From d39e521472b715a73a2bac96c457b2375aa2e999 Mon Sep 17 00:00:00 2001 From: Vicki Pfau Date: Sat, 21 Aug 2021 18:27:06 -0700 Subject: [PATCH] FFmpeg: Use refcounted buffers for frame data --- src/feature/ffmpeg/ffmpeg-encoder.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/src/feature/ffmpeg/ffmpeg-encoder.c b/src/feature/ffmpeg/ffmpeg-encoder.c index 17cf11e6c..6ee405840 100644 --- a/src/feature/ffmpeg/ffmpeg-encoder.c +++ b/src/feature/ffmpeg/ffmpeg-encoder.c @@ -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