Method open_audio

src/FFmpegWriter.cpp:1340–1412

open audio codec

// open audio codec
void FFmpegWriter::open_audio(AVFormatContext *oc, AVStream *st) {
	const AVCodec *codec;
	AV_GET_CODEC_FROM_STREAM(st, audio_codec_ctx)

	// Audio encoding does not typically use more than 2 threads (most codecs use 1 thread)
	audio_codec_ctx->thread_count = std::min(FF_AUDIO_NUM_PROCESSORS, 2);

	// Find the audio encoder
	codec = avcodec_find_encoder_by_name(info.acodec.c_str());
	if (!codec)
		codec = avcodec_find_encoder(audio_codec_ctx->codec_id);
	if (!codec)
		throw InvalidCodec("Could not find codec", path);

	// Init options
	AVDictionary *opts = NULL;
	av_dict_set(&opts, "strict", "experimental", 0);

	// Open the codec
	if (avcodec_open2(audio_codec_ctx, codec, &opts) < 0)
		throw InvalidCodec("Could not open audio codec", path);
	AV_COPY_PARAMS_FROM_CONTEXT(st, audio_codec_ctx);

	// Free options
	av_dict_free(&opts);

	// Calculate the size of the input frame (i..e how many samples per packet), and the output buffer
	// TODO: Ugly hack for PCM codecs (will be removed ASAP with new PCM support to compute the input frame size in samples
	if (audio_codec_ctx->frame_size <= 1) {
		// No frame size found... so calculate
		audio_input_frame_size = 50000 / info.channels;

		int s = AV_FIND_DECODER_CODEC_ID(st);
		switch (s) {
			case AV_CODEC_ID_PCM_S16LE:
			case AV_CODEC_ID_PCM_S16BE:
			case AV_CODEC_ID_PCM_U16LE:
			case AV_CODEC_ID_PCM_U16BE:
				audio_input_frame_size >>= 1;
				break;
			default:
				break;
		}
	} else {
		// Set frame size based on the codec
		audio_input_frame_size = audio_codec_ctx->frame_size;
	}

	// Set the initial frame size (since it might change during resampling)
	initial_audio_input_frame_size = audio_input_frame_size;

	// Allocate array for samples
	samples = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE];

	// Set audio output buffer (used to store the encoded audio)
	audio_outbuf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
	audio_outbuf = new uint8_t[audio_outbuf_size];