decode FLAC file with avcodec_decode_audio4 not work well

Question

I use FFmpeg to decode my flac file and write it to pcm file, then use GoldenWave to play it with pcm signed 16bit, little endian, mono and the total play time is ok.

I doubt i write the 2 channel file in one place, but i don't know how to get every signal channel and write it to pcm file.

any help? thank you.

while (av_read_frame(fmt_ctx, &pkt) >= 0) {
            AVPacket orig_pkt = pkt;
            do {
                ret = decode_packet(&got_frame, 0);
                if (ret < 0)
                    break;
                pkt.data += ret;
                pkt.size -= ret;
            } while (pkt.size > 0);
            av_free_packet(&orig_pkt);
        }

        pkt.data = NULL;
        pkt.size = 0;
        do {
            decode_packet(&got_frame, 1);
            LOG("flush cached frames");
        } while (got_frame);



static int decode_packet(int *got_frame, int cached)
{
    int ret = 0;
    int decoded = pkt.size;

    *got_frame = 0;

    if (pkt.stream_index == audio_stream_idx) {
        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
            LOG("Error decoding audio frame (%s)\n", av_err2str(ret));
            return ret;
        }

        decoded = FFMIN(ret, pkt.size);

        if (*got_frame) {
            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(audio_dec_ctx->sample_fmt);
            //decode packet  nb_samples:4608, xx:2, unpadded_linesize: 9216
            LOG("decode packet  nb_samples:%d, xx:%d, unpadded_linesize: %d",
                    frame->nb_samples, av_get_bytes_per_sample(audio_dec_ctx->sample_fmt), unpadded_linesize);
            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);

            //int nb_sample = frame->nb_samples;

            //fwrite(frame->extended_data[0], 1, nb_sample, audio_dst_file);
            //fwrite(frame->extended_data[0] + nb_sample, 1, nb_sample, audio_dst_file);

        }
    }

    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
        av_frame_unref(frame);

    return decoded;
}

Answer 1

You didn't describe the problem you're having, but from what you're writing, I see two problems:

1. you're not checking the raw audio format of the frame, see frame->format (or audio_dec_ctx->sample_fmt). You're writing it as if it were AV_SAMPLE_FMT_S16, but you're not checking that it is
2. your unpadded_linesize is not multiplied by the number of channels (see e.g. frame->channels)