Android AudioFlinger（五）—— 揭开AudioMixer面纱

前言：

在 Android 音频系统中，AudioMixer 是音频框架中一个关键的组件，用于处理多路音频流的混音操作。它主要存在于音频回放路径中，是 AudioFlinger 服务的一部分。

上一节我们讲threadloop的时候，提到了一个函数prepareTracks_l，在这个函数的最后就调用了 mAudioMixer->create、mAudioMixer->setParameter去设置参数，channel、format、volume等等。

AudioMixer继承自 AudioMixerBase，当我们去看AudioMixer的构造函数的时候发现并没有做任何操作

那他的初始化代码在哪里呢？

走进AudioMixer：

我们看prepareTracks_l内关于mAudioMixer的调用流程就可以发现，他首先调用了create函数，然而Audiomixer内部却没有实现create接口，我们追溯到它的父类，发现在AudioMixerBase对象种定义了create接口并且实现了。

我们粗略的看下create里主要做了什么,代码多我做了删减。

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;}
}

可以看到除了一开始做了channel和format的判断，后面基本上就是对track的初始化，像volume、channel、format、sampleRate还有Hook的初始化。

初始化完成后就开始调用AudioMixer内部的接口了，我们依次往下看发现还有getUnreleasedFrames、setParameter、setBufferProvider、process等。
我们先看下setParameter,当属性变化的时候就会调用到这里。

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);}
}

函数的主要结构就是一个switch,首先通过trackId找到对应的track对象，然后去设置对应track的parameter参数，例如 CHANNEL_MASK、FORMAT、MAIN_BUFFER等。

这只是设置参数，那混音在哪里呢？我们继续往下看process

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

这里主要就是调用mHook，mHook是一个函数指针，他会根据不同的场景分别调用不同的函数。

• process__nop：初始值
• process__genericResampling：对两路以上的track进行重采样操作
• process__genericNoResampling：对两路以上的track不进行重采样操作
• process__validate：这个函数就是根据当前的不同情况将mHook指向不同的函数
• process__oneTrack16BitsStereoNoResampling：只有一路track，16bit，立体声的时候不进行重采样

process_hook_t mHook = &AudioMixerBase::process__nop;

mHook初始化的时候指向的是process__nop

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

process__validate是在invalidate函数里幅值给了mHook 指针。

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());}}}}}
}

这个函数首先使用while循环来遍历每一个track，然后通过 NEEDS_RESAMPLE、NEEDS_AUX、NEEDS_CHANNEL_1、NEEDS_MUTE等判断，最终得到resampling、all16BitsStereoNoResample、volumeRamp的值，然后基于这几个值来决定调用，mHook来指向哪一个函数。

至于音频流数据是如何混到一起的，我们后面章节再来进一步分析。