前言：

在 Android 音頻系統中，AudioMixer 是音頻框架中一個關鍵的組件，用于處理多路音頻流的混音操作。它主要存在于音頻回放路徑中，是 AudioFlinger 服務的一部分。

上一節我們講threadloop的時候，提到了一個函數prepareTracks_l，在這個函數的最后就調用了 mAudioMixer->create、mAudioMixer->setParameter去設置參數，channel、format、volume等等。

AudioMixer繼承自 AudioMixerBase，當我們去看AudioMixer的構造函數的時候發現并沒有做任何操作

那他的初始化代碼在哪里呢？

走進AudioMixer：

我們看prepareTracks_l內關于mAudioMixer的調用流程就可以發現，他首先調用了create函數，然而Audiomixer內部卻沒有實現create接口，我們追溯到它的父類，發現在AudioMixerBase對象種定義了create接口并且實現了。

我們粗略的看下create里主要做了什么,代碼多我做了刪減。

status_t AudioMixerBase::create(int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
{LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);if (!isValidChannelMask(channelMask)) {ALOGE("%s invalid channelMask: %#x", __func__, channelMask);return BAD_VALUE;}if (!isValidFormat(format)) {ALOGE("%s invalid format: %#x", __func__, format);return BAD_VALUE;}auto t = preCreateTrack();{t->needs = 0;t->volume[0] = 0;...t->channelCount = audio_channel_count_from_out_mask(channelMask);t->enabled = false;t->channelMask = channelMask;t->sessionId = sessionId;t->hook = NULL;...// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)t->sampleRate = mSampleRate;t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;t->mFormat = format;t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);t->mInputFrameSize = audio_bytes_per_frame(t->channelCount, t->mFormat);status_t status = postCreateTrack(t.get());if (status != OK) return status;mTracks[name] = t;return OK;}
}

可以看到除了一開始做了channel和format的判斷，后面基本上就是對track的初始化，像volume、channel、format、sampleRate還有Hook的初始化。

初始化完成后就開始調用AudioMixer內部的接口了，我們依次往下看發現還有getUnreleasedFrames、setParameter、setBufferProvider、process等。
我們先看下setParameter,當屬性變化的時候就會調用到這里。

void AudioMixer::setParameter(int name, int target, int param, void *value)
{LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);const std::shared_ptr<Track> &track = getTrack(name);int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));int32_t *valueBuf = reinterpret_cast<int32_t*>(value);switch (target) {case TRACK:switch (param) {case CHANNEL_MASK: {const audio_channel_mask_t trackChannelMask =static_cast<audio_channel_mask_t>(valueInt);if (setChannelMasks(name, trackChannelMask,static_cast<audio_channel_mask_t>(track->mMixerChannelMask | track->mMixerHapticChannelMask))) {ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);invalidate();}} break;case MAIN_BUFFER:if (track->mainBuffer != valueBuf) {track->mainBuffer = valueBuf;ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);if (track->mKeepContractedChannels) {track->prepareForAdjustChannels(mFrameCount);}invalidate();}break;case AUX_BUFFER:AudioMixerBase::setParameter(name, target, param, value);break;case FORMAT: {audio_format_t format = static_cast<audio_format_t>(valueInt);if (track->mFormat != format) {ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);track->mFormat = format;ALOGV("setParameter(TRACK, FORMAT, %#x)", format);track->prepareForReformat();invalidate();}} break;case MIXER_FORMAT: {audio_format_t format = static_cast<audio_format_t>(valueInt);if (track->mMixerFormat != format) {track->mMixerFormat = format;ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);if (track->mKeepContractedChannels) {track->prepareForAdjustChannels(mFrameCount);}}} break;case MIXER_CHANNEL_MASK: {const audio_channel_mask_t mixerChannelMask =static_cast<audio_channel_mask_t>(valueInt);if (setChannelMasks(name, static_cast<audio_channel_mask_t>(track->channelMask | track->mHapticChannelMask),mixerChannelMask)) {ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);invalidate();}} break;
...default:LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);}break;case RESAMPLE:case RAMP_VOLUME:case VOLUME:AudioMixerBase::setParameter(name, target, param, value);break;case TIMESTRETCH:switch (param) {case PLAYBACK_RATE: {const AudioPlaybackRate *playbackRate =reinterpret_cast<AudioPlaybackRate*>(value);
...} break;default:LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);}break;default:LOG_ALWAYS_FATAL("setParameter: bad target %d", target);}
}

函數的主要結構就是一個switch,首先通過trackId找到對應的track對象，然后去設置對應track的parameter參數，例如 CHANNEL_MASK、FORMAT、MAIN_BUFFER等。

這只是設置參數，那混音在哪里呢？我們繼續往下看process

void process() {preProcess();(this->*mHook)();postProcess();
}

這里主要就是調用mHook，mHook是一個函數指針，他會根據不同的場景分別調用不同的函數。

• process__nop：初始值
• process__genericResampling：對兩路以上的track進行重采樣操作
• process__genericNoResampling：對兩路以上的track不進行重采樣操作
• process__validate：這個函數就是根據當前的不同情況將mHook指向不同的函數
• process__oneTrack16BitsStereoNoResampling：只有一路track，16bit，立體聲的時候不進行重采樣

process_hook_t mHook = &AudioMixerBase::process__nop;

mHook初始化的時候指向的是process__nop

void invalidate() {mHook = &AudioMixerBase::process__validate;}

process__validate是在invalidate函數里幅值給了mHook 指針。

void AudioMixerBase::process__validate()
{// select the processing hooksmHook = &AudioMixerBase::process__nop;if (mEnabled.size() > 0) {if (resampling) {if (mOutputTemp.get() == nullptr) {mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);}if (mResampleTemp.get() == nullptr) {mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);}mHook = &AudioMixerBase::process__genericResampling;} else {// we keep temp arrays around.mHook = &AudioMixerBase::process__genericNoResampling;if (all16BitsStereoNoResample && !volumeRamp) {if (mEnabled.size() == 1) {const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];if ((t->needs & NEEDS_MUTE) == 0) {// The check prevents a muted track from acquiring a process hook.//// This is dangerous if the track is MONO as that requires// special case handling due to implicit channel duplication.// Stereo or Multichannel should actually be fine here.mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat,t->useStereoVolume());}}}}}
}

這個函數首先使用while循環來遍歷每一個track，然后通過 NEEDS_RESAMPLE、NEEDS_AUX、NEEDS_CHANNEL_1、NEEDS_MUTE等判斷，最終得到resampling、all16BitsStereoNoResample、volumeRamp的值，然后基于這幾個值來決定調用，mHook來指向哪一個函數。

至于音頻流數據是如何混到一起的，我們后面章節再來進一步分析。