepare(wasapi, codec); + return wasapi->AudioRender->Prepare(wasapi, codec); } static int tdav_consumer_wasapi_start(tmedia_consumer_t* self) { - tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; + tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; - WASAPI_DEBUG_INFO("tdav_consumer_wasapi_start()"); + WASAPI_DEBUG_INFO("tdav_consumer_wasapi_start()"); - if (!wasapi || !wasapi->AudioRender) { - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!wasapi || !wasapi->AudioRender) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->AudioRender->Start(); + return wasapi->AudioRender->Start(); } static int tdav_consumer_wasapi_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) -{ - tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; - if (!wasapi || !wasapi->AudioRender || !buffer || !size) { - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - return wasapi->AudioRender->Consume(buffer, size, proto_hdr); +{ + tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; + if (!wasapi || !wasapi->AudioRender || !buffer || !size) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + return wasapi->AudioRender->Consume(buffer, size, proto_hdr); } static int tdav_consumer_wasapi_pause(tmedia_consumer_t* self) { - tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; + tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; - if (!wasapi || !wasapi->AudioRender){ - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!wasapi || !wasapi->AudioRender) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->AudioRender->Pause(); + return wasapi->AudioRender->Pause(); } static int tdav_consumer_wasapi_stop(tmedia_consumer_t* self) { - tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; + tdav_consumer_wasapi_t* wasapi = (tdav_consumer_wasapi_t*)self; - WASAPI_DEBUG_INFO("tdav_consumer_wasapi_stop()"); + WASAPI_DEBUG_INFO("tdav_consumer_wasapi_stop()"); - if (!wasapi || !wasapi->AudioRender) { - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if (!wasapi || !wasapi->AudioRender) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->AudioRender->Stop(); + return wasapi->AudioRender->Stop(); } @@ -201,288 +200,288 @@ static int tdav_consumer_wasapi_stop(tmedia_consumer_t* self) Doubango::VoIP::AudioRender::AudioRender() - : m_pDevice(nullptr) - , m_hMutex(nullptr) - , m_pClient(nullptr) - , m_hEvent(nullptr) - , m_pAsyncThread(nullptr) - , m_pWrappedConsumer(nullptr) - , m_nBytesPerNotif(0) - , m_nSourceFrameSizeInBytes(0) - , m_nMaxFrameCount(0) - , m_nPtime(0) - , m_bStarted(false) - , m_bPrepared(false) - , m_bPaused(false) + : m_pDevice(nullptr) + , m_hMutex(nullptr) + , m_pClient(nullptr) + , m_hEvent(nullptr) + , m_pAsyncThread(nullptr) + , m_pWrappedConsumer(nullptr) + , m_nBytesPerNotif(0) + , m_nSourceFrameSizeInBytes(0) + , m_nMaxFrameCount(0) + , m_nPtime(0) + , m_bStarted(false) + , m_bPrepared(false) + , m_bPaused(false) { - memset(&m_ring, 0, sizeof(m_ring)); + memset(&m_ring, 0, sizeof(m_ring)); - if (!(m_hMutex = tsk_mutex_create())) { - throw ref new Platform::FailureException(L"Failed to create mutex"); - } + if (!(m_hMutex = tsk_mutex_create())) { + throw ref new Platform::FailureException(L"Failed to create mutex"); + } } Doubango::VoIP::AudioRender::~AudioRender() { - Stop(); - UnPrepare(); + Stop(); + UnPrepare(); - tsk_mutex_destroy(&m_hMutex); + tsk_mutex_destroy(&m_hMutex); } int Doubango::VoIP::AudioRender::Prepare(tdav_consumer_wasapi_t* wasapi, const tmedia_codec_t* codec) { - HRESULT hr = E_FAIL; - int ret = 0; - WAVEFORMATEX wfx = {0}; - AudioClientProperties properties = {0}; - LPCWSTR pwstrRenderId = nullptr; + HRESULT hr = E_FAIL; + int ret = 0; + WAVEFORMATEX wfx = {0}; + AudioClientProperties properties = {0}; + LPCWSTR pwstrRenderId = nullptr; - #define WASAPI_SET_ERROR(code) ret = (code); goto bail; +#define WASAPI_SET_ERROR(code) ret = (code); goto bail; - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - if (m_bPrepared) { - WASAPI_DEBUG_INFO("Already prepared"); - goto bail; - } + if (m_bPrepared) { + WASAPI_DEBUG_INFO("Already prepared"); + goto bail; + } - if (!wasapi || !codec) { - WASAPI_DEBUG_ERROR("Invalid parameter"); - WASAPI_SET_ERROR(-1); - } + if (!wasapi || !codec) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + WASAPI_SET_ERROR(-1); + } - if (m_pDevice || m_pClient) { - WASAPI_DEBUG_ERROR("consumer already prepared"); - WASAPI_SET_ERROR(-2); - } + if (m_pDevice || m_pClient) { + WASAPI_DEBUG_ERROR("consumer already prepared"); + WASAPI_SET_ERROR(-2); + } pwstrRenderId = GetDefaultAudioRenderId(AudioDeviceRole::Communications); if (NULL == pwstrRenderId) { - tdav_win32_print_error("GetDefaultAudioRenderId", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-3); + tdav_win32_print_error("GetDefaultAudioRenderId", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-3); } hr = ActivateAudioInterface(pwstrRenderId, __uuidof(IAudioClient2), (void**)&m_pDevice); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-4); - } - + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-4); + } + if (SUCCEEDED(hr)) { properties.cbSize = sizeof AudioClientProperties; properties.eCategory = AudioCategory_Communications; hr = m_pDevice->SetClientProperties(&properties); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("SetClientProperties", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-5); - } - } - else { - tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-6); - } - - /* Set best format */ - { - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = TMEDIA_CONSUMER(wasapi)->audio.in.channels; - wfx.nSamplesPerSec = TMEDIA_CONSUMER(wasapi)->audio.in.rate; - wfx.wBitsPerSample = TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample; - wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); - wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); - - PWAVEFORMATEX pwfxClosestMatch = NULL; - hr = m_pDevice->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &wfx, &pwfxClosestMatch); - if (hr != S_OK && hr != S_FALSE) { - tdav_win32_print_error("IsFormatSupported", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-8); - } - - if (hr == S_FALSE) { - if (!pwfxClosestMatch) { - WASAPI_DEBUG_ERROR("malloc(%d) failed", sizeof(WAVEFORMATEX)); - WASAPI_SET_ERROR(-7); - } - - wfx.nSamplesPerSec = pwfxClosestMatch->nSamplesPerSec; - wfx.nChannels = pwfxClosestMatch->nChannels; + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("SetClientProperties", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-5); + } + } + else { + tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-6); + } + + /* Set best format */ + { + wfx.wFormatTag = WAVE_FORMAT_PCM; + wfx.nChannels = TMEDIA_CONSUMER(wasapi)->audio.in.channels; + wfx.nSamplesPerSec = TMEDIA_CONSUMER(wasapi)->audio.in.rate; + wfx.wBitsPerSample = TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample; + wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); + wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); + + PWAVEFORMATEX pwfxClosestMatch = NULL; + hr = m_pDevice->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &wfx, &pwfxClosestMatch); + if (hr != S_OK && hr != S_FALSE) { + tdav_win32_print_error("IsFormatSupported", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-8); + } + + if (hr == S_FALSE) { + if (!pwfxClosestMatch) { + WASAPI_DEBUG_ERROR("malloc(%d) failed", sizeof(WAVEFORMATEX)); + WASAPI_SET_ERROR(-7); + } + + wfx.nSamplesPerSec = pwfxClosestMatch->nSamplesPerSec; + wfx.nChannels = pwfxClosestMatch->nChannels; #if 0 - wfx.wBitsPerSample = pwfxClosestMatch->wBitsPerSample; + wfx.wBitsPerSample = pwfxClosestMatch->wBitsPerSample; #endif - wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8); - wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; - // Request resampler - TMEDIA_CONSUMER(wasapi)->audio.out.rate = (uint32_t)wfx.nSamplesPerSec; - TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample = (uint8_t)wfx.wBitsPerSample; - TMEDIA_CONSUMER(wasapi)->audio.out.channels = (uint8_t)wfx.nChannels; - - WASAPI_DEBUG_INFO("Audio device format fallback: rate=%d, bps=%d, channels=%d", wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nChannels); - } - if (pwfxClosestMatch) { - CoTaskMemFree(pwfxClosestMatch); - } - } - - m_nSourceFrameSizeInBytes = (wfx.wBitsPerSample >> 3) * wfx.nChannels; - m_nBytesPerNotif = ((wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(wasapi)->audio.ptime) / 1000); - - // Initialize - hr = m_pDevice->Initialize( - AUDCLNT_SHAREMODE_SHARED, - AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - WASAPI_MILLIS_TO_100NS(TDAV_WASAPI_CONSUMER_NOTIF_POS_COUNT * TMEDIA_CONSUMER(wasapi)->audio.ptime), - 0, - &wfx, - NULL); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("#WASAPI: Render::Initialize", hr); - WASAPI_SET_ERROR(-9); - } - - REFERENCE_TIME DefaultDevicePeriod, MinimumDevicePeriod; - hr = m_pDevice->GetDevicePeriod(&DefaultDevicePeriod, &MinimumDevicePeriod); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("GetDevicePeriod", hr); - WASAPI_SET_ERROR(-10); - } - hr = m_pDevice->GetBufferSize(&m_nMaxFrameCount); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("GetBufferSize", hr); - WASAPI_SET_ERROR(-10); - } - - WASAPI_DEBUG_INFO("#WASAPI (Playback): BufferSize=%u, DefaultDevicePeriod=%lld ms, MinimumDevicePeriod=%lldms", m_nMaxFrameCount, WASAPI_100NS_TO_MILLIS(DefaultDevicePeriod), WASAPI_100NS_TO_MILLIS(MinimumDevicePeriod)); - - if (!m_hEvent) { - if (!(m_hEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE))) { - tdav_win32_print_error("CreateEventEx(EVENT_MODIFY_STATE | SYNCHRONIZE)", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-11); - } - } - - hr = m_pDevice->SetEventHandle(m_hEvent); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("SetEventHandle", hr); - WASAPI_SET_ERROR(-12); - } - + wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8); + wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; + // Request resampler + TMEDIA_CONSUMER(wasapi)->audio.out.rate = (uint32_t)wfx.nSamplesPerSec; + TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample = (uint8_t)wfx.wBitsPerSample; + TMEDIA_CONSUMER(wasapi)->audio.out.channels = (uint8_t)wfx.nChannels; + + WASAPI_DEBUG_INFO("Audio device format fallback: rate=%d, bps=%d, channels=%d", wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nChannels); + } + if (pwfxClosestMatch) { + CoTaskMemFree(pwfxClosestMatch); + } + } + + m_nSourceFrameSizeInBytes = (wfx.wBitsPerSample >> 3) * wfx.nChannels; + m_nBytesPerNotif = ((wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(wasapi)->audio.ptime) / 1000); + + // Initialize + hr = m_pDevice->Initialize( + AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + WASAPI_MILLIS_TO_100NS(TDAV_WASAPI_CONSUMER_NOTIF_POS_COUNT * TMEDIA_CONSUMER(wasapi)->audio.ptime), + 0, + &wfx, + NULL); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("#WASAPI: Render::Initialize", hr); + WASAPI_SET_ERROR(-9); + } + + REFERENCE_TIME DefaultDevicePeriod, MinimumDevicePeriod; + hr = m_pDevice->GetDevicePeriod(&DefaultDevicePeriod, &MinimumDevicePeriod); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("GetDevicePeriod", hr); + WASAPI_SET_ERROR(-10); + } + hr = m_pDevice->GetBufferSize(&m_nMaxFrameCount); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("GetBufferSize", hr); + WASAPI_SET_ERROR(-10); + } + + WASAPI_DEBUG_INFO("#WASAPI (Playback): BufferSize=%u, DefaultDevicePeriod=%lld ms, MinimumDevicePeriod=%lldms", m_nMaxFrameCount, WASAPI_100NS_TO_MILLIS(DefaultDevicePeriod), WASAPI_100NS_TO_MILLIS(MinimumDevicePeriod)); + + if (!m_hEvent) { + if (!(m_hEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE))) { + tdav_win32_print_error("CreateEventEx(EVENT_MODIFY_STATE | SYNCHRONIZE)", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-11); + } + } + + hr = m_pDevice->SetEventHandle(m_hEvent); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("SetEventHandle", hr); + WASAPI_SET_ERROR(-12); + } + hr = m_pDevice->GetService(__uuidof(IAudioRenderClient), (void**)&m_pClient); - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("GetService", hr); - WASAPI_SET_ERROR(-14); - } - - m_ring.chunck.size = (TMEDIA_CONSUMER(wasapi)->audio.ptime * TMEDIA_CONSUMER(wasapi)->audio.out.rate * ((TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample >> 3) * TMEDIA_CONSUMER(wasapi)->audio.out.channels)) / 1000; - m_ring.size = TDAV_WASAPI_CONSUMER_NOTIF_POS_COUNT * m_ring.chunck.size; - if (!(m_ring.chunck.buffer = tsk_realloc(m_ring.chunck.buffer, m_ring.chunck.size))) { - m_ring.size = 0; - WASAPI_DEBUG_ERROR("Failed to allocate new buffer"); - WASAPI_SET_ERROR(-15); - } - if (!m_ring.buffer) { - m_ring.buffer = speex_buffer_init(m_ring.size); - } - else { - int sret; - if ((sret = speex_buffer_resize(m_ring.buffer, m_ring.size)) < 0) { - WASAPI_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", m_ring.size, sret); - WASAPI_SET_ERROR(-16); - } - } - if (!m_ring.buffer) { - WASAPI_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", m_ring.size); - WASAPI_SET_ERROR(-17); - } + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("GetService", hr); + WASAPI_SET_ERROR(-14); + } + + m_ring.chunck.size = (TMEDIA_CONSUMER(wasapi)->audio.ptime * TMEDIA_CONSUMER(wasapi)->audio.out.rate * ((TMEDIA_CONSUMER(wasapi)->audio.bits_per_sample >> 3) * TMEDIA_CONSUMER(wasapi)->audio.out.channels)) / 1000; + m_ring.size = TDAV_WASAPI_CONSUMER_NOTIF_POS_COUNT * m_ring.chunck.size; + if (!(m_ring.chunck.buffer = tsk_realloc(m_ring.chunck.buffer, m_ring.chunck.size))) { + m_ring.size = 0; + WASAPI_DEBUG_ERROR("Failed to allocate new buffer"); + WASAPI_SET_ERROR(-15); + } + if (!m_ring.buffer) { + m_ring.buffer = speex_buffer_init(m_ring.size); + } + else { + int sret; + if ((sret = speex_buffer_resize(m_ring.buffer, m_ring.size)) < 0) { + WASAPI_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", m_ring.size, sret); + WASAPI_SET_ERROR(-16); + } + } + if (!m_ring.buffer) { + WASAPI_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", m_ring.size); + WASAPI_SET_ERROR(-17); + } bail: - if (pwstrRenderId) { + if (pwstrRenderId) { CoTaskMemFree((LPVOID)pwstrRenderId); } - if (ret != 0) { - UnPrepare(); - } + if (ret != 0) { + UnPrepare(); + } - if ((m_bPrepared = (ret == 0))) { - m_pWrappedConsumer = wasapi; - m_nPtime = TMEDIA_CONSUMER(wasapi)->audio.ptime; - } + if ((m_bPrepared = (ret == 0))) { + m_pWrappedConsumer = wasapi; + m_nPtime = TMEDIA_CONSUMER(wasapi)->audio.ptime; + } - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return ret; + return ret; } int Doubango::VoIP::AudioRender::UnPrepare() { - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - if (m_hEvent) { - CloseHandle(m_hEvent), m_hEvent = nullptr; - } - if (m_pDevice) { - m_pDevice->Release(), m_pDevice = nullptr; - } - if (m_pClient) { - m_pClient->Release(), m_pClient = nullptr; - } + if (m_hEvent) { + CloseHandle(m_hEvent), m_hEvent = nullptr; + } + if (m_pDevice) { + m_pDevice->Release(), m_pDevice = nullptr; + } + if (m_pClient) { + m_pClient->Release(), m_pClient = nullptr; + } - TSK_FREE(m_ring.chunck.buffer); - if (m_ring.buffer) { - speex_buffer_destroy(m_ring.buffer); - m_ring.buffer = nullptr; - } + TSK_FREE(m_ring.chunck.buffer); + if (m_ring.buffer) { + speex_buffer_destroy(m_ring.buffer); + m_ring.buffer = nullptr; + } - m_pWrappedConsumer = nullptr; + m_pWrappedConsumer = nullptr; - m_bPrepared = false; + m_bPrepared = false; - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return 0; + return 0; } int Doubango::VoIP::AudioRender::Start() { - tsk_mutex_lock(m_hMutex); - - if (m_bStarted) { - WASAPI_DEBUG_INFO("already started"); - goto bail; - } - if (!m_bPrepared) { - WASAPI_DEBUG_ERROR("not prepared"); - goto bail; - } - - m_pAsyncThread = Windows::System::Threading::ThreadPool::RunAsync(ref new Windows::System::Threading::WorkItemHandler(this, &Doubango::VoIP::AudioRender::AsyncThread), - Windows::System::Threading::WorkItemPriority::High, - Windows::System::Threading::WorkItemOptions::TimeSliced); - - if ((m_bStarted = (m_pAsyncThread != nullptr))) { - HRESULT hr = m_pDevice->Start(); - if(!SUCCEEDED(hr)) { - tdav_win32_print_error("Device::Start", hr); - Stop(); - } - m_bPaused = false; - } - + tsk_mutex_lock(m_hMutex); + + if (m_bStarted) { + WASAPI_DEBUG_INFO("already started"); + goto bail; + } + if (!m_bPrepared) { + WASAPI_DEBUG_ERROR("not prepared"); + goto bail; + } + + m_pAsyncThread = Windows::System::Threading::ThreadPool::RunAsync(ref new Windows::System::Threading::WorkItemHandler(this, &Doubango::VoIP::AudioRender::AsyncThread), + Windows::System::Threading::WorkItemPriority::High, + Windows::System::Threading::WorkItemOptions::TimeSliced); + + if ((m_bStarted = (m_pAsyncThread != nullptr))) { + HRESULT hr = m_pDevice->Start(); + if(!SUCCEEDED(hr)) { + tdav_win32_print_error("Device::Start", hr); + Stop(); + } + m_bPaused = false; + } + bail: - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); return (m_bStarted ? 0 : -2); } int Doubango::VoIP::AudioRender::Stop() { - m_bStarted = false; + m_bStarted = false; - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - if (m_hEvent) { - SetEvent(m_hEvent); + if (m_hEvent) { + SetEvent(m_hEvent); } if (m_pAsyncThread) { @@ -491,118 +490,118 @@ int Doubango::VoIP::AudioRender::Stop() m_pAsyncThread = nullptr; } - if (m_pDevice) { - m_pDevice->Stop(); - } + if (m_pDevice) { + m_pDevice->Stop(); + } - // will be prepared again before next start() - UnPrepare(); + // will be prepared again before next start() + UnPrepare(); - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return 0; + return 0; } int Doubango::VoIP::AudioRender::Pause() { - m_bPaused = true; + m_bPaused = true; - return 0; + return 0; } int Doubango::VoIP::AudioRender::Consume(const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - int ret; - // tsk_mutex_lock(m_hMutex); - ret = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer), buffer, size, proto_hdr); // thread-safe - // tsk_mutex_unlock(m_hMutex); - return ret; + int ret; + // tsk_mutex_lock(m_hMutex); + ret = tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer), buffer, size, proto_hdr); // thread-safe + // tsk_mutex_unlock(m_hMutex); + return ret; } tsk_size_t Doubango::VoIP::AudioRender::Read(void* data, tsk_size_t size) { - tsk_ssize_t retSize = 0, availSize; - - m_ring.leftBytes += size; - while (m_ring.leftBytes >= (tsk_ssize_t)m_ring.chunck.size) { - m_ring.leftBytes -= m_ring.chunck.size; - retSize = (tsk_ssize_t)tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer), m_ring.chunck.buffer, m_ring.chunck.size); - tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer)); - speex_buffer_write(m_ring.buffer, m_ring.chunck.buffer, retSize); - } - // IMPORTANT: looks like there is a bug in speex: continously trying to read more than avail - // many times can corrupt the buffer. At least on OS X 1.5 + tsk_ssize_t retSize = 0, availSize; + + m_ring.leftBytes += size; + while (m_ring.leftBytes >= (tsk_ssize_t)m_ring.chunck.size) { + m_ring.leftBytes -= m_ring.chunck.size; + retSize = (tsk_ssize_t)tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer), m_ring.chunck.buffer, m_ring.chunck.size); + tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(m_pWrappedConsumer)); + speex_buffer_write(m_ring.buffer, m_ring.chunck.buffer, retSize); + } + // IMPORTANT: looks like there is a bug in speex: continously trying to read more than avail + // many times can corrupt the buffer. At least on OS X 1.5 #if 0 - if (speex_buffer_get_available(m_ring.buffer) >= (tsk_ssize_t)size) { - retSize = speex_buffer_read(m_ring.buffer, data, size); - } - else{ - memset(data, 0, size); - } + if (speex_buffer_get_available(m_ring.buffer) >= (tsk_ssize_t)size) { + retSize = speex_buffer_read(m_ring.buffer, data, size); + } + else { + memset(data, 0, size); + } #else - availSize = speex_buffer_get_available(m_ring.buffer); - if (availSize == 0) { - memset(data, 0, size); - } - else { - retSize = speex_buffer_read(m_ring.buffer, data, min(availSize, (tsk_ssize_t)size)); - if (availSize < (tsk_ssize_t)size) { - memset(((uint8_t*)data) + availSize, 0, (size - availSize)); - } - } + availSize = speex_buffer_get_available(m_ring.buffer); + if (availSize == 0) { + memset(data, 0, size); + } + else { + retSize = speex_buffer_read(m_ring.buffer, data, min(availSize, (tsk_ssize_t)size)); + if (availSize < (tsk_ssize_t)size) { + memset(((uint8_t*)data) + availSize, 0, (size - availSize)); + } + } #endif - return retSize; + return retSize; } void Doubango::VoIP::AudioRender::AsyncThread(Windows::Foundation::IAsyncAction^ operation) { - HRESULT hr = S_OK; - INT32 nFramesToWrite; - UINT32 nPadding, nRead; - DWORD retval; - - WASAPI_DEBUG_INFO("#WASAPI: __playback_thread -- START"); - - #define BREAK_WHILE tsk_mutex_unlock(m_hMutex); break; - - while (m_bStarted && SUCCEEDED(hr)) { - retval = WaitForSingleObjectEx(m_hEvent, /*m_nPtime*/INFINITE, FALSE); - - tsk_mutex_lock(m_hMutex); - - if (!m_bStarted) { - BREAK_WHILE; - } - - if (retval == WAIT_OBJECT_0) { - hr = m_pDevice->GetCurrentPadding(&nPadding); - if (SUCCEEDED(hr)) { - BYTE* pRenderBuffer = NULL; - nFramesToWrite = m_nMaxFrameCount - nPadding; - - if (nFramesToWrite > 0) { - hr = m_pClient->GetBuffer(nFramesToWrite, &pRenderBuffer); - if (SUCCEEDED(hr)) { - nRead = Read(pRenderBuffer, (nFramesToWrite * m_nSourceFrameSizeInBytes)); - - // Release the buffer - hr = m_pClient->ReleaseBuffer(nFramesToWrite, (nRead == 0) ? AUDCLNT_BUFFERFLAGS_SILENT : 0); - } - } - } - } - - tsk_mutex_unlock(m_hMutex); - }// end-of-while - - if (!SUCCEEDED(hr)) { - tdav_win32_print_error("AsyncThread: ", hr); - } - - - WASAPI_DEBUG_INFO("__playback_thread(%s) -- STOP", (SUCCEEDED(hr) && retval == WAIT_OBJECT_0) ? "OK" : "NOK"); + HRESULT hr = S_OK; + INT32 nFramesToWrite; + UINT32 nPadding, nRead; + DWORD retval; + + WASAPI_DEBUG_INFO("#WASAPI: __playback_thread -- START"); + +#define BREAK_WHILE tsk_mutex_unlock(m_hMutex); break; + + while (m_bStarted && SUCCEEDED(hr)) { + retval = WaitForSingleObjectEx(m_hEvent, /*m_nPtime*/INFINITE, FALSE); + + tsk_mutex_lock(m_hMutex); + + if (!m_bStarted) { + BREAK_WHILE; + } + + if (retval == WAIT_OBJECT_0) { + hr = m_pDevice->GetCurrentPadding(&nPadding); + if (SUCCEEDED(hr)) { + BYTE* pRenderBuffer = NULL; + nFramesToWrite = m_nMaxFrameCount - nPadding; + + if (nFramesToWrite > 0) { + hr = m_pClient->GetBuffer(nFramesToWrite, &pRenderBuffer); + if (SUCCEEDED(hr)) { + nRead = Read(pRenderBuffer, (nFramesToWrite * m_nSourceFrameSizeInBytes)); + + // Release the buffer + hr = m_pClient->ReleaseBuffer(nFramesToWrite, (nRead == 0) ? AUDCLNT_BUFFERFLAGS_SILENT : 0); + } + } + } + } + + tsk_mutex_unlock(m_hMutex); + }// end-of-while + + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("AsyncThread: ", hr); + } + + + WASAPI_DEBUG_INFO("__playback_thread(%s) -- STOP", (SUCCEEDED(hr) && retval == WAIT_OBJECT_0) ? "OK" : "NOK"); } @@ -617,56 +616,54 @@ void Doubango::VoIP::AudioRender::AsyncThread(Windows::Foundation::IAsyncAction^ /* constructor */ static tsk_object_t* tdav_consumer_wasapi_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_wasapi_t *wasapi = (tdav_consumer_wasapi_t*)self; - if (wasapi) { - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(wasapi)); - /* init self */ - - wasapi->AudioRender = ref new Doubango::VoIP::AudioRender(); - } - return self; + tdav_consumer_wasapi_t *wasapi = (tdav_consumer_wasapi_t*)self; + if (wasapi) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(wasapi)); + /* init self */ + + wasapi->AudioRender = ref new Doubango::VoIP::AudioRender(); + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_wasapi_dtor(tsk_object_t * self) -{ - tdav_consumer_wasapi_t *wasapi = (tdav_consumer_wasapi_t*)self; - if (wasapi) { - /* stop */ - tdav_consumer_wasapi_stop((tmedia_consumer_t*)self); - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(wasapi)); - /* deinit self */ - if (wasapi->AudioRender) { - delete wasapi->AudioRender; - wasapi->AudioRender = nullptr; - } - } - - return self; +{ + tdav_consumer_wasapi_t *wasapi = (tdav_consumer_wasapi_t*)self; + if (wasapi) { + /* stop */ + tdav_consumer_wasapi_stop((tmedia_consumer_t*)self); + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(wasapi)); + /* deinit self */ + if (wasapi->AudioRender) { + delete wasapi->AudioRender; + wasapi->AudioRender = nullptr; + } + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_wasapi_def_s = -{ - sizeof(tdav_consumer_wasapi_t), - tdav_consumer_wasapi_ctor, - tdav_consumer_wasapi_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_wasapi_def_s = { + sizeof(tdav_consumer_wasapi_t), + tdav_consumer_wasapi_ctor, + tdav_consumer_wasapi_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_wasapi_plugin_def_s = -{ - &tdav_consumer_wasapi_def_s, - - tmedia_audio, - "Microsoft Windows Audio Session API (WASAPI) consumer", - - tdav_consumer_wasapi_set, - tdav_consumer_wasapi_prepare, - tdav_consumer_wasapi_start, - tdav_consumer_wasapi_consume, - tdav_consumer_wasapi_pause, - tdav_consumer_wasapi_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_wasapi_plugin_def_s = { + &tdav_consumer_wasapi_def_s, + + tmedia_audio, + "Microsoft Windows Audio Session API (WASAPI) consumer", + + tdav_consumer_wasapi_set, + tdav_consumer_wasapi_prepare, + tdav_consumer_wasapi_start, + tdav_consumer_wasapi_consume, + tdav_consumer_wasapi_pause, + tdav_consumer_wasapi_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_wasapi_plugin_def_t = &tdav_consumer_wasapi_plugin_def_s; diff --git a/tinyDAV/src/audio/wasapi/tdav_producer_wasapi.cxx b/tinyDAV/src/audio/wasapi/tdav_producer_wasapi.cxx index 7d172a2..1a2b186 100755 --- a/tinyDAV/src/audio/wasapi/tdav_producer_wasapi.cxx +++ b/tinyDAV/src/audio/wasapi/tdav_producer_wasapi.cxx @@ -1,18 +1,18 @@ /*Copyright (C) 2013 Mamadou DIOP * Copyright (C) 2013-2014 Doubango Telecom <http://www.doubango.org> -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. */ @@ -24,7 +24,7 @@ #if HAVE_WASAPI -#include "tinydav/audio/tdav_producer_audio.h" +#include "tinydav/audio/tdav_producer_audio.h" #include "tsk_memory.h" #include "tsk_string.h" @@ -51,59 +51,58 @@ struct tdav_producer_wasapi_s; namespace Doubango { - namespace VoIP - { - ref class AudioCapture sealed - { - public: - virtual ~AudioCapture(); - internal: - AudioCapture(); - - int Prepare(struct tdav_producer_wasapi_s* wasapi, const tmedia_codec_t* codec); - int UnPrepare(); - int Start(); - int Stop(); - int Pause(); - - private: - void AsyncThread(Windows::Foundation::IAsyncAction^ operation); - - private: - tsk_mutex_handle_t* m_hMutex; - IAudioClient2* m_pDevice; - IAudioCaptureClient* m_pClient; - HANDLE m_hCaptureEvent; - HANDLE m_hShutdownEvent; - Windows::Foundation::IAsyncAction^ m_pAsyncThread; - INT32 m_nBytesPerNotif; - INT32 m_nSourceFrameSizeInBytes; - - struct{ - tmedia_producer_enc_cb_f fn; - const void* pcData; - } m_callback; - - struct { - struct { - void* buffer; - tsk_size_t size; - } chunck; - SpeexBuffer* buffer; - tsk_size_t size; - } m_ring; - bool m_bStarted; - bool m_bPrepared; - bool m_bPaused; - }; - } +namespace VoIP +{ +ref class AudioCapture sealed +{ +public: + virtual ~AudioCapture(); +internal: + AudioCapture(); + + int Prepare(struct tdav_producer_wasapi_s* wasapi, const tmedia_codec_t* codec); + int UnPrepare(); + int Start(); + int Stop(); + int Pause(); + +private: + void AsyncThread(Windows::Foundation::IAsyncAction^ operation); + +private: + tsk_mutex_handle_t* m_hMutex; + IAudioClient2* m_pDevice; + IAudioCaptureClient* m_pClient; + HANDLE m_hCaptureEvent; + HANDLE m_hShutdownEvent; + Windows::Foundation::IAsyncAction^ m_pAsyncThread; + INT32 m_nBytesPerNotif; + INT32 m_nSourceFrameSizeInBytes; + + struct { + tmedia_producer_enc_cb_f fn; + const void* pcData; + } m_callback; + + struct { + struct { + void* buffer; + tsk_size_t size; + } chunck; + SpeexBuffer* buffer; + tsk_size_t size; + } m_ring; + bool m_bStarted; + bool m_bPrepared; + bool m_bPaused; +}; +} } -typedef struct tdav_producer_wasapi_s -{ - TDAV_DECLARE_PRODUCER_AUDIO; +typedef struct tdav_producer_wasapi_s { + TDAV_DECLARE_PRODUCER_AUDIO; - Doubango::VoIP::AudioCapture ^audioCapture; + Doubango::VoIP::AudioCapture ^audioCapture; } tdav_producer_wasapi_t; @@ -112,94 +111,94 @@ extern "C" void tdav_win32_print_error(const char* func, HRESULT hr); /* ============ Media Producer Interface ================= */ static int tdav_producer_wasapi_set(tmedia_producer_t* self, const tmedia_param_t* param) -{ - tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; - if (param->plugin_type == tmedia_ppt_producer) { - if (param->value_type == tmedia_pvt_int32) { - if (tsk_striequals(param->key, "volume")) { - return 0; - } - else if (tsk_striequals(param->key, "mute")) { - //wasapi->mute = (TSK_TO_INT32((uint8_t*)param->value) != 0); +{ + tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; + if (param->plugin_type == tmedia_ppt_producer) { + if (param->value_type == tmedia_pvt_int32) { + if (tsk_striequals(param->key, "volume")) { + return 0; + } + else if (tsk_striequals(param->key, "mute")) { + //wasapi->mute = (TSK_TO_INT32((uint8_t*)param->value) != 0); #if !FIXME_SEND_SILENCE_ON_MUTE - //if(wasapi->started){ - // if(wasapi->mute){ - //IDirectSoundCaptureBuffer_Stop(wasapi->captureBuffer); - // } - // else{ - //IDirectSoundCaptureBuffer_Start(wasapi->captureBuffer, DSBPLAY_LOOPING); - // } - //} + //if(wasapi->started){ + // if(wasapi->mute){ + //IDirectSoundCaptureBuffer_Stop(wasapi->captureBuffer); + // } + // else{ + //IDirectSoundCaptureBuffer_Start(wasapi->captureBuffer, DSBPLAY_LOOPING); + // } + //} #endif - return 0; - } - } - } - return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); + return 0; + } + } + } + return tdav_producer_audio_set(TDAV_PRODUCER_AUDIO(self), param); } static int tdav_producer_wasapi_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; + tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; - if(!wasapi || !codec || !wasapi->audioCapture){ - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!wasapi || !codec || !wasapi->audioCapture) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - /* codec should have ptime */ + /* codec should have ptime */ TMEDIA_PRODUCER(wasapi)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); TMEDIA_PRODUCER(wasapi)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); TMEDIA_PRODUCER(wasapi)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); - WASAPI_DEBUG_INFO("channels=%d, rate=%d, ptime=%d", - TMEDIA_PRODUCER(wasapi)->audio.channels, - TMEDIA_PRODUCER(wasapi)->audio.rate, - TMEDIA_PRODUCER(wasapi)->audio.ptime); + WASAPI_DEBUG_INFO("channels=%d, rate=%d, ptime=%d", + TMEDIA_PRODUCER(wasapi)->audio.channels, + TMEDIA_PRODUCER(wasapi)->audio.rate, + TMEDIA_PRODUCER(wasapi)->audio.ptime); - return wasapi->audioCapture->Prepare(wasapi, codec); + return wasapi->audioCapture->Prepare(wasapi, codec); } static int tdav_producer_wasapi_start(tmedia_producer_t* self) { - tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; + tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; - WASAPI_DEBUG_INFO("tdav_producer_wasapi_start()"); + WASAPI_DEBUG_INFO("tdav_producer_wasapi_start()"); - if(!wasapi || !wasapi->audioCapture){ - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!wasapi || !wasapi->audioCapture) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->audioCapture->Start(); + return wasapi->audioCapture->Start(); } static int tdav_producer_wasapi_pause(tmedia_producer_t* self) { - tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; + tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; - if(!wasapi || !wasapi->audioCapture){ - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!wasapi || !wasapi->audioCapture) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->audioCapture->Pause(); + return wasapi->audioCapture->Pause(); } static int tdav_producer_wasapi_stop(tmedia_producer_t* self) { - tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; + tdav_producer_wasapi_t* wasapi = (tdav_producer_wasapi_t*)self; - WASAPI_DEBUG_INFO("tdav_producer_wasapi_stop()"); + WASAPI_DEBUG_INFO("tdav_producer_wasapi_stop()"); - if(!wasapi || !wasapi->audioCapture){ - WASAPI_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!wasapi || !wasapi->audioCapture) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return wasapi->audioCapture->Stop(); + return wasapi->audioCapture->Stop(); } @@ -209,406 +208,389 @@ static int tdav_producer_wasapi_stop(tmedia_producer_t* self) Doubango::VoIP::AudioCapture::AudioCapture() - : m_pDevice(nullptr) - , m_hMutex(nullptr) - , m_pClient(nullptr) - , m_hCaptureEvent(nullptr) - , m_hShutdownEvent(nullptr) - , m_pAsyncThread(nullptr) - , m_nBytesPerNotif(0) - , m_nSourceFrameSizeInBytes(0) - , m_bStarted(false) - , m_bPrepared(false) - , m_bPaused(false) + : m_pDevice(nullptr) + , m_hMutex(nullptr) + , m_pClient(nullptr) + , m_hCaptureEvent(nullptr) + , m_hShutdownEvent(nullptr) + , m_pAsyncThread(nullptr) + , m_nBytesPerNotif(0) + , m_nSourceFrameSizeInBytes(0) + , m_bStarted(false) + , m_bPrepared(false) + , m_bPaused(false) { - m_callback.fn = nullptr, m_callback.pcData = nullptr; - memset(&m_ring, 0, sizeof(m_ring)); + m_callback.fn = nullptr, m_callback.pcData = nullptr; + memset(&m_ring, 0, sizeof(m_ring)); - if(!(m_hMutex = tsk_mutex_create())){ - throw ref new Platform::FailureException(L"Failed to create mutex"); - } + if(!(m_hMutex = tsk_mutex_create())) { + throw ref new Platform::FailureException(L"Failed to create mutex"); + } } Doubango::VoIP::AudioCapture::~AudioCapture() { - Stop(); - UnPrepare(); + Stop(); + UnPrepare(); - tsk_mutex_destroy(&m_hMutex); + tsk_mutex_destroy(&m_hMutex); } int Doubango::VoIP::AudioCapture::Prepare(tdav_producer_wasapi_t* wasapi, const tmedia_codec_t* codec) { - HRESULT hr = E_FAIL; - int ret = 0; - WAVEFORMATEX wfx = {0}; - AudioClientProperties properties = {0}; - LPCWSTR pwstrCaptureId = nullptr; + HRESULT hr = E_FAIL; + int ret = 0; + WAVEFORMATEX wfx = {0}; + AudioClientProperties properties = {0}; + LPCWSTR pwstrCaptureId = nullptr; - #define WASAPI_SET_ERROR(code) ret = (code); goto bail; +#define WASAPI_SET_ERROR(code) ret = (code); goto bail; - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - if(m_bPrepared) - { - WASAPI_DEBUG_INFO("#WASAPI: Audio producer already prepared"); - goto bail; - } + if(m_bPrepared) { + WASAPI_DEBUG_INFO("#WASAPI: Audio producer already prepared"); + goto bail; + } - if(!wasapi || !codec) - { - WASAPI_DEBUG_ERROR("Invalid parameter"); - WASAPI_SET_ERROR(-1); - } + if(!wasapi || !codec) { + WASAPI_DEBUG_ERROR("Invalid parameter"); + WASAPI_SET_ERROR(-1); + } - if(m_pDevice || m_pClient){ - WASAPI_DEBUG_ERROR("Producer already prepared"); - WASAPI_SET_ERROR(-2); - } + if(m_pDevice || m_pClient) { + WASAPI_DEBUG_ERROR("Producer already prepared"); + WASAPI_SET_ERROR(-2); + } pwstrCaptureId = GetDefaultAudioCaptureId(AudioDeviceRole::Communications); - if (NULL == pwstrCaptureId){ - tdav_win32_print_error("GetDefaultAudioCaptureId", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-3); + if (NULL == pwstrCaptureId) { + tdav_win32_print_error("GetDefaultAudioCaptureId", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-3); } hr = ActivateAudioInterface(pwstrCaptureId, __uuidof(IAudioClient2), (void**)&m_pDevice); - if(!SUCCEEDED(hr)){ - tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-4); - } - - if (SUCCEEDED(hr)){ + if(!SUCCEEDED(hr)) { + tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-4); + } + + if (SUCCEEDED(hr)) { properties.cbSize = sizeof AudioClientProperties; properties.eCategory = AudioCategory_Communications; hr = m_pDevice->SetClientProperties(&properties); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("SetClientProperties", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-5); - } - } - else{ - tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-6); - } - - /* Set best format */ - { - wfx.wFormatTag = WAVE_FORMAT_PCM; - wfx.nChannels = TMEDIA_PRODUCER(wasapi)->audio.channels; - wfx.nSamplesPerSec = TMEDIA_PRODUCER(wasapi)->audio.rate; - wfx.wBitsPerSample = TMEDIA_PRODUCER(wasapi)->audio.bits_per_sample; - wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); - wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); - - PWAVEFORMATEX pwfxClosestMatch = NULL; - hr = m_pDevice->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &wfx, &pwfxClosestMatch); - if(hr != S_OK && hr != S_FALSE) - { - tdav_win32_print_error("IsFormatSupported", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-8); - } - - if(hr == S_FALSE) - { - if(!pwfxClosestMatch) - { - WASAPI_DEBUG_ERROR("malloc(%d) failed", sizeof(WAVEFORMATEX)); - WASAPI_SET_ERROR(-7); - } - wfx.nChannels = pwfxClosestMatch->nChannels; - wfx.nSamplesPerSec = pwfxClosestMatch->nSamplesPerSec; + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("SetClientProperties", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-5); + } + } + else { + tdav_win32_print_error("ActivateAudioInterface", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-6); + } + + /* Set best format */ + { + wfx.wFormatTag = WAVE_FORMAT_PCM; + wfx.nChannels = TMEDIA_PRODUCER(wasapi)->audio.channels; + wfx.nSamplesPerSec = TMEDIA_PRODUCER(wasapi)->audio.rate; + wfx.wBitsPerSample = TMEDIA_PRODUCER(wasapi)->audio.bits_per_sample; + wfx.nBlockAlign = (wfx.nChannels * wfx.wBitsPerSample/8); + wfx.nAvgBytesPerSec = (wfx.nSamplesPerSec * wfx.nBlockAlign); + + PWAVEFORMATEX pwfxClosestMatch = NULL; + hr = m_pDevice->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &wfx, &pwfxClosestMatch); + if(hr != S_OK && hr != S_FALSE) { + tdav_win32_print_error("IsFormatSupported", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-8); + } + + if(hr == S_FALSE) { + if(!pwfxClosestMatch) { + WASAPI_DEBUG_ERROR("malloc(%d) failed", sizeof(WAVEFORMATEX)); + WASAPI_SET_ERROR(-7); + } + wfx.nChannels = pwfxClosestMatch->nChannels; + wfx.nSamplesPerSec = pwfxClosestMatch->nSamplesPerSec; #if 0 - wfx.wBitsPerSample = pwfxClosestMatch->wBitsPerSample; + wfx.wBitsPerSample = pwfxClosestMatch->wBitsPerSample; #endif - wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8); - wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; - // Request resampler - TMEDIA_PRODUCER(wasapi)->audio.rate = (uint32_t)wfx.nSamplesPerSec; - TMEDIA_PRODUCER(wasapi)->audio.bits_per_sample = (uint8_t)wfx.wBitsPerSample; - TMEDIA_PRODUCER(wasapi)->audio.channels = (uint8_t)wfx.nChannels; - - WASAPI_DEBUG_INFO("Audio device format fallback: rate=%d, bps=%d, channels=%d", wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nChannels); - } - if(pwfxClosestMatch) - { - CoTaskMemFree(pwfxClosestMatch); - } - } - - m_nSourceFrameSizeInBytes = (wfx.wBitsPerSample >> 3) * wfx.nChannels; - m_nBytesPerNotif = ((wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(wasapi)->audio.ptime)/1000); - - // Initialize - hr = m_pDevice->Initialize( - AUDCLNT_SHAREMODE_SHARED, - AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - (TDAV_WASAPI_PRODUCER_NOTIF_POS_COUNT * WASAPI_MILLIS_TO_100NS(TMEDIA_PRODUCER(wasapi)->audio.ptime)), - 0, - &wfx, - NULL); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("#WASAPI: Capture::SetClientProperties", hr); - WASAPI_SET_ERROR(-9); - } - - REFERENCE_TIME DefaultDevicePeriod, MinimumDevicePeriod; - hr = m_pDevice->GetDevicePeriod(&DefaultDevicePeriod, &MinimumDevicePeriod); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("GetDevicePeriod", hr); - WASAPI_SET_ERROR(-10); - } - WASAPI_DEBUG_INFO("#WASAPI(Capture): DefaultDevicePeriod=%lld ms, MinimumDevicePeriod=%lldms", WASAPI_100NS_TO_MILLIS(DefaultDevicePeriod), WASAPI_100NS_TO_MILLIS(MinimumDevicePeriod)); - - if(!m_hCaptureEvent){ - if(!(m_hCaptureEvent = CreateEventEx(NULL, NULL, 0, EVENT_ALL_ACCESS))){ - tdav_win32_print_error("CreateEventEx(Capture)", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-11); - } - } - if(!m_hShutdownEvent){ - if(!(m_hShutdownEvent = CreateEventEx(NULL, NULL, CREATE_EVENT_MANUAL_RESET, EVENT_ALL_ACCESS))){ - tdav_win32_print_error("CreateEventEx(Shutdown)", HRESULT_FROM_WIN32(GetLastError())); - WASAPI_SET_ERROR(-12); - } - } - + wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8); + wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; + // Request resampler + TMEDIA_PRODUCER(wasapi)->audio.rate = (uint32_t)wfx.nSamplesPerSec; + TMEDIA_PRODUCER(wasapi)->audio.bits_per_sample = (uint8_t)wfx.wBitsPerSample; + TMEDIA_PRODUCER(wasapi)->audio.channels = (uint8_t)wfx.nChannels; + + WASAPI_DEBUG_INFO("Audio device format fallback: rate=%d, bps=%d, channels=%d", wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nChannels); + } + if(pwfxClosestMatch) { + CoTaskMemFree(pwfxClosestMatch); + } + } + + m_nSourceFrameSizeInBytes = (wfx.wBitsPerSample >> 3) * wfx.nChannels; + m_nBytesPerNotif = ((wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(wasapi)->audio.ptime)/1000); + + // Initialize + hr = m_pDevice->Initialize( + AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + (TDAV_WASAPI_PRODUCER_NOTIF_POS_COUNT * WASAPI_MILLIS_TO_100NS(TMEDIA_PRODUCER(wasapi)->audio.ptime)), + 0, + &wfx, + NULL); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("#WASAPI: Capture::SetClientProperties", hr); + WASAPI_SET_ERROR(-9); + } + + REFERENCE_TIME DefaultDevicePeriod, MinimumDevicePeriod; + hr = m_pDevice->GetDevicePeriod(&DefaultDevicePeriod, &MinimumDevicePeriod); + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("GetDevicePeriod", hr); + WASAPI_SET_ERROR(-10); + } + WASAPI_DEBUG_INFO("#WASAPI(Capture): DefaultDevicePeriod=%lld ms, MinimumDevicePeriod=%lldms", WASAPI_100NS_TO_MILLIS(DefaultDevicePeriod), WASAPI_100NS_TO_MILLIS(MinimumDevicePeriod)); + + if(!m_hCaptureEvent) { + if(!(m_hCaptureEvent = CreateEventEx(NULL, NULL, 0, EVENT_ALL_ACCESS))) { + tdav_win32_print_error("CreateEventEx(Capture)", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-11); + } + } + if(!m_hShutdownEvent) { + if(!(m_hShutdownEvent = CreateEventEx(NULL, NULL, CREATE_EVENT_MANUAL_RESET, EVENT_ALL_ACCESS))) { + tdav_win32_print_error("CreateEventEx(Shutdown)", HRESULT_FROM_WIN32(GetLastError())); + WASAPI_SET_ERROR(-12); + } + } + hr = m_pDevice->SetEventHandle(m_hCaptureEvent); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("SetEventHandle", hr); - WASAPI_SET_ERROR(-13); - } - + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("SetEventHandle", hr); + WASAPI_SET_ERROR(-13); + } + hr = m_pDevice->GetService(__uuidof(IAudioCaptureClient), (void**)&m_pClient); - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("GetService", hr); - WASAPI_SET_ERROR(-14); - } - - int packetperbuffer = (1000 / TMEDIA_PRODUCER(wasapi)->audio.ptime); - m_ring.chunck.size = wfx.nSamplesPerSec * (wfx.wBitsPerSample >> 3) / packetperbuffer; - WASAPI_DEBUG_INFO("#WASAPI: Audio producer ring chunk size = %u", m_ring.chunck.size); - // allocate our chunck buffer - if(!(m_ring.chunck.buffer = tsk_realloc(m_ring.chunck.buffer, m_ring.chunck.size))){ - WASAPI_DEBUG_ERROR("Failed to allocate new buffer"); - WASAPI_SET_ERROR(-15); - } - // create ringbuffer - m_ring.size = TDAV_WASAPI_PRODUCER_NOTIF_POS_COUNT * m_ring.chunck.size; - WASAPI_DEBUG_INFO("#WASAPI: Audio producer ring size = %u", m_ring.size); - if(!m_ring.buffer){ - m_ring.buffer = speex_buffer_init(m_ring.size); - } - else { - int sret; - if((sret = speex_buffer_resize(m_ring.buffer, m_ring.size)) < 0){ - WASAPI_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", m_ring.size, sret); - WASAPI_SET_ERROR(-16); - } - } - if(!m_ring.buffer){ - WASAPI_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", m_ring.size); - WASAPI_SET_ERROR(-17); - } - - m_callback.fn = TMEDIA_PRODUCER(wasapi)->enc_cb.callback; - m_callback.pcData = TMEDIA_PRODUCER(wasapi)->enc_cb.callback_data; + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("GetService", hr); + WASAPI_SET_ERROR(-14); + } + + int packetperbuffer = (1000 / TMEDIA_PRODUCER(wasapi)->audio.ptime); + m_ring.chunck.size = wfx.nSamplesPerSec * (wfx.wBitsPerSample >> 3) / packetperbuffer; + WASAPI_DEBUG_INFO("#WASAPI: Audio producer ring chunk size = %u", m_ring.chunck.size); + // allocate our chunck buffer + if(!(m_ring.chunck.buffer = tsk_realloc(m_ring.chunck.buffer, m_ring.chunck.size))) { + WASAPI_DEBUG_ERROR("Failed to allocate new buffer"); + WASAPI_SET_ERROR(-15); + } + // create ringbuffer + m_ring.size = TDAV_WASAPI_PRODUCER_NOTIF_POS_COUNT * m_ring.chunck.size; + WASAPI_DEBUG_INFO("#WASAPI: Audio producer ring size = %u", m_ring.size); + if(!m_ring.buffer) { + m_ring.buffer = speex_buffer_init(m_ring.size); + } + else { + int sret; + if((sret = speex_buffer_resize(m_ring.buffer, m_ring.size)) < 0) { + WASAPI_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", m_ring.size, sret); + WASAPI_SET_ERROR(-16); + } + } + if(!m_ring.buffer) { + WASAPI_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", m_ring.size); + WASAPI_SET_ERROR(-17); + } + + m_callback.fn = TMEDIA_PRODUCER(wasapi)->enc_cb.callback; + m_callback.pcData = TMEDIA_PRODUCER(wasapi)->enc_cb.callback_data; bail: - if (pwstrCaptureId){ + if (pwstrCaptureId) { CoTaskMemFree((LPVOID)pwstrCaptureId); } - if(ret != 0){ - UnPrepare(); - } - m_bPrepared = (ret == 0); + if(ret != 0) { + UnPrepare(); + } + m_bPrepared = (ret == 0); - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return ret; + return ret; } int Doubango::VoIP::AudioCapture::UnPrepare() { - tsk_mutex_lock(m_hMutex); - - if(m_hCaptureEvent) - { - CloseHandle(m_hCaptureEvent), m_hCaptureEvent = nullptr; - } - if(m_hShutdownEvent) - { - CloseHandle(m_hShutdownEvent), m_hShutdownEvent = nullptr; - } - if(m_pDevice) - { - m_pDevice->Release(), m_pDevice = nullptr; - } - if(m_pClient) - { - m_pClient->Release(), m_pClient = nullptr; - } - - TSK_FREE(m_ring.chunck.buffer); - if(m_ring.buffer){ - speex_buffer_destroy(m_ring.buffer); - m_ring.buffer = nullptr; - } - - m_callback.fn = nullptr; - m_callback.pcData = nullptr; - - m_bPrepared = false; - - tsk_mutex_unlock(m_hMutex); - - return 0; + tsk_mutex_lock(m_hMutex); + + if(m_hCaptureEvent) { + CloseHandle(m_hCaptureEvent), m_hCaptureEvent = nullptr; + } + if(m_hShutdownEvent) { + CloseHandle(m_hShutdownEvent), m_hShutdownEvent = nullptr; + } + if(m_pDevice) { + m_pDevice->Release(), m_pDevice = nullptr; + } + if(m_pClient) { + m_pClient->Release(), m_pClient = nullptr; + } + + TSK_FREE(m_ring.chunck.buffer); + if(m_ring.buffer) { + speex_buffer_destroy(m_ring.buffer); + m_ring.buffer = nullptr; + } + + m_callback.fn = nullptr; + m_callback.pcData = nullptr; + + m_bPrepared = false; + + tsk_mutex_unlock(m_hMutex); + + return 0; } int Doubango::VoIP::AudioCapture::Start() { - tsk_mutex_lock(m_hMutex); - - if(m_bStarted) - { - WASAPI_DEBUG_INFO("#WASAPI: Audio producer already started"); - goto bail; - } - if(!m_bPrepared) - { - WASAPI_DEBUG_ERROR("Audio producer not prepared"); - goto bail; - } - - m_pAsyncThread = Windows::System::Threading::ThreadPool::RunAsync(ref new Windows::System::Threading::WorkItemHandler(this, &Doubango::VoIP::AudioCapture::AsyncThread), - Windows::System::Threading::WorkItemPriority::High, - Windows::System::Threading::WorkItemOptions::TimeSliced); - - if((m_bStarted = (m_pAsyncThread != nullptr))) - { - HRESULT hr = m_pDevice->Start(); - if(!SUCCEEDED(hr)) - { - tdav_win32_print_error("Device::Start", hr); - Stop(); - } - m_bPaused = false; - } - + tsk_mutex_lock(m_hMutex); + + if(m_bStarted) { + WASAPI_DEBUG_INFO("#WASAPI: Audio producer already started"); + goto bail; + } + if(!m_bPrepared) { + WASAPI_DEBUG_ERROR("Audio producer not prepared"); + goto bail; + } + + m_pAsyncThread = Windows::System::Threading::ThreadPool::RunAsync(ref new Windows::System::Threading::WorkItemHandler(this, &Doubango::VoIP::AudioCapture::AsyncThread), + Windows::System::Threading::WorkItemPriority::High, + Windows::System::Threading::WorkItemOptions::TimeSliced); + + if((m_bStarted = (m_pAsyncThread != nullptr))) { + HRESULT hr = m_pDevice->Start(); + if(!SUCCEEDED(hr)) { + tdav_win32_print_error("Device::Start", hr); + Stop(); + } + m_bPaused = false; + } + bail: - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); return (m_bStarted ? 0 : -2); } int Doubango::VoIP::AudioCapture::Stop() { - m_bStarted = false; + m_bStarted = false; - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - if (m_hShutdownEvent) - { - SetEvent(m_hShutdownEvent); + if (m_hShutdownEvent) { + SetEvent(m_hShutdownEvent); } - if (m_pAsyncThread) - { + if (m_pAsyncThread) { m_pAsyncThread->Cancel(); m_pAsyncThread->Close(); m_pAsyncThread = nullptr; } - if(m_pDevice) - { - m_pDevice->Stop(); - } + if(m_pDevice) { + m_pDevice->Stop(); + } - // will be prepared again before next start() - UnPrepare(); + // will be prepared again before next start() + UnPrepare(); - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return 0; + return 0; } int Doubango::VoIP::AudioCapture::Pause() { - tsk_mutex_lock(m_hMutex); + tsk_mutex_lock(m_hMutex); - m_bPaused = true; + m_bPaused = true; - tsk_mutex_unlock(m_hMutex); + tsk_mutex_unlock(m_hMutex); - return 0; + return 0; } void Doubango::VoIP::AudioCapture::AsyncThread(Windows::Foundation::IAsyncAction^ operation) { - HRESULT hr = S_OK; - BYTE* pbData = nullptr; + HRESULT hr = S_OK; + BYTE* pbData = nullptr; UINT32 nFrames = 0; DWORD dwFlags = 0; - UINT32 incomingBufferSize; - INT32 avail; - UINT32 nNextPacketSize; - - HANDLE eventHandles[] = { - m_hCaptureEvent, // WAIT_OBJECT0 - m_hShutdownEvent // WAIT_OBJECT1 - }; - - WASAPI_DEBUG_INFO("#WASAPI: __record_thread -- START"); - - #define BREAK_WHILE tsk_mutex_unlock(m_hMutex); break; - - while(m_bStarted && SUCCEEDED(hr)){ - DWORD waitResult = WaitForMultipleObjectsEx(SIZEOF_ARRAY(eventHandles), eventHandles, FALSE, INFINITE, FALSE); - - tsk_mutex_lock(m_hMutex); - - if(!m_bStarted){ - BREAK_WHILE; - } - - if(waitResult == WAIT_OBJECT_0 && m_callback.fn) { - hr = m_pClient->GetNextPacketSize(&nNextPacketSize); - while(SUCCEEDED(hr) && nNextPacketSize >0){ - hr = m_pClient->GetBuffer(&pbData, &nFrames, &dwFlags, nullptr, nullptr); - if(SUCCEEDED(hr) && pbData && nFrames){ - if((dwFlags & AUDCLNT_BUFFERFLAGS_SILENT) != AUDCLNT_BUFFERFLAGS_SILENT){ - incomingBufferSize = nFrames * m_nSourceFrameSizeInBytes; - speex_buffer_write(m_ring.buffer, pbData, incomingBufferSize); - avail = speex_buffer_get_available(m_ring.buffer); - while (m_bStarted && avail >= (INT32)m_ring.chunck.size) { - avail -= speex_buffer_read(m_ring.buffer, m_ring.chunck.buffer, m_ring.chunck.size); - m_callback.fn(m_callback.pcData, m_ring.chunck.buffer, m_ring.chunck.size); - } - } - - if (SUCCEEDED(hr)){ - hr = m_pClient->ReleaseBuffer(nFrames); - } - if (SUCCEEDED(hr)){ - hr = m_pClient->GetNextPacketSize(&nNextPacketSize); - } - } - } - } - else if(waitResult != WAIT_OBJECT_0){ - BREAK_WHILE; - } - - tsk_mutex_unlock(m_hMutex); - }// end-of-while - - if (!SUCCEEDED(hr)){ - tdav_win32_print_error("AsyncThread: ", hr); - } - - - WASAPI_DEBUG_INFO("WASAPI: __record_thread(%s) -- STOP", SUCCEEDED(hr) ? "OK": "NOK"); + UINT32 incomingBufferSize; + INT32 avail; + UINT32 nNextPacketSize; + + HANDLE eventHandles[] = { + m_hCaptureEvent, // WAIT_OBJECT0 + m_hShutdownEvent // WAIT_OBJECT1 + }; + + WASAPI_DEBUG_INFO("#WASAPI: __record_thread -- START"); + +#define BREAK_WHILE tsk_mutex_unlock(m_hMutex); break; + + while(m_bStarted && SUCCEEDED(hr)) { + DWORD waitResult = WaitForMultipleObjectsEx(SIZEOF_ARRAY(eventHandles), eventHandles, FALSE, INFINITE, FALSE); + + tsk_mutex_lock(m_hMutex); + + if(!m_bStarted) { + BREAK_WHILE; + } + + if(waitResult == WAIT_OBJECT_0 && m_callback.fn) { + hr = m_pClient->GetNextPacketSize(&nNextPacketSize); + while(SUCCEEDED(hr) && nNextPacketSize >0) { + hr = m_pClient->GetBuffer(&pbData, &nFrames, &dwFlags, nullptr, nullptr); + if(SUCCEEDED(hr) && pbData && nFrames) { + if((dwFlags & AUDCLNT_BUFFERFLAGS_SILENT) != AUDCLNT_BUFFERFLAGS_SILENT) { + incomingBufferSize = nFrames * m_nSourceFrameSizeInBytes; + speex_buffer_write(m_ring.buffer, pbData, incomingBufferSize); + avail = speex_buffer_get_available(m_ring.buffer); + while (m_bStarted && avail >= (INT32)m_ring.chunck.size) { + avail -= speex_buffer_read(m_ring.buffer, m_ring.chunck.buffer, m_ring.chunck.size); + m_callback.fn(m_callback.pcData, m_ring.chunck.buffer, m_ring.chunck.size); + } + } + + if (SUCCEEDED(hr)) { + hr = m_pClient->ReleaseBuffer(nFrames); + } + if (SUCCEEDED(hr)) { + hr = m_pClient->GetNextPacketSize(&nNextPacketSize); + } + } + } + } + else if(waitResult != WAIT_OBJECT_0) { + BREAK_WHILE; + } + + tsk_mutex_unlock(m_hMutex); + }// end-of-while + + if (!SUCCEEDED(hr)) { + tdav_win32_print_error("AsyncThread: ", hr); + } + + + WASAPI_DEBUG_INFO("WASAPI: __record_thread(%s) -- STOP", SUCCEEDED(hr) ? "OK": "NOK"); } @@ -623,55 +605,53 @@ void Doubango::VoIP::AudioCapture::AsyncThread(Windows::Foundation::IAsyncAction /* constructor */ static tsk_object_t* tdav_producer_wasapi_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_wasapi_t *wasapi = (tdav_producer_wasapi_t*)self; - if(wasapi){ - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(wasapi)); - /* init self */ - - wasapi->audioCapture = ref new Doubango::VoIP::AudioCapture(); - } - return self; + tdav_producer_wasapi_t *wasapi = (tdav_producer_wasapi_t*)self; + if(wasapi) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(wasapi)); + /* init self */ + + wasapi->audioCapture = ref new Doubango::VoIP::AudioCapture(); + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_wasapi_dtor(tsk_object_t * self) -{ - tdav_producer_wasapi_t *wasapi = (tdav_producer_wasapi_t*)self; - if(wasapi){ - /* stop */ - tdav_producer_wasapi_stop((tmedia_producer_t*)self); - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(wasapi)); - /* deinit self */ - if(wasapi->audioCapture){ - delete wasapi->audioCapture; - wasapi->audioCapture = nullptr; - } - } - - return self; +{ + tdav_producer_wasapi_t *wasapi = (tdav_producer_wasapi_t*)self; + if(wasapi) { + /* stop */ + tdav_producer_wasapi_stop((tmedia_producer_t*)self); + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(wasapi)); + /* deinit self */ + if(wasapi->audioCapture) { + delete wasapi->audioCapture; + wasapi->audioCapture = nullptr; + } + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_wasapi_def_s = -{ - sizeof(tdav_producer_wasapi_t), - tdav_producer_wasapi_ctor, - tdav_producer_wasapi_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_wasapi_def_s = { + sizeof(tdav_producer_wasapi_t), + tdav_producer_wasapi_ctor, + tdav_producer_wasapi_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_wasapi_plugin_def_s = -{ - &tdav_producer_wasapi_def_s, - - tmedia_audio, - "Microsoft Windows Audio Session API (WASAPI) producer", - - tdav_producer_wasapi_set, - tdav_producer_wasapi_prepare, - tdav_producer_wasapi_start, - tdav_producer_wasapi_pause, - tdav_producer_wasapi_stop +static const tmedia_producer_plugin_def_t tdav_producer_wasapi_plugin_def_s = { + &tdav_producer_wasapi_def_s, + + tmedia_audio, + "Microsoft Windows Audio Session API (WASAPI) producer", + + tdav_producer_wasapi_set, + tdav_producer_wasapi_prepare, + tdav_producer_wasapi_start, + tdav_producer_wasapi_pause, + tdav_producer_wasapi_stop }; const tmedia_producer_plugin_def_t *tdav_producer_wasapi_plugin_def_t = &tdav_producer_wasapi_plugin_def_s; diff --git a/tinyDAV/src/audio/waveapi/tdav_consumer_waveapi.c b/tinyDAV/src/audio/waveapi/tdav_consumer_waveapi.c index 1883fa4..bf95818 100755 --- a/tinyDAV/src/audio/waveapi/tdav_consumer_waveapi.c +++ b/tinyDAV/src/audio/waveapi/tdav_consumer_waveapi.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2010-2015 Mamadou DIOP -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -36,151 +36,151 @@ static void print_last_error(MMRESULT mmrError, const char* func) { - static char buffer_err[TDAV_WAVEAPI_CONSUMER_ERROR_BUFF_COUNT]; + static char buffer_err[TDAV_WAVEAPI_CONSUMER_ERROR_BUFF_COUNT]; - waveOutGetErrorTextA(mmrError, buffer_err, sizeof(buffer_err)); - TSK_DEBUG_ERROR("%s() error: %s", func, buffer_err); + waveOutGetErrorTextA(mmrError, buffer_err, sizeof(buffer_err)); + TSK_DEBUG_ERROR("%s() error: %s", func, buffer_err); } static int free_wavehdr(tdav_consumer_waveapi_t* consumer, tsk_size_t index) { - if(!consumer || index >= sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!consumer || index >= sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - TSK_FREE(consumer->hWaveHeaders[index]->lpData); - TSK_FREE(consumer->hWaveHeaders[index]); + TSK_FREE(consumer->hWaveHeaders[index]->lpData); + TSK_FREE(consumer->hWaveHeaders[index]); - return 0; + return 0; } static int create_wavehdr(tdav_consumer_waveapi_t* consumer, tsk_size_t index) { - if(!consumer || index >= sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(consumer->hWaveHeaders[index]){ - free_wavehdr(consumer, index); - } - - consumer->hWaveHeaders[index] = tsk_calloc(1, sizeof(WAVEHDR)); - consumer->hWaveHeaders[index]->lpData = tsk_calloc(1, consumer->bytes_per_notif); - consumer->hWaveHeaders[index]->dwBufferLength = (DWORD)consumer->bytes_per_notif; - consumer->hWaveHeaders[index]->dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP; - consumer->hWaveHeaders[index]->dwLoops = 0x01; - consumer->hWaveHeaders[index]->dwUser = index; - - return 0; + if(!consumer || index >= sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(consumer->hWaveHeaders[index]) { + free_wavehdr(consumer, index); + } + + consumer->hWaveHeaders[index] = tsk_calloc(1, sizeof(WAVEHDR)); + consumer->hWaveHeaders[index]->lpData = tsk_calloc(1, consumer->bytes_per_notif); + consumer->hWaveHeaders[index]->dwBufferLength = (DWORD)consumer->bytes_per_notif; + consumer->hWaveHeaders[index]->dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP; + consumer->hWaveHeaders[index]->dwLoops = 0x01; + consumer->hWaveHeaders[index]->dwUser = index; + + return 0; } static int write_wavehdr(tdav_consumer_waveapi_t* consumer, tsk_size_t index) { - MMRESULT result; - - if(!consumer || !consumer->hWaveHeaders[index] || !consumer->hWaveOut){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - result = waveOutPrepareHeader(consumer->hWaveOut, consumer->hWaveHeaders[index], sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutPrepareHeader"); - return -2; - } - - result = waveOutWrite(consumer->hWaveOut, consumer->hWaveHeaders[index], sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutWrite"); - return -3; - } - - return 0; + MMRESULT result; + + if(!consumer || !consumer->hWaveHeaders[index] || !consumer->hWaveOut) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + result = waveOutPrepareHeader(consumer->hWaveOut, consumer->hWaveHeaders[index], sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutPrepareHeader"); + return -2; + } + + result = waveOutWrite(consumer->hWaveOut, consumer->hWaveHeaders[index], sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutWrite"); + return -3; + } + + return 0; } static int play_wavehdr(tdav_consumer_waveapi_t* consumer, LPWAVEHDR lpHdr) { - MMRESULT result; - tsk_size_t out_size; - - if(!consumer || !lpHdr || !consumer->hWaveOut){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - result = waveOutUnprepareHeader(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutUnprepareHeader"); - return -2; - } - - // - // - // Fill lpHdr->Data with decoded data - // - // - if((out_size = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), lpHdr->lpData, lpHdr->dwBufferLength))){ - //memcpy(lpHdr->lpData, data, lpHdr->dwBufferLength); - //TSK_FREE(data); - } - else{ - /* Put silence */ - memset(lpHdr->lpData, 0, lpHdr->dwBufferLength); - } - - if(!consumer->started){ - return 0; - } - - result = waveOutPrepareHeader(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutPrepareHeader"); - return -3; - } - - result = waveOutWrite(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutWrite"); - return -4; - } - - return 0; + MMRESULT result; + tsk_size_t out_size; + + if(!consumer || !lpHdr || !consumer->hWaveOut) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + result = waveOutUnprepareHeader(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutUnprepareHeader"); + return -2; + } + + // + // + // Fill lpHdr->Data with decoded data + // + // + if((out_size = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), lpHdr->lpData, lpHdr->dwBufferLength))) { + //memcpy(lpHdr->lpData, data, lpHdr->dwBufferLength); + //TSK_FREE(data); + } + else { + /* Put silence */ + memset(lpHdr->lpData, 0, lpHdr->dwBufferLength); + } + + if(!consumer->started) { + return 0; + } + + result = waveOutPrepareHeader(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutPrepareHeader"); + return -3; + } + + result = waveOutWrite(consumer->hWaveOut, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutWrite"); + return -4; + } + + return 0; } static void* TSK_STDCALL __playback_thread(void *param) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)param; - DWORD dwEvent; - tsk_size_t i; + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)param; + DWORD dwEvent; + tsk_size_t i; + + TSK_DEBUG_INFO("__playback_thread -- START"); - TSK_DEBUG_INFO("__playback_thread -- START"); + SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); - SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); + for(;;) { + dwEvent = WaitForMultipleObjects(2, consumer->events, FALSE, INFINITE); - for(;;){ - dwEvent = WaitForMultipleObjects(2, consumer->events, FALSE, INFINITE); + if (dwEvent == 1) { + break; + } - if (dwEvent == 1){ - break; - } + else if (dwEvent == 0) { + EnterCriticalSection(&consumer->cs); + for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR); i++) { + if(consumer->hWaveHeaders[i] && (consumer->hWaveHeaders[i]->dwFlags & WHDR_DONE)) { + play_wavehdr(consumer, consumer->hWaveHeaders[i]); + } + } + LeaveCriticalSection(&consumer->cs); + } + } - else if (dwEvent == 0){ - EnterCriticalSection(&consumer->cs); - for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR); i++){ - if(consumer->hWaveHeaders[i] && (consumer->hWaveHeaders[i]->dwFlags & WHDR_DONE)){ - play_wavehdr(consumer, consumer->hWaveHeaders[i]); - } - } - LeaveCriticalSection(&consumer->cs); - } - } + TSK_DEBUG_INFO("__playback_thread -- STOP"); - TSK_DEBUG_INFO("__playback_thread -- STOP"); - - return tsk_null; + return tsk_null; } @@ -193,137 +193,137 @@ static void* TSK_STDCALL __playback_thread(void *param) /* ============ Media Consumer Interface ================= */ int tdav_consumer_waveapi_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; - tsk_size_t i; - - if(!consumer || !codec && codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); - TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); - - /* codec should have ptime */ - - - /* Format */ - ZeroMemory(&consumer->wfx, sizeof(WAVEFORMATEX)); - consumer->wfx.wFormatTag = WAVE_FORMAT_PCM; - consumer->wfx.nChannels = TMEDIA_CONSUMER(consumer)->audio.in.channels; - consumer->wfx.nSamplesPerSec = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate; - consumer->wfx.wBitsPerSample = TMEDIA_CONSUMER(consumer)->audio.bits_per_sample; - consumer->wfx.nBlockAlign = (consumer->wfx.nChannels * consumer->wfx.wBitsPerSample/8); - consumer->wfx.nAvgBytesPerSec = (consumer->wfx.nSamplesPerSec * consumer->wfx.nBlockAlign); - - /* Average bytes (count) for each notification */ - consumer->bytes_per_notif = ((consumer->wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(consumer)->audio.ptime)/1000); - - /* create buffers */ - for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(consumer->hWaveHeaders[0]); i++){ - create_wavehdr(consumer, i); - } - - return 0; + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; + tsk_size_t i; + + if(!consumer || !codec && codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + TMEDIA_CONSUMER(consumer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.channels = TMEDIA_CODEC_CHANNELS_AUDIO_DECODING(codec); + TMEDIA_CONSUMER(consumer)->audio.in.rate = TMEDIA_CODEC_RATE_DECODING(codec); + + /* codec should have ptime */ + + + /* Format */ + ZeroMemory(&consumer->wfx, sizeof(WAVEFORMATEX)); + consumer->wfx.wFormatTag = WAVE_FORMAT_PCM; + consumer->wfx.nChannels = TMEDIA_CONSUMER(consumer)->audio.in.channels; + consumer->wfx.nSamplesPerSec = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate; + consumer->wfx.wBitsPerSample = TMEDIA_CONSUMER(consumer)->audio.bits_per_sample; + consumer->wfx.nBlockAlign = (consumer->wfx.nChannels * consumer->wfx.wBitsPerSample/8); + consumer->wfx.nAvgBytesPerSec = (consumer->wfx.nSamplesPerSec * consumer->wfx.nBlockAlign); + + /* Average bytes (count) for each notification */ + consumer->bytes_per_notif = ((consumer->wfx.nAvgBytesPerSec * TMEDIA_CONSUMER(consumer)->audio.ptime)/1000); + + /* create buffers */ + for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(consumer->hWaveHeaders[0]); i++) { + create_wavehdr(consumer, i); + } + + return 0; } int tdav_consumer_waveapi_start(tmedia_consumer_t* self) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; - MMRESULT result; - tsk_size_t i; - - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(consumer->started || consumer->hWaveOut){ - TSK_DEBUG_WARN("Consumer already started"); - return 0; - } - - /* create events */ - if(!consumer->events[0]){ - consumer->events[0] = CreateEvent(NULL, FALSE, FALSE, NULL); - } - if(!consumer->events[1]){ - consumer->events[1] = CreateEvent(NULL, FALSE, FALSE, NULL); - } - - /* open */ - result = waveOutOpen((HWAVEOUT *)&consumer->hWaveOut, WAVE_MAPPER, &consumer->wfx, (DWORD)consumer->events[0], (DWORD_PTR)consumer, CALLBACK_EVENT); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveOutOpen"); - return -2; - } - - /* write */ - for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(consumer->hWaveHeaders[0]); i++){ - write_wavehdr(consumer, i); - } - - /* start thread */ - consumer->started = tsk_true; - tsk_thread_create(&consumer->tid[0], __playback_thread, consumer); - - return 0; + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; + MMRESULT result; + tsk_size_t i; + + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(consumer->started || consumer->hWaveOut) { + TSK_DEBUG_WARN("Consumer already started"); + return 0; + } + + /* create events */ + if(!consumer->events[0]) { + consumer->events[0] = CreateEvent(NULL, FALSE, FALSE, NULL); + } + if(!consumer->events[1]) { + consumer->events[1] = CreateEvent(NULL, FALSE, FALSE, NULL); + } + + /* open */ + result = waveOutOpen((HWAVEOUT *)&consumer->hWaveOut, WAVE_MAPPER, &consumer->wfx, (DWORD)consumer->events[0], (DWORD_PTR)consumer, CALLBACK_EVENT); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveOutOpen"); + return -2; + } + + /* write */ + for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(consumer->hWaveHeaders[0]); i++) { + write_wavehdr(consumer, i); + } + + /* start thread */ + consumer->started = tsk_true; + tsk_thread_create(&consumer->tid[0], __playback_thread, consumer); + + return 0; } int tdav_consumer_waveapi_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; - - if(!consumer || !buffer || !size){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - /* buffer is already decoded */ - return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; + + if(!consumer || !buffer || !size) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + /* buffer is already decoded */ + return tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), buffer, size, proto_hdr); } int tdav_consumer_waveapi_pause(tmedia_consumer_t* self) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; - if(!consumer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!consumer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return 0; + return 0; } int tdav_consumer_waveapi_stop(tmedia_consumer_t* self) { - tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; - MMRESULT result; + tdav_consumer_waveapi_t* consumer = (tdav_consumer_waveapi_t*)self; + MMRESULT result; - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if(!consumer->started){ - TSK_DEBUG_WARN("Consumer not started"); - return 0; - } + if(!consumer->started) { + TSK_DEBUG_WARN("Consumer not started"); + return 0; + } - /* stop thread */ - if(consumer->tid[0]){ - SetEvent(consumer->events[1]); - tsk_thread_join(&(consumer->tid[0])); - } + /* stop thread */ + if(consumer->tid[0]) { + SetEvent(consumer->events[1]); + tsk_thread_join(&(consumer->tid[0])); + } - /* should be done here */ - consumer->started = tsk_false; + /* should be done here */ + consumer->started = tsk_false; - if(consumer->hWaveOut && ((result = waveOutReset(consumer->hWaveOut)) != MMSYSERR_NOERROR)){ - print_last_error(result, "waveOutReset"); - } + if(consumer->hWaveOut && ((result = waveOutReset(consumer->hWaveOut)) != MMSYSERR_NOERROR)) { + print_last_error(result, "waveOutReset"); + } - return 0; + return 0; } @@ -333,69 +333,67 @@ int tdav_consumer_waveapi_stop(tmedia_consumer_t* self) /* constructor */ static tsk_object_t* tdav_consumer_waveapi_ctor(tsk_object_t * self, va_list * app) { - tdav_consumer_waveapi_t *consumer = self; - if(consumer){ - /* init base */ - tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); - /* init self */ - InitializeCriticalSection(&consumer->cs); - } - return self; + tdav_consumer_waveapi_t *consumer = self; + if(consumer) { + /* init base */ + tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer)); + /* init self */ + InitializeCriticalSection(&consumer->cs); + } + return self; } /* destructor */ static tsk_object_t* tdav_consumer_waveapi_dtor(tsk_object_t * self) -{ - tdav_consumer_waveapi_t *consumer = self; - if(consumer){ - tsk_size_t i; - - /* stop */ - if(consumer->started){ - tdav_consumer_waveapi_stop(self); - } - - /* deinit base */ - tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); - /* deinit self */ - for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR); i++){ - free_wavehdr(consumer, i); - } - if(consumer->hWaveOut){ - waveOutClose(consumer->hWaveOut); - } - if(consumer->events[0]){ - CloseHandle(consumer->events[0]); - } - if(consumer->events[1]){ - CloseHandle(consumer->events[1]); - } - DeleteCriticalSection(&consumer->cs); - } - - return self; +{ + tdav_consumer_waveapi_t *consumer = self; + if(consumer) { + tsk_size_t i; + + /* stop */ + if(consumer->started) { + tdav_consumer_waveapi_stop(self); + } + + /* deinit base */ + tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer)); + /* deinit self */ + for(i = 0; i< sizeof(consumer->hWaveHeaders)/sizeof(LPWAVEHDR); i++) { + free_wavehdr(consumer, i); + } + if(consumer->hWaveOut) { + waveOutClose(consumer->hWaveOut); + } + if(consumer->events[0]) { + CloseHandle(consumer->events[0]); + } + if(consumer->events[1]) { + CloseHandle(consumer->events[1]); + } + DeleteCriticalSection(&consumer->cs); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_consumer_waveapi_def_s = -{ - sizeof(tdav_consumer_waveapi_t), - tdav_consumer_waveapi_ctor, - tdav_consumer_waveapi_dtor, - tdav_consumer_audio_cmp, +static const tsk_object_def_t tdav_consumer_waveapi_def_s = { + sizeof(tdav_consumer_waveapi_t), + tdav_consumer_waveapi_ctor, + tdav_consumer_waveapi_dtor, + tdav_consumer_audio_cmp, }; /* plugin definition*/ -static const tmedia_consumer_plugin_def_t tdav_consumer_waveapi_plugin_def_s = -{ - &tdav_consumer_waveapi_def_s, - - tmedia_audio, - "Microsoft WaveAPI consumer", - - tdav_consumer_waveapi_set, - tdav_consumer_waveapi_prepare, - tdav_consumer_waveapi_start, - tdav_consumer_waveapi_consume, - tdav_consumer_waveapi_pause, - tdav_consumer_waveapi_stop +static const tmedia_consumer_plugin_def_t tdav_consumer_waveapi_plugin_def_s = { + &tdav_consumer_waveapi_def_s, + + tmedia_audio, + "Microsoft WaveAPI consumer", + + tdav_consumer_waveapi_set, + tdav_consumer_waveapi_prepare, + tdav_consumer_waveapi_start, + tdav_consumer_waveapi_consume, + tdav_consumer_waveapi_pause, + tdav_consumer_waveapi_stop }; const tmedia_consumer_plugin_def_t *tdav_consumer_waveapi_plugin_def_t = &tdav_consumer_waveapi_plugin_def_s; diff --git a/tinyDAV/src/audio/waveapi/tdav_producer_waveapi.c b/tinyDAV/src/audio/waveapi/tdav_producer_waveapi.c index d077790..375668c 100755 --- a/tinyDAV/src/audio/waveapi/tdav_producer_waveapi.c +++ b/tinyDAV/src/audio/waveapi/tdav_producer_waveapi.c @@ -1,18 +1,18 @@ /* * Copyright (C) 2010-2015 Mamadou DIOP. -* +* * This file is part of Open Source Doubango Framework. * * DOUBANGO is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. -* +* * DOUBANGO is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. -* +* * You should have received a copy of the GNU General Public License * along with DOUBANGO. * @@ -35,150 +35,152 @@ static void print_last_error(MMRESULT mmrError, const char* func) { - static char buffer_err[TDAV_WAVEAPI_PRODUCER_ERROR_BUFF_COUNT]; + static char buffer_err[TDAV_WAVEAPI_PRODUCER_ERROR_BUFF_COUNT]; - waveInGetErrorTextA(mmrError, buffer_err, sizeof(buffer_err)); - TSK_DEBUG_ERROR("%s() error: %s", func, buffer_err); + waveInGetErrorTextA(mmrError, buffer_err, sizeof(buffer_err)); + TSK_DEBUG_ERROR("%s() error: %s", func, buffer_err); } static int free_wavehdr(tdav_producer_waveapi_t* producer, tsk_size_t index) { - if(!producer || index >= sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!producer || index >= sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - TSK_FREE(producer->hWaveHeaders[index]->lpData); - TSK_FREE(producer->hWaveHeaders[index]); + TSK_FREE(producer->hWaveHeaders[index]->lpData); + TSK_FREE(producer->hWaveHeaders[index]); - return 0; + return 0; } static int create_wavehdr(tdav_producer_waveapi_t* producer, tsk_size_t index) { - if(!producer || index >= sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR)){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(producer->hWaveHeaders[index]){ - free_wavehdr(producer, index); - } - - producer->hWaveHeaders[index] = tsk_calloc(1, sizeof(WAVEHDR)); - producer->hWaveHeaders[index]->lpData = tsk_calloc(1, producer->bytes_per_notif); - producer->hWaveHeaders[index]->dwBufferLength = (DWORD)producer->bytes_per_notif; - producer->hWaveHeaders[index]->dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP; - producer->hWaveHeaders[index]->dwLoops = 0x01; - producer->hWaveHeaders[index]->dwUser = index; - - return 0; + if(!producer || index >= sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR)) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(producer->hWaveHeaders[index]) { + free_wavehdr(producer, index); + } + + producer->hWaveHeaders[index] = tsk_calloc(1, sizeof(WAVEHDR)); + producer->hWaveHeaders[index]->lpData = tsk_calloc(1, producer->bytes_per_notif); + producer->hWaveHeaders[index]->dwBufferLength = (DWORD)producer->bytes_per_notif; + producer->hWaveHeaders[index]->dwFlags = WHDR_BEGINLOOP | WHDR_ENDLOOP; + producer->hWaveHeaders[index]->dwLoops = 0x01; + producer->hWaveHeaders[index]->dwUser = index; + + return 0; } static int add_wavehdr(tdav_producer_waveapi_t* producer, tsk_size_t index) { - MMRESULT result; - - if(!producer || !producer->hWaveHeaders[index] || !producer->hWaveIn){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - result = waveInPrepareHeader(producer->hWaveIn, producer->hWaveHeaders[index], sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInPrepareHeader"); - return -2; - } - - result = waveInAddBuffer(producer->hWaveIn, producer->hWaveHeaders[index], sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInAddBuffer"); - return -3; - } - - return 0; + MMRESULT result; + + if(!producer || !producer->hWaveHeaders[index] || !producer->hWaveIn) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + result = waveInPrepareHeader(producer->hWaveIn, producer->hWaveHeaders[index], sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInPrepareHeader"); + return -2; + } + + result = waveInAddBuffer(producer->hWaveIn, producer->hWaveHeaders[index], sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInAddBuffer"); + return -3; + } + + return 0; } static int record_wavehdr(tdav_producer_waveapi_t* producer, LPWAVEHDR lpHdr) { - MMRESULT result; + MMRESULT result; - if(!producer || !lpHdr || !producer->hWaveIn){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!producer || !lpHdr || !producer->hWaveIn) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - // - // Alert the session that there is new data to send over the network - // - if(TMEDIA_PRODUCER(producer)->enc_cb.callback){ + // + // Alert the session that there is new data to send over the network + // + if(TMEDIA_PRODUCER(producer)->enc_cb.callback) { #if 0 - { - static FILE* f = NULL; - if(!f) f = fopen("./waveapi_producer.raw", "w+"); - fwrite(lpHdr->lpData, 1, lpHdr->dwBytesRecorded, f); - } + { + static FILE* f = NULL; + if(!f) { + f = fopen("./waveapi_producer.raw", "w+"); + } + fwrite(lpHdr->lpData, 1, lpHdr->dwBytesRecorded, f); + } #endif - TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data, lpHdr->lpData, lpHdr->dwBytesRecorded); - } - - if(!producer->started){ - return 0; - } - - result = waveInUnprepareHeader(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInUnprepareHeader"); - return -2; - } - - result = waveInPrepareHeader(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInPrepareHeader"); - return -3; - } - - result = waveInAddBuffer(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInAddBuffer"); - return -4; - } - - return 0; + TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data, lpHdr->lpData, lpHdr->dwBytesRecorded); + } + + if(!producer->started) { + return 0; + } + + result = waveInUnprepareHeader(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInUnprepareHeader"); + return -2; + } + + result = waveInPrepareHeader(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInPrepareHeader"); + return -3; + } + + result = waveInAddBuffer(producer->hWaveIn, lpHdr, sizeof(WAVEHDR)); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInAddBuffer"); + return -4; + } + + return 0; } static void* TSK_STDCALL __record_thread(void *param) { - tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)param; - DWORD dwEvent; - tsk_size_t i; + tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)param; + DWORD dwEvent; + tsk_size_t i; - TSK_DEBUG_INFO("__record_thread -- START"); + TSK_DEBUG_INFO("__record_thread -- START"); - // SetPriorityClass(GetCurrentThread(), REALTIME_PRIORITY_CLASS); + // SetPriorityClass(GetCurrentThread(), REALTIME_PRIORITY_CLASS); - for(;;){ - dwEvent = WaitForMultipleObjects(2, producer->events, FALSE, INFINITE); + for(;;) { + dwEvent = WaitForMultipleObjects(2, producer->events, FALSE, INFINITE); - if (dwEvent == 1){ - break; - } + if (dwEvent == 1) { + break; + } - else if (dwEvent == 0){ - EnterCriticalSection(&producer->cs); - for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(producer->hWaveHeaders[0]); i++){ - if(producer->hWaveHeaders[i] && (producer->hWaveHeaders[i]->dwFlags & WHDR_DONE)){ - record_wavehdr(producer, producer->hWaveHeaders[i]); - } - } - LeaveCriticalSection(&producer->cs); - } - } + else if (dwEvent == 0) { + EnterCriticalSection(&producer->cs); + for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(producer->hWaveHeaders[0]); i++) { + if(producer->hWaveHeaders[i] && (producer->hWaveHeaders[i]->dwFlags & WHDR_DONE)) { + record_wavehdr(producer, producer->hWaveHeaders[i]); + } + } + LeaveCriticalSection(&producer->cs); + } + } - TSK_DEBUG_INFO("__record_thread() -- STOP"); - + TSK_DEBUG_INFO("__record_thread() -- STOP"); - return tsk_null; + + return tsk_null; } @@ -191,131 +193,131 @@ static void* TSK_STDCALL __record_thread(void *param) /* ============ Media Producer Interface ================= */ int tdav_producer_waveapi_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec) { - tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; - tsk_size_t i; - - if(!producer || !codec && codec->plugin){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); - TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); - /* codec should have ptime */ - - - /* Format */ - ZeroMemory(&producer->wfx, sizeof(WAVEFORMATEX)); - producer->wfx.wFormatTag = WAVE_FORMAT_PCM; - producer->wfx.nChannels = TMEDIA_PRODUCER(producer)->audio.channels; - producer->wfx.nSamplesPerSec = TMEDIA_PRODUCER(producer)->audio.rate; - producer->wfx.wBitsPerSample = TMEDIA_PRODUCER(producer)->audio.bits_per_sample; - producer->wfx.nBlockAlign = (producer->wfx.nChannels * producer->wfx.wBitsPerSample/8); - producer->wfx.nAvgBytesPerSec = (producer->wfx.nSamplesPerSec * producer->wfx.nBlockAlign); - - /* Average bytes (count) for each notification */ - producer->bytes_per_notif = ((producer->wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(producer)->audio.ptime)/1000); - - /* create buffers */ - for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(producer->hWaveHeaders[0]); i++){ - create_wavehdr(producer, i); - } - - return 0; + tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; + tsk_size_t i; + + if(!producer || !codec && codec->plugin) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + TMEDIA_PRODUCER(producer)->audio.channels = TMEDIA_CODEC_CHANNELS_AUDIO_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.rate = TMEDIA_CODEC_RATE_ENCODING(codec); + TMEDIA_PRODUCER(producer)->audio.ptime = TMEDIA_CODEC_PTIME_AUDIO_ENCODING(codec); + /* codec should have ptime */ + + + /* Format */ + ZeroMemory(&producer->wfx, sizeof(WAVEFORMATEX)); + producer->wfx.wFormatTag = WAVE_FORMAT_PCM; + producer->wfx.nChannels = TMEDIA_PRODUCER(producer)->audio.channels; + producer->wfx.nSamplesPerSec = TMEDIA_PRODUCER(producer)->audio.rate; + producer->wfx.wBitsPerSample = TMEDIA_PRODUCER(producer)->audio.bits_per_sample; + producer->wfx.nBlockAlign = (producer->wfx.nChannels * producer->wfx.wBitsPerSample/8); + producer->wfx.nAvgBytesPerSec = (producer->wfx.nSamplesPerSec * producer->wfx.nBlockAlign); + + /* Average bytes (count) for each notification */ + producer->bytes_per_notif = ((producer->wfx.nAvgBytesPerSec * TMEDIA_PRODUCER(producer)->audio.ptime)/1000); + + /* create buffers */ + for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(producer->hWaveHeaders[0]); i++) { + create_wavehdr(producer, i); + } + + return 0; } int tdav_producer_waveapi_start(tmedia_producer_t* self) { - tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; - MMRESULT result; - tsk_size_t i; - - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } - - if(producer->started || producer->hWaveIn){ - TSK_DEBUG_WARN("Producer already started"); - return 0; - } - - /* create events */ - if(!producer->events[0]){ - producer->events[0] = CreateEvent(NULL, FALSE, FALSE, NULL); - } - if(!producer->events[1]){ - producer->events[1] = CreateEvent(NULL, FALSE, FALSE, NULL); - } - - /* open */ - result = waveInOpen((HWAVEIN *)&producer->hWaveIn, /*WAVE_MAPPER*/0, &producer->wfx, (DWORD)producer->events[0], (DWORD_PTR)producer, CALLBACK_EVENT); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInOpen"); - return -2; - } - - /* start */ - result = waveInStart(producer->hWaveIn); - if(result != MMSYSERR_NOERROR){ - print_last_error(result, "waveInStart"); - return -2; - } - - /* write */ - for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR); i++){ - add_wavehdr(producer, i); - } - - /* start thread */ - producer->started = tsk_true; - tsk_thread_create(&producer->tid[0], __record_thread, producer); - - return 0; + tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; + MMRESULT result; + tsk_size_t i; + + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } + + if(producer->started || producer->hWaveIn) { + TSK_DEBUG_WARN("Producer already started"); + return 0; + } + + /* create events */ + if(!producer->events[0]) { + producer->events[0] = CreateEvent(NULL, FALSE, FALSE, NULL); + } + if(!producer->events[1]) { + producer->events[1] = CreateEvent(NULL, FALSE, FALSE, NULL); + } + + /* open */ + result = waveInOpen((HWAVEIN *)&producer->hWaveIn, /*WAVE_MAPPER*/0, &producer->wfx, (DWORD)producer->events[0], (DWORD_PTR)producer, CALLBACK_EVENT); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInOpen"); + return -2; + } + + /* start */ + result = waveInStart(producer->hWaveIn); + if(result != MMSYSERR_NOERROR) { + print_last_error(result, "waveInStart"); + return -2; + } + + /* write */ + for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR); i++) { + add_wavehdr(producer, i); + } + + /* start thread */ + producer->started = tsk_true; + tsk_thread_create(&producer->tid[0], __record_thread, producer); + + return 0; } int tdav_producer_waveapi_pause(tmedia_producer_t* self) { - tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; + tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; - if(!producer){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!producer) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - return 0; + return 0; } int tdav_producer_waveapi_stop(tmedia_producer_t* self) { - tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; - MMRESULT result; + tdav_producer_waveapi_t* producer = (tdav_producer_waveapi_t*)self; + MMRESULT result; - if(!self){ - TSK_DEBUG_ERROR("Invalid parameter"); - return -1; - } + if(!self) { + TSK_DEBUG_ERROR("Invalid parameter"); + return -1; + } - if(!producer->started){ - TSK_DEBUG_WARN("Producer not started"); - return 0; - } + if(!producer->started) { + TSK_DEBUG_WARN("Producer not started"); + return 0; + } - /* stop thread */ - if(producer->tid[0]){ - SetEvent(producer->events[1]); - tsk_thread_join(&(producer->tid[0])); - } + /* stop thread */ + if(producer->tid[0]) { + SetEvent(producer->events[1]); + tsk_thread_join(&(producer->tid[0])); + } - /* should be done here */ - producer->started = tsk_false; + /* should be done here */ + producer->started = tsk_false; - if(producer->hWaveIn && (((result = waveInReset(producer->hWaveIn)) != MMSYSERR_NOERROR) || ((result = waveInClose(producer->hWaveIn)) != MMSYSERR_NOERROR))){ - print_last_error(result, "waveInReset/waveInClose"); - } + if(producer->hWaveIn && (((result = waveInReset(producer->hWaveIn)) != MMSYSERR_NOERROR) || ((result = waveInClose(producer->hWaveIn)) != MMSYSERR_NOERROR))) { + print_last_error(result, "waveInReset/waveInClose"); + } - return 0; + return 0; } @@ -325,68 +327,66 @@ int tdav_producer_waveapi_stop(tmedia_producer_t* self) /* constructor */ static tsk_object_t* tdav_producer_waveapi_ctor(tsk_object_t * self, va_list * app) { - tdav_producer_waveapi_t *producer = self; - if(producer){ - /* init base */ - tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); - /* init self */ - InitializeCriticalSection(&producer->cs); - } - return self; + tdav_producer_waveapi_t *producer = self; + if(producer) { + /* init base */ + tdav_producer_audio_init(TDAV_PRODUCER_AUDIO(producer)); + /* init self */ + InitializeCriticalSection(&producer->cs); + } + return self; } /* destructor */ static tsk_object_t* tdav_producer_waveapi_dtor(tsk_object_t * self) -{ - tdav_producer_waveapi_t *producer = self; - if(producer){ - tsk_size_t i; - - /* stop */ - if(producer->started){ - tdav_producer_waveapi_stop(self); - } - - /* deinit base */ - tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); - /* deinit self */ - for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR); i++){ - free_wavehdr(producer, i); - } - if(producer->hWaveIn){ - waveInClose(producer->hWaveIn); - } - if(producer->events[0]){ - CloseHandle(producer->events[0]); - } - if(producer->events[1]){ - CloseHandle(producer->events[1]); - } - DeleteCriticalSection(&producer->cs); - } - - return self; +{ + tdav_producer_waveapi_t *producer = self; + if(producer) { + tsk_size_t i; + + /* stop */ + if(producer->started) { + tdav_producer_waveapi_stop(self); + } + + /* deinit base */ + tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer)); + /* deinit self */ + for(i = 0; i< sizeof(producer->hWaveHeaders)/sizeof(LPWAVEHDR); i++) { + free_wavehdr(producer, i); + } + if(producer->hWaveIn) { + waveInClose(producer->hWaveIn); + } + if(producer->events[0]) { + CloseHandle(producer->events[0]); + } + if(producer->events[1]) { + CloseHandle(producer->events[1]); + } + DeleteCriticalSection(&producer->cs); + } + + return self; } /* object definition */ -static const tsk_object_def_t tdav_producer_waveapi_def_s = -{ - sizeof(tdav_producer_waveapi_t), - tdav_producer_waveapi_ctor, - tdav_producer_waveapi_dtor, - tdav_producer_audio_cmp, +static const tsk_object_def_t tdav_producer_waveapi_def_s = { + sizeof(tdav_producer_waveapi_t), + tdav_producer_waveapi_ctor, + tdav_producer_waveapi_dtor, + tdav_producer_audio_cmp, }; /* plugin definition*/ -static const tmedia_producer_plugin_def_t tdav_producer_waveapi_plugin_def_s = -{ - &tdav_producer_waveapi_def_s, - - tmedia_audio, - "Microsoft WaveAPI producer", - - tdav_producer_waveapi_set, - tdav_producer_waveapi_prepare, - tdav_producer_waveapi_start, - tdav_producer_waveapi_pause, - tdav_producer_waveapi_stop +static const tmedia_producer_plugin_def_t tdav_producer_waveapi_plugin_def_s = { + &tdav_producer_waveapi_def_s, + + tmedia_audio, + "Microsoft WaveAPI producer", + + tdav_producer_waveapi_set, + tdav_producer_waveapi_prepare, + tdav_producer_waveapi_start, + tdav_producer_waveapi_pause, + tdav_producer_waveapi_stop }; const tmedia_producer_plugin_def_t *tdav_producer_waveapi_plugin_def_t = &tdav_producer_waveapi_plugin_def_s; |
