diff options
Diffstat (limited to 'libavdevice/alsa-audio-enc.c')
-rw-r--r-- | libavdevice/alsa-audio-enc.c | 69 |
1 files changed, 63 insertions, 6 deletions
diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c index bb4575f..43d097d 100644 --- a/libavdevice/alsa-audio-enc.c +++ b/libavdevice/alsa-audio-enc.c @@ -3,20 +3,20 @@ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) * - * 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 */ @@ -38,19 +38,26 @@ */ #include <alsa/asoundlib.h> -#include "libavformat/avformat.h" +#include "libavutil/time.h" +#include "libavformat/internal.h" +#include "avdevice.h" #include "alsa-audio.h" static av_cold int audio_write_header(AVFormatContext *s1) { AlsaData *s = s1->priv_data; - AVStream *st; + AVStream *st = NULL; unsigned int sample_rate; enum AVCodecID codec_id; int res; + if (s1->nb_streams != 1 || s1->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) { + av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n"); + return AVERROR(EINVAL); + } st = s1->streams[0]; + sample_rate = st->codec->sample_rate; codec_id = st->codec->codec_id; res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, @@ -61,6 +68,7 @@ static av_cold int audio_write_header(AVFormatContext *s1) st->codec->sample_rate, sample_rate); goto fail; } + avpriv_set_pts_info(st, 64, 1, sample_rate); return res; @@ -77,6 +85,10 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) uint8_t *buf = pkt->data; size /= s->frame_size; + if (pkt->dts != AV_NOPTS_VALUE) + s->timestamp = pkt->dts; + s->timestamp += pkt->duration ? pkt->duration : size; + if (s->reorder_func) { if (size > s->reorder_buf_size) if (ff_alsa_extend_reorder_buf(s, size)) @@ -101,6 +113,47 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) return 0; } +static int audio_write_frame(AVFormatContext *s1, int stream_index, + AVFrame **frame, unsigned flags) +{ + AlsaData *s = s1->priv_data; + AVPacket pkt; + + /* ff_alsa_open() should have accepted only supported formats */ + if ((flags & AV_WRITE_UNCODED_FRAME_QUERY)) + return av_sample_fmt_is_planar(s1->streams[stream_index]->codec->sample_fmt) ? + AVERROR(EINVAL) : 0; + /* set only used fields */ + pkt.data = (*frame)->data[0]; + pkt.size = (*frame)->nb_samples * s->frame_size; + pkt.dts = (*frame)->pkt_dts; + pkt.duration = av_frame_get_pkt_duration(*frame); + return audio_write_packet(s1, &pkt); +} + +static void +audio_get_output_timestamp(AVFormatContext *s1, int stream, + int64_t *dts, int64_t *wall) +{ + AlsaData *s = s1->priv_data; + snd_pcm_sframes_t delay = 0; + *wall = av_gettime(); + snd_pcm_delay(s->h, &delay); + *dts = s->timestamp - delay; +} + +static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) +{ + return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK); +} + +static const AVClass alsa_muxer_class = { + .class_name = "ALSA muxer", + .item_name = av_default_item_name, + .version = LIBAVUTIL_VERSION_INT, + .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, +}; + AVOutputFormat ff_alsa_muxer = { .name = "alsa", .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), @@ -110,5 +163,9 @@ AVOutputFormat ff_alsa_muxer = { .write_header = audio_write_header, .write_packet = audio_write_packet, .write_trailer = ff_alsa_close, + .write_uncoded_frame = audio_write_frame, + .get_device_list = audio_get_device_list, + .get_output_timestamp = audio_get_output_timestamp, .flags = AVFMT_NOFILE, + .priv_class = &alsa_muxer_class, }; |