【Android多媒体（重采样与混频）】从playback流程分析AudioResampler，AudioMixer

文章大致内容介绍

本文主要讲述播放音乐流程，AudioResampler是如何生效的

• 播放流程
• AudioMixer分析
• 从MixerThread分析

播放流程

在Android中间，如果使用硬解码OffloadThread是不会出现混频和重采样的，但是播放需要软解的音频，则会按需要重采样。
为了方便研究，这里修改软件配置文件，让primary compress_offload仅仅支持8000的采样率播放，音乐播放会去跑MixerThread线程，这个线程上面才会出现重采样。
configs/msm8909/audio_policy_configuration.xml修改如下：

@@ -69,7 +69,7 @@
                 <mixPort name="compressed_offload" role="source"
                          flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
                     <profile name="" format="AUDIO_FORMAT_MP3"
-                             samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                             samplingRates="8000"
                              channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_MONO"/>

使用播放流程如下：

AudioMixer分析：

我们从log出发，查看代码调用流程

AudioMixer: add track (0)
AudioMixer::prepareForDownmix(0xb4f7d060) with mask 0x3
AudioMixer::prepareForReformat(0xb4f7d060) with format 0x1
AudioMixer: enable(0) AudioMixer: setParameter(VOLUME, VOLUME0: 0000)
AudioMixer: setParameter(VOLUME, VOLUME1: 0000)
AudioMixer: Creating resampler from track 44100 Hz to device 48000 Hz
AudioResampler: resampler load 0 -> 6 MHz due to delta +6 MHz from quality 6
AudioResampler: Create dynamic Resampler = 6
AudioMixer: setParameter(RESAMPLE, SAMPLE_RATE, 44100) AudioMixer: setParameter(TRACK, MIXER_FORMAT, 0x5) AudioMixer: setParameter(TRACK, MAIN_BUFFER, 0xb4f7a000) AudioMixer: mixer configuration change: 1
AudioMixer:activeTracks (00000001) all16BitsStereoNoResample=0, resampling=1, volumeRamp=0
AudioMixer: track__Resample

这段log的调用流程

1，AudioFlinger::MixerThread::checkForNewParameter_l,
2，AudioFlinger::MixerThread::getTrackName_l
3， AudioMixer::getTrackName(audio_channel_mask_t
channelMask,audio_format_t format, int sessionId)
4，调用prepare处理

prepareForDownmix这个函数，是用来处理多通道，将多通道转化，按照条件选择DownMixer或者Remixer，函数如下：

status_t AudioMixer::track_t::prepareForDownmix()
{
    ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
            this, channelMask);

    // discard the previous downmixer if there was one
    unprepareForDownmix();
    // MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks
    // are not the same and not handled internally, as mono -> stereo currently is.
    if (channelMask == mMixerChannelMask
            || (channelMask == AUDIO_CHANNEL_OUT_MONO
                    && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
    //mMixerChannelMask 在getTrackName中间被赋值为双声道，判断是否需要downMixer
        return NO_ERROR;
    }
    // DownmixerBufferProvider is only used for position masks.
    if (audio_channel_mask_get_representation(channelMask)
                == AUDIO_CHANNEL_REPRESENTATION_POSITION
            && DownmixerBufferProvider::isMultichannelCapable()) {
     //条件满足channelMask&0x3为0，即声道数大于2，并且系统支持兼容多声道，则初始化相应的bufferprovider用来改变通道数
        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
                mMixerChannelMask,
                AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
                sampleRate, sessionId, kCopyBufferFrameCount);

        if (pDbp->isValid()) { // if constructor completed properly
            mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
            downmixerBufferProvider = pDbp;
            reconfigureBufferProviders();
            return NO_ERROR;
        }
        delete pDbp;
    }
    //使用remixer的情况

    // Effect downmixer does not accept the channel conversion.  Let's use our remixer.
    RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
            mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
    // Remix always finds a conversion whereas Downmixer effect above may fail.
    downmixerBufferProvider = pRbp;
    reconfigureBufferProviders();
    return NO_ERROR;
}

prepareForReformat()判断是否需要格式转换：

status_t AudioMixer::track_t::prepareForReformat()
{
    ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
    // discard previous reformatters
    unprepareForReformat();
    // only configure reformatters as needed
    const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
            ? mDownmixRequiresFormat : mMixerInFormat;
    bool requiresReconfigure = false;
    if (mFormat != targetFormat) {
        mReformatBufferProvider = new ReformatBufferProvider(
                audio_channel_count_from_out_mask(channelMask),
                mFormat,
                targetFormat,
                kCopyBufferFrameCount);
        requiresReconfigure = true;
    }
    if (targetFormat != mMixerInFormat) {
        mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
                audio_channel_count_from_out_mask(mMixerChannelMask),
                targetFormat,
                mMixerInFormat,
                kCopyBufferFrameCount);
        requiresReconfigure = true;
    }
    if (requiresReconfigure) {
        reconfigureBufferProviders();
    }
    return NO_ERROR;
}

从AudioMixer到AudioResampler

调用的时序图如下：

voidAudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
它是把各个track的声音数据相加。所谓声音数据，可以认为是一个个的采样点，Android默认支持的采样精度是16bit的，格式为signedPCM，所以每个采样点用有符号的16位数int16_t表示。如果直接加16bit的数据，肯定会造成16bit的值溢出，Android的做法是强转成int32_t，相加，并把和赋值给了32bit的数。注意，相加前乘上了音量，而表达音量的数据类型也是int32_t。这样，就能保证在这个过程中是不会溢出的。

void ditherAndClamp(int32_t* out, constint32_t *sums, size_t c) 将32bit的输出转成16bit，再把右声道放高16bit，左声道放低16bit输出

AudioMixer->AudioResampler传递设置参数处理主要的函数：prepareTracks_l()

这个函数中设置重采样参数等操作
// prepareTracks_l() must be called with ThreadBase::mLock held
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
        Vector< sp<Track> > *tracksToRemove)
{

    mixer_state mixerStatus = MIXER_IDLE;
    // find out which tracks need to be processed
    size_t count = mActiveTracks.size();
    size_t mixedTracks = 0;
    size_t tracksWithEffect = 0;
    // counts only _active_ fast tracks
    size_t fastTracks = 0;
    uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset

    float masterVolume = mMasterVolume;
    bool masterMute = mMasterMute;
#ifdef DOLBY_ENABLE // DOLBY_DAP_PREGAIN
    // The maximum volume of left channel and right channel for pregain calculation.
    uint32_t max_vol = 0;
    float dvlf = 0.0f;
    float dvrf = 0.0f;
#endif // DOLBY_END

    if (masterMute) {
        masterVolume = 0;
    }
    // Delegate master volume control to effect in output mix effect chain if needed
    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
    if (chain != 0) {
        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
        chain->setVolume_l(&v, &v);
        masterVolume = (float)((v + (