音频MP2编码和解码可产生一半数据C ++和FFMPEG

Question

I have 16 bit PCM data with stereo setup which I grab from a microphone.

Once I get the data I encode it using the following encoder settings

AVCodec* audio_codec = avcodec_find_encoder(AV_CODEC_ID_MP2);

AVCodecContext* audio_codec_ctx = avcodec_alloc_context3(audio_codec);
audio_codec_ctx->bit_rate = 64000;
audio_codec_ctx->channels = 2;
audio_codec_ctx->channel_layout = AV_CH_LAYOUT_STEREO;
audio_codec_ctx->sample_rate = 44100;
audio_codec_ctx->sample_fmt = AV_SAMPLE_FMT_S16;

When I pass the audio data into the encoder I see that it takes 4608 bytes of data each time and encodes it correctly to MP2 data. PCM data grabbed by the microphone is 88320 bytes and the encoder takes 4608 bytes each time and compresses it.

If I take each 4608 byte section that has been encoded and pass it through a decoder with the same settings as above but with a decoder.

AVCodecID audio_codec_id = AV_CODEC_ID_MP2;
AVCodec * audio_decodec = avcodec_find_decoder(audio_codec_id);

audio_decodecContext = avcodec_alloc_context3(audio_decodec); 
audio_decodecContext->bit_rate = 64000;
audio_decodecContext->channels = 2;
audio_decodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
audio_decodecContext->sample_rate = 44100;
audio_decodecContext->sample_fmt = AV_SAMPLE_FMT_S16;

The decoding works and is successful but when I look at the data size it is exactly half 2034 of what was encoded. I dont understand why that would be. I would of imagined I would get 4608 considering the encoder and decoder are the same.

Can anyone shed some light into why this would be happening. Anything I should be setting?

Answer 1

The requested decoder sample format should be set using audio_decodecContext->request_sample_fmt . sample_fmt is set by the decoder itself, and may be different, in which case you should use libswresample to convert between sample formats.