关于sonic实现音频变速不变调的问题

SPeak

可以提供一个最小可复现示例项目的zip或git仓库链接, 以及崩溃时的调用栈...
(大面积代码更建议用文本, 截图不方便)

hammerers

我写了一个能够复现该问题的项目git@github.com:hammerers/sonic_ffmpeg_sdl_audio_player.git

SPeak

@hammerers 是只要speed不是1就会报错小于1和大于1有区别吗, crash的时候调用栈有吗

补充仓库地址: https://github.com/hammerers/sonic_ffmpeg_sdl_audio_player

hammerers

debug的时候有三个地方会中断，甚至有时候可以播放出变速的声音只是是全损音质，debug的调用栈显示的都是SDL的dll，或者在sonic.c
Screenshot 2025-08-15 175323.png

hammerers

speed大于小于一都会有这种情况

SPeak

这里往stream里写数据的时候没有加filled的偏移, 导致重复把数据写到了buff的开头部分

hammerers

此回复已被删除！

hammerers

天哪，太粗心了，谢谢！尽管修复了它之后还是会崩溃

hammerers

此回复已被删除！

Cat-Guijun

hi,友友有解决这个情况吗，我也是当speed不为一时就报错了，我完全不知道如何查找崩溃，找了一天了，下面是我的代码：
void AudioDecodeThread::run()
{
// 加入sonic
// 初始化部分：先创建一个流
sonicStream stream = sonicCreateStream(m_videoPlayControl->m_audioCodecCtx->sample_rate, 2);
if (!stream)
{
m_videoPlayControl->m_eventHandler->onVideoPlayerError(0, "sonicCreateStream error!");
}

    if (!initSwrCtx())
    {
        m_videoPlayControl->m_eventHandler->onVideoPlayerError(0, "initSwrCtx error");
        return;
    }

    while (true)
    {
        if (m_videoPlayControl->m_currentState == State::STOP)
        {
            break;
        }
        else if (m_videoPlayControl->m_currentState == State::PAUSE)
        {
            std::this_thread::sleep_for(std::chrono::milliseconds(10));
            continue;
        }

        AVPacket *pkt = m_videoPlayControl->m_audioPacketQueue.pop();
        if (pkt == nullptr)
        {
            if (m_videoPlayControl->m_isReadFinished)
            {
                break;
            }
            continue;
        }

        if (memcmp(pkt->data, FLUSHDATA, strlen(FLUSHDATA)) == 0)
        {
            avcodec_flush_buffers(m_videoPlayControl->m_audioCodecCtx);
            av_packet_free(&pkt);
            continue;
        }

        int ret = avcodec_send_packet(m_videoPlayControl->m_audioCodecCtx, pkt);
        if (ret < 0)
        {
            char errbuf[1024] = { 0 };
            av_strerror(ret, errbuf, sizeof(errbuf));
            fprintf(stderr, "avcodec_send_packet: %s (error pts: %d)\n", errbuf, pkt->pts);
            av_packet_free(&pkt);
            continue;
        }

        AVFrame *frame = av_frame_alloc();
        while (avcodec_receive_frame(m_videoPlayControl->m_audioCodecCtx, frame) == 0)
        {
            // 读取frame, swr转换, 放到frameQueue里面去
            int inSamples = frame->nb_samples;
            // int outSamples = av_rescale_rnd(inSamples, 1, 1, AV_ROUND_UP);

            // 废除手动分配的 audioBuffer，改用 resampleFrame 内部缓冲区
            AVFrame *resampleFrame = av_frame_alloc();
            resampleFrame->format = AV_SAMPLE_FMT_S16;
            av_channel_layout_default(&resampleFrame->ch_layout, 2); // 双声道
            resampleFrame->sample_rate = frame->sample_rate;         // 正确设置采样率
            resampleFrame->nb_samples = inSamples;

            // 让 FFmpeg 自动分配缓冲区（确保地址和大小正确）
            if (av_frame_get_buffer(resampleFrame, 0) < 0)
            {
                av_frame_free(&resampleFrame);
                continue;
            }

            int len = swr_convert(m_swrCtx, resampleFrame->data, resampleFrame->nb_samples, (const uint8_t **)frame->data, inSamples);

            // 加入sonic

            sonicSetSpeed(stream, m_videoPlayControl->m_speed);
            // 4. 写入音频数据到Sonic ,这里的bug

            short *valid_audio_data = (short *)resampleFrame->data[0];
            int total_samples = len; // 单声道总样本数

            std::cout << len << std::endl;

            ret = sonicWriteShortToStream(stream, valid_audio_data, total_samples);
            if (ret == 0)
            {
                m_videoPlayControl->m_eventHandler->onVideoPlayerError(0, "sonicWriteShortToStream error");
                continue;
            }

            int availableSamples = sonicSamplesAvailable(stream); //
            if (availableSamples <= 0)
            {
                continue;
            }

            // 根据实际需要的样本数分配缓冲区
            short *output_pcm = new short[availableSamples * 2];
            int samples = sonicReadShortFromStream(stream, output_pcm, availableSamples);
            if (samples <= 0)
            {
                delete[] output_pcm;
                av_frame_free(&resampleFrame);
                continue;
            }

            std::cout << inSamples << "---" << len << "---" << availableSamples << "---" << samples << std::endl;

            PCMFrame *pcm = new PCMFrame();
            pcm->setBuffer((uint8_t *)output_pcm, samples * 4);
            pcm->setPts(frame->pts);
            pcm->setSampleRate(frame->sample_rate);
            m_videoPlayControl->m_audioFrameQueue.push(pcm);

            delete[] output_pcm;
            av_frame_free(&resampleFrame);
        }
        av_frame_free(&frame);
        av_packet_free(&pkt);
    }
    sonicDestroyStream(stream);
}

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