Question

我使用FFmpeg解码我的flac文件并将其写入pcm文件，然后使用GoldenWave与pcm signed 16bit, little endian, mono一起播放，总播放时间正常。

我怀疑我是否在一个地方写了2声道文件，但我不知道如何获取每个信号通道并将其写入pcm文件。

任何帮助？谢谢。

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:

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
your unpadded_linesize is not multiplied by the number of channels (see e.g. frame->channels)

使用avcodec_decode_audio4解码FLAC文件效果不佳

1 个答案: