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FFmpeg: Use refcounted buffers for frame data

This commit is contained in:
Vicki Pfau 2021-08-21 18:27:06 -07:00
parent 94124db16c
commit d39e521472
1 changed files with 5 additions and 7 deletions
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12
src/feature/ffmpeg/ffmpeg-encoder.c
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@ -322,6 +322,7 @@ bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
encoder->audioFrame->nb_samples = encoder->audio->frame_size;
encoder->audioFrame->format = encoder->audio->sample_fmt;
encoder->audioFrame->pts = 0;
encoder->audioFrame->channel_layout = AV_CH_LAYOUT_STEREO;
#ifdef USE_LIBAVRESAMPLE
encoder->resampleContext = avresample_alloc_context();
av_opt_set_int(encoder->resampleContext, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
@ -338,9 +339,7 @@ bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
#endif
encoder->audioBufferSize = (encoder->audioFrame->nb_samples * PREFERRED_SAMPLE_RATE / encoder->sampleRate) * 4;
encoder->audioBuffer = av_malloc(encoder->audioBufferSize);
encoder->postaudioBufferSize = av_samples_get_buffer_size(0, encoder->audio->channels, encoder->audio->frame_size, encoder->audio->sample_fmt, 0);
encoder->postaudioBuffer = av_malloc(encoder->postaudioBufferSize);
avcodec_fill_audio_frame(encoder->audioFrame, encoder->audio->channels, encoder->audio->sample_fmt, (const uint8_t*) encoder->postaudioBuffer, encoder->postaudioBufferSize, 0);
av_frame_get_buffer(encoder->audioFrame, 0);
if (encoder->audio->codec->id == AV_CODEC_ID_AAC &&
(strcasecmp(encoder->containerFormat, "mp4") == 0||
@ -517,7 +516,7 @@ bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
encoder->videoFrame->height = encoder->video->height;
encoder->videoFrame->pts = 0;
_ffmpegSetVideoDimensions(&encoder->d, encoder->iwidth, encoder->iheight);
av_image_alloc(encoder->videoFrame->data, encoder->videoFrame->linesize, encoder->videoFrame->width, encoder->videoFrame->height, encoder->videoFrame->format, 32);
av_frame_get_buffer(encoder->videoFrame, 32);
#ifdef FFMPEG_USE_CODECPAR
avcodec_parameters_from_context(encoder->videoStream->codecpar, encoder->video);
#endif
@ -692,7 +691,6 @@ void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right
return;
}
int channelSize = 2 * av_get_bytes_per_sample(encoder->audio->sample_fmt);
encoder->currentAudioSample = 0;
#ifdef USE_LIBAVRESAMPLE
avresample_convert(encoder->resampleContext, 0, 0, 0,
@ -704,7 +702,7 @@ void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_make_writable(encoder->audioFrame);
#endif
int samples = avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize);
int samples = avresample_read(encoder->resampleContext, encoder->audioFrame->data, encoder->audioFrame->nb_samples);
#else
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_make_writable(encoder->audioFrame);
@ -713,7 +711,7 @@ void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right
swr_convert(encoder->resampleContext, NULL, 0, (const uint8_t**) &encoder->audioBuffer, encoder->audioBufferSize / 4);
return;
}
int samples = swr_convert(encoder->resampleContext, encoder->audioFrame->data, encoder->postaudioBufferSize / channelSize,
int samples = swr_convert(encoder->resampleContext, encoder->audioFrame->data, encoder->audioFrame->nb_samples,
(const uint8_t**) &encoder->audioBuffer, encoder->audioBufferSize / 4);
#endif