obs-ffmpeg: Enable multiple audio tracks for FFmpeg output
This allows multiple audio tracks for the FFmpeg output. Closes obsproject/obs-studio#1351
This commit is contained in:
pkviet
jp9000
parent
b8a3ae1b10
commit
340fb9d669
@ -45,6 +45,8 @@ struct ffmpeg_cfg {
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int audio_encoder_id;
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const char *video_settings;
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const char *audio_settings;
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int audio_mix_count;
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int audio_tracks;
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enum AVPixelFormat format;
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enum AVColorRange color_range;
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enum AVColorSpace color_space;
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@ -56,7 +58,7 @@ struct ffmpeg_cfg {
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struct ffmpeg_data {
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AVStream *video;
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AVStream *audio;
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AVStream **audio_streams;
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AVCodec *acodec;
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AVCodec *vcodec;
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AVFormatContext *output;
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@ -68,14 +70,18 @@ struct ffmpeg_data {
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uint64_t start_timestamp;
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int64_t total_samples;
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int64_t total_samples[MAX_AUDIO_MIXES];
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uint32_t audio_samplerate;
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enum audio_format audio_format;
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size_t audio_planes;
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size_t audio_size;
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struct circlebuf excess_frames[MAX_AV_PLANES];
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uint8_t *samples[MAX_AV_PLANES];
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AVFrame *aframe;
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int num_audio_streams;
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/* audio_tracks is a bitmask storing the indices of the mixes */
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int audio_tracks;
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struct circlebuf excess_frames[MAX_AUDIO_MIXES][MAX_AV_PLANES];
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uint8_t *samples[MAX_AUDIO_MIXES][MAX_AV_PLANES];
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AVFrame *aframe[MAX_AUDIO_MIXES];
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struct ffmpeg_cfg config;
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@ -273,9 +279,9 @@ static bool create_video_stream(struct ffmpeg_data *data)
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return true;
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}
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static bool open_audio_codec(struct ffmpeg_data *data)
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static bool open_audio_codec(struct ffmpeg_data *data, int idx)
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{
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AVCodecContext *context = data->audio->codec;
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AVCodecContext *context = data->audio_streams[idx]->codec;
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char **opts = strlist_split(data->config.audio_settings, ' ', false);
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int ret;
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@ -284,16 +290,16 @@ static bool open_audio_codec(struct ffmpeg_data *data)
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strlist_free(opts);
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}
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data->aframe = av_frame_alloc();
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if (!data->aframe) {
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data->aframe[idx] = av_frame_alloc();
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if (!data->aframe[idx]) {
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blog(LOG_WARNING, "Failed to allocate audio frame");
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return false;
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}
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data->aframe->format = context->sample_fmt;
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data->aframe->channels = context->channels;
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data->aframe->channel_layout = context->channel_layout;
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data->aframe->sample_rate = context->sample_rate;
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data->aframe[idx]->format = context->sample_fmt;
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data->aframe[idx]->channels = context->channels;
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data->aframe[idx]->channel_layout = context->channel_layout;
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data->aframe[idx]->sample_rate = context->sample_rate;
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context->strict_std_compliance = -2;
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@ -306,7 +312,7 @@ static bool open_audio_codec(struct ffmpeg_data *data)
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data->frame_size = context->frame_size ? context->frame_size : 1024;
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ret = av_samples_alloc(data->samples, NULL, context->channels,
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ret = av_samples_alloc(data->samples[idx], NULL, context->channels,
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data->frame_size, context->sample_fmt, 0);
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if (ret < 0) {
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blog(LOG_WARNING, "Failed to create audio buffer: %s",
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@ -317,9 +323,10 @@ static bool open_audio_codec(struct ffmpeg_data *data)
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return true;
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}
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static bool create_audio_stream(struct ffmpeg_data *data)
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static bool create_audio_stream(struct ffmpeg_data *data, int idx)
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{
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AVCodecContext *context;
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AVStream *stream;
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struct obs_audio_info aoi;
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if (!obs_get_audio_info(&aoi)) {
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@ -327,17 +334,18 @@ static bool create_audio_stream(struct ffmpeg_data *data)
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return false;
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}
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if (!new_stream(data, &data->audio, &data->acodec,
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if (!new_stream(data, &stream, &data->acodec,
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data->output->oformat->audio_codec,
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data->config.audio_encoder))
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return false;
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context = data->audio->codec;
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context->bit_rate = data->config.audio_bitrate * 1000;
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context->time_base = (AVRational){ 1, aoi.samples_per_sec };
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context->channels = get_audio_channels(aoi.speakers);
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context->sample_rate = aoi.samples_per_sec;
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context->channel_layout =
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data->audio_streams[idx] = stream;
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context = data->audio_streams[idx]->codec;
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context->bit_rate = data->config.audio_bitrate * 1000;
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context->time_base = (AVRational){ 1, aoi.samples_per_sec };
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context->channels = get_audio_channels(aoi.speakers);
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context->sample_rate = aoi.samples_per_sec;
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context->channel_layout =
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av_get_default_channel_layout(context->channels);
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//AVlib default channel layout for 5 channels is 5.0 ; fix for 4.1
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@ -347,7 +355,7 @@ static bool create_audio_stream(struct ffmpeg_data *data)
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context->sample_fmt = data->acodec->sample_fmts ?
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data->acodec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
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data->audio->time_base = context->time_base;
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data->audio_streams[idx]->time_base = context->time_base;
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data->audio_samplerate = aoi.samples_per_sec;
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data->audio_format = convert_ffmpeg_sample_format(context->sample_fmt);
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@ -357,7 +365,7 @@ static bool create_audio_stream(struct ffmpeg_data *data)
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if (data->output->oformat->flags & AVFMT_GLOBALHEADER)
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context->flags |= CODEC_FLAG_GLOBAL_H;
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return open_audio_codec(data);
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return open_audio_codec(data, idx);
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}
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static inline bool init_streams(struct ffmpeg_data *data)
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@ -368,9 +376,14 @@ static inline bool init_streams(struct ffmpeg_data *data)
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if (!create_video_stream(data))
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return false;
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if (format->audio_codec != AV_CODEC_ID_NONE)
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if (!create_audio_stream(data))
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return false;
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if (format->audio_codec != AV_CODEC_ID_NONE && data->num_audio_streams) {
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data->audio_streams = calloc(1,
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data->num_audio_streams * sizeof(void*));
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for (int i = 0; i < data->num_audio_streams; i++) {
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if (!create_audio_stream(data, i))
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return false;
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}
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}
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return true;
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}
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@ -457,12 +470,14 @@ static void close_video(struct ffmpeg_data *data)
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static void close_audio(struct ffmpeg_data *data)
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{
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for (size_t i = 0; i < MAX_AV_PLANES; i++)
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circlebuf_free(&data->excess_frames[i]);
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for (int idx = 0; idx < data->num_audio_streams; idx++) {
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for (size_t i = 0; i < MAX_AV_PLANES; i++)
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circlebuf_free(&data->excess_frames[idx][i]);
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av_freep(&data->samples[0]);
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avcodec_close(data->audio->codec);
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av_frame_free(&data->aframe);
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av_freep(&data->samples[idx][0]);
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avcodec_close(data->audio_streams[idx]->codec);
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av_frame_free(&data->aframe[idx]);
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}
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}
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static void ffmpeg_data_free(struct ffmpeg_data *data)
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@ -472,8 +487,11 @@ static void ffmpeg_data_free(struct ffmpeg_data *data)
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if (data->video)
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close_video(data);
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if (data->audio)
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if (data->audio_streams) {
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close_audio(data);
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free(data->audio_streams);
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data->audio_streams = NULL;
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}
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if (data->output) {
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if ((data->output->oformat->flags & AVFMT_NOFILE) == 0)
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@ -528,7 +546,8 @@ static bool ffmpeg_data_init(struct ffmpeg_data *data,
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memset(data, 0, sizeof(struct ffmpeg_data));
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data->config = *config;
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data->num_audio_streams = config->audio_mix_count;
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data->audio_tracks = config->audio_tracks;
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if (!config->url || !*config->url)
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return false;
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@ -766,7 +785,7 @@ static void receive_video(void *param, struct video_data *frame)
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data->total_frames++;
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}
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static void encode_audio(struct ffmpeg_output *output,
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static void encode_audio(struct ffmpeg_output *output, int idx,
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struct AVCodecContext *context, size_t block_size)
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{
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struct ffmpeg_data *data = &output->ff_data;
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@ -775,13 +794,13 @@ static void encode_audio(struct ffmpeg_output *output,
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int ret, got_packet;
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size_t total_size = data->frame_size * block_size * context->channels;
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data->aframe->nb_samples = data->frame_size;
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data->aframe->pts = av_rescale_q(data->total_samples,
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data->aframe[idx]->nb_samples = data->frame_size;
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data->aframe[idx]->pts = av_rescale_q(data->total_samples[idx],
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(AVRational){1, context->sample_rate},
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context->time_base);
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ret = avcodec_fill_audio_frame(data->aframe, context->channels,
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context->sample_fmt, data->samples[0],
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ret = avcodec_fill_audio_frame(data->aframe[idx], context->channels,
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context->sample_fmt, data->samples[idx][0],
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(int)total_size, 1);
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if (ret < 0) {
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blog(LOG_WARNING, "encode_audio: avcodec_fill_audio_frame "
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@ -789,10 +808,10 @@ static void encode_audio(struct ffmpeg_output *output,
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return;
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}
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data->total_samples += data->frame_size;
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data->total_samples[idx] += data->frame_size;
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#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(57, 40, 101)
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ret = avcodec_send_frame(context, data->aframe);
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ret = avcodec_send_frame(context, data->aframe[idx]);
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if (ret == 0)
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ret = avcodec_receive_packet(context, &packet);
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@ -801,7 +820,7 @@ static void encode_audio(struct ffmpeg_output *output,
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if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
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ret = 0;
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#else
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ret = avcodec_encode_audio2(context, &packet, data->aframe,
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ret = avcodec_encode_audio2(context, &packet, data->aframe[idx],
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&got_packet);
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#endif
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if (ret < 0) {
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@ -813,11 +832,13 @@ static void encode_audio(struct ffmpeg_output *output,
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if (!got_packet)
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return;
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packet.pts = rescale_ts(packet.pts, context, data->audio->time_base);
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packet.dts = rescale_ts(packet.dts, context, data->audio->time_base);
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packet.pts = rescale_ts(packet.pts, context,
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data->audio_streams[idx]->time_base);
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packet.dts = rescale_ts(packet.dts, context,
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data->audio_streams[idx]->time_base);
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packet.duration = (int)av_rescale_q(packet.duration, context->time_base,
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data->audio->time_base);
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packet.stream_index = data->audio->index;
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data->audio_streams[idx]->time_base);
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packet.stream_index = data->audio_streams[idx]->index;
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pthread_mutex_lock(&output->write_mutex);
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da_push_back(output->packets, &packet);
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@ -853,18 +874,34 @@ static bool prepare_audio(struct ffmpeg_data *data,
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return true;
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}
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static void receive_audio(void *param, struct audio_data *frame)
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/* Given a bitmask for the selected tracks and the mix index,
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* this returns the stream index which will be passed to the muxer. */
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static int get_track_order(int track_config, size_t mix_index)
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{
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int position = 0;
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for (size_t i = 0; i < mix_index; i++) {
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if (track_config & 1 << i)
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position++;
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}
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return position;
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}
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static void receive_audio(void *param, size_t mix_idx, struct audio_data *frame)
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{
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struct ffmpeg_output *output = param;
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struct ffmpeg_data *data = &output->ff_data;
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size_t frame_size_bytes;
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struct audio_data in;
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int track_order;
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// codec doesn't support audio or none configured
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if (!data->audio)
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/* check that the track was selected */
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if ((data->audio_tracks & (1 << mix_idx)) == 0)
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return;
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AVCodecContext *context = data->audio->codec;
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/* get track order (first selected, etc ...) */
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track_order = get_track_order(data->audio_tracks, mix_idx);
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AVCodecContext *context = data->audio_streams[track_order]->codec;
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if (!data->start_timestamp)
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return;
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@ -877,15 +914,16 @@ static void receive_audio(void *param, struct audio_data *frame)
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frame_size_bytes = (size_t)data->frame_size * data->audio_size;
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for (size_t i = 0; i < data->audio_planes; i++)
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circlebuf_push_back(&data->excess_frames[i], in.data[i],
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in.frames * data->audio_size);
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circlebuf_push_back(&data->excess_frames[track_order][i],
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in.data[i], in.frames * data->audio_size);
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while (data->excess_frames[0].size >= frame_size_bytes) {
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while (data->excess_frames[track_order][0].size >= frame_size_bytes) {
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for (size_t i = 0; i < data->audio_planes; i++)
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circlebuf_pop_front(&data->excess_frames[i],
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data->samples[i], frame_size_bytes);
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circlebuf_pop_front(&data->excess_frames[track_order][i],
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data->samples[track_order][i],
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frame_size_bytes);
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encode_audio(output, context, data->audio_size);
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encode_audio(output, track_order, context, data->audio_size);
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}
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}
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@ -901,7 +939,7 @@ static uint64_t get_packet_sys_dts(struct ffmpeg_output *output,
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time_base = data->video->time_base;
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start_ts = output->video_start_ts;
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} else {
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time_base = data->audio->time_base;
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time_base = data->audio_streams[0]->time_base;
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start_ts = output->audio_start_ts;
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}
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@ -990,6 +1028,18 @@ static inline const char *get_string_or_null(obs_data_t *settings,
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return value;
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}
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static int get_audio_mix_count(int audio_mix_mask)
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{
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int mix_count = 0;
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for (int i = 0; i < MAX_AUDIO_MIXES; i++) {
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if ((audio_mix_mask & (1 << i)) != 0) {
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mix_count++;
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}
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}
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return mix_count;
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}
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static bool try_connect(struct ffmpeg_output *output)
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{
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video_t *video = obs_output_video(output->output);
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@ -1025,6 +1075,8 @@ static bool try_connect(struct ffmpeg_output *output)
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config.height = (int)obs_output_get_height(output->output);
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config.format = obs_to_ffmpeg_video_format(
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video_output_get_format(video));
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config.audio_tracks = (int)obs_output_get_mixers(output->output);
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config.audio_mix_count = get_audio_mix_count(config.audio_tracks);
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if (format_is_yuv(voi->format)) {
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config.color_range = voi->range == VIDEO_RANGE_FULL ?
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@ -1157,13 +1209,15 @@ static uint64_t ffmpeg_output_total_bytes(void *data)
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struct obs_output_info ffmpeg_output = {
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.id = "ffmpeg_output",
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.flags = OBS_OUTPUT_AUDIO | OBS_OUTPUT_VIDEO,
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.flags = OBS_OUTPUT_AUDIO |
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OBS_OUTPUT_VIDEO |
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OBS_OUTPUT_MULTI_TRACK,
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.get_name = ffmpeg_output_getname,
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.create = ffmpeg_output_create,
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.destroy = ffmpeg_output_destroy,
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.start = ffmpeg_output_start,
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.stop = ffmpeg_output_stop,
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.raw_video = receive_video,
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.raw_audio = receive_audio,
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.raw_audio2 = receive_audio,
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.get_total_bytes = ffmpeg_output_total_bytes,
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};
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