diff options
Diffstat (limited to 'libavcodec/libopusdec.c')
-rw-r--r-- | libavcodec/libopusdec.c | 127 |
1 files changed, 103 insertions, 24 deletions
diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c index 1dac1a0..3d2ee5b 100644 --- a/libavcodec/libopusdec.c +++ b/libavcodec/libopusdec.c @@ -2,27 +2,30 @@ * Opus decoder using libopus * Copyright (c) 2012 Nicolas George * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <opus.h> #include <opus_multistream.h> +#include "libavutil/internal.h" #include "libavutil/intreadwrite.h" +#include "libavutil/ffmath.h" +#include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" @@ -31,7 +34,15 @@ #include "libopus.h" struct libopus_context { + AVClass *class; OpusMSDecoder *dec; + int pre_skip; +#ifndef OPUS_SET_GAIN + union { int i; double d; } gain; +#endif +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + int apply_phase_inv; +#endif }; #define OPUS_HEAD_SIZE 19 @@ -42,12 +53,6 @@ static av_cold int libopus_decode_init(AVCodecContext *avc) int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled; uint8_t mapping_arr[8] = { 0, 1 }, *mapping; - if (avc->channels <= 0) { - av_log(avc, AV_LOG_WARNING, - "Invalid number of channels %d, defaulting to stereo\n", avc->channels); - avc->channels = 2; - } - avc->channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->channels == 1) ? 1 : 2; if (avc->channels <= 0) { av_log(avc, AV_LOG_WARNING, @@ -58,10 +63,9 @@ static av_cold int libopus_decode_init(AVCodecContext *avc) avc->sample_rate = 48000; avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ? AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16; - avc->channel_layout = avc->channels > 8 ? 0 : - ff_vorbis_channel_layouts[avc->channels - 1]; if (avc->extradata_size >= OPUS_HEAD_SIZE) { + opus->pre_skip = AV_RL16(avc->extradata + 10); gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16); channel_map = AV_RL8 (avc->extradata + 18); } @@ -82,14 +86,35 @@ static av_cold int libopus_decode_init(AVCodecContext *avc) mapping = mapping_arr; } - if (avc->channels > 2 && avc->channels <= 8) { - const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1]; - int ch; + if (channel_map == 1) { + avc->channel_layout = avc->channels > 8 ? 0 : + ff_vorbis_channel_layouts[avc->channels - 1]; + if (avc->channels > 2 && avc->channels <= 8) { + const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1]; + int ch; - /* Remap channels from Vorbis order to libav order */ - for (ch = 0; ch < avc->channels; ch++) - mapping_arr[ch] = mapping[vorbis_offset[ch]]; - mapping = mapping_arr; + /* Remap channels from Vorbis order to ffmpeg order */ + for (ch = 0; ch < avc->channels; ch++) + mapping_arr[ch] = mapping[vorbis_offset[ch]]; + mapping = mapping_arr; + } + } else if (channel_map == 2) { + int ambisonic_order = ff_sqrt(avc->channels) - 1; + if (avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) && + avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) + 2) { + av_log(avc, AV_LOG_ERROR, + "Channel mapping 2 is only specified for channel counts" + " which can be written as (n + 1)^2 or (n + 2)^2 + 2" + " for nonnegative integer n\n"); + return AVERROR_INVALIDDATA; + } + if (avc->channels > 227) { + av_log(avc, AV_LOG_ERROR, "Too many channels\n"); + return AVERROR_INVALIDDATA; + } + avc->channel_layout = 0; + } else { + avc->channel_layout = 0; } opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels, @@ -101,12 +126,32 @@ static av_cold int libopus_decode_init(AVCodecContext *avc) return ff_opus_error_to_averror(ret); } +#ifdef OPUS_SET_GAIN ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db)); if (ret != OPUS_OK) av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n", opus_strerror(ret)); +#else + { + double gain_lin = ff_exp10(gain_db / (20.0 * 256)); + if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) + opus->gain.d = gain_lin; + else + opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX); + } +#endif + +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + ret = opus_multistream_decoder_ctl(opus->dec, + OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv)); + if (ret != OPUS_OK) + av_log(avc, AV_LOG_WARNING, + "Unable to set phase inversion: %s\n", + opus_strerror(ret)); +#endif - avc->delay = 3840; /* Decoder delay (in samples) at 48kHz */ + /* Decoder delay (in samples) at 48kHz */ + avc->delay = avc->internal->skip_samples = opus->pre_skip; return 0; } @@ -129,11 +174,8 @@ static int libopus_decode(AVCodecContext *avc, void *data, int ret, nb_samples; frame->nb_samples = MAX_FRAME_SIZE; - ret = ff_get_buffer(avc, frame, 0); - if (ret < 0) { - av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n"); + if ((ret = ff_get_buffer(avc, frame, 0)) < 0) return ret; - } if (avc->sample_fmt == AV_SAMPLE_FMT_S16) nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size, @@ -150,6 +192,21 @@ static int libopus_decode(AVCodecContext *avc, void *data, return ff_opus_error_to_averror(nb_samples); } +#ifndef OPUS_SET_GAIN + { + int i = avc->channels * nb_samples; + if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) { + float *pcm = (float *)frame->data[0]; + for (; i > 0; i--, pcm++) + *pcm = av_clipf(*pcm * opus->gain.d, -1, 1); + } else { + int16_t *pcm = (int16_t *)frame->data[0]; + for (; i > 0; i--, pcm++) + *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16); + } + } +#endif + frame->nb_samples = nb_samples; *got_frame_ptr = 1; @@ -161,8 +218,29 @@ static void libopus_flush(AVCodecContext *avc) struct libopus_context *opus = avc->priv_data; opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE); + /* The stream can have been extracted by a tool that is not Opus-aware. + Therefore, any packet can become the first of the stream. */ + avc->internal->skip_samples = opus->pre_skip; } + +#define OFFSET(x) offsetof(struct libopus_context, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM +static const AVOption libopusdec_options[] = { +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS }, +#endif + { NULL }, +}; + +static const AVClass libopusdec_class = { + .class_name = "libopusdec", + .item_name = av_default_item_name, + .option = libopusdec_options, + .version = LIBAVUTIL_VERSION_INT, +}; + + AVCodec ff_libopus_decoder = { .name = "libopus", .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"), @@ -177,5 +255,6 @@ AVCodec ff_libopus_decoder = { .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, + .priv_class = &libopusdec_class, .wrapper_name = "libopus", }; |