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AudioMixer是Android的混音器，通过混音器可以把各个音轨的音频数据混合在一起，然后输出到音频设备。
AudioMixer在MixerThread的构造函数内创建：
AudioFlinger::MixerThread::MixerThread(...)
{
...
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
...
}
这说明了一个MixerThread对应一个AudioMixer。
而且MixerThread传了两个参数给AudioMixer：
在上一篇描述MixerThread的时候说过，prepareTrack_l内会配置AudioMixer的参数，现在来详细分析一下各个参数的作用。
设置混音的源buffer，name为传入的索引，track即从mActiveTracks取出来的Track
关于索引name，在这里深入分析，name的获取过程如下：
int name = track->name();
+
+--> int name() const { return mName; }
+
+--> mName = thread->getTrackName_l(channelMask, sessionId);
+
+--> return mAudioMixer->getTrackName(channelMask, sessionId);
+
+--> uint32_t names = (~mTrackNames) & mConfiguredNames;
|
+--> int n = __builtin_ctz(names);
names为索引的集合，names的每一个bit代表不同的索引，names上的某个bit为1，就代表该bit可以取出来作为索引，__builtin_ctz的作用是计算names的低位0的个数，即可以取出最低位为1的bit作为索引。如下：
11111111111111111111000000000000
^
低位有12个0，则取bit12作为索引，那么返回的索引值为1<<12
决定names的参数有两个：
enable方法只是把track的enabled置为true，然后调用invalidateState(1 << name);表明需要调用刷新函数。
void AudioMixer::enable(int name)
{
name -= TRACK0;
track_t& track = mState.tracks[name];
if (!track.enabled) {
track.enabled = true;
invalidateState(1 << name);
}
}
分别设置左右声道音量，然后调用invalidateState(1 << name);表明需要调用刷新函数。
case VOLUME0:
case VOLUME1:
if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
track.prevVolume[param-VOLUME0] = valueInt << 16;
track.volumeInc[param-VOLUME0] = 0;
}
保证传进来的PCM数据为16bit
case FORMAT:
ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
break;
设置通道数，mask：单音轨（mono），双音轨（stereo）…
case CHANNEL_MASK: {
audio_channel_mask_t mask = (audio_channel_mask_t) value;
if (track.channelMask != mask) {
uint32_t channelCount = popcount(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
track.channelMask = mask; //设置mask
track.channelCount = channelCount; //更新音轨数目
// the mask has changed, does this track need a downmixer?
initTrackDownmix(&mState.tracks[name], name, mask);
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
invalidateState(1 << name);
}
设置当前track的采样频率为reqSampleRate，并要求AudioMixer对当前track进行重采样，输出频率为当前AudioMixer的输出频率mSampleRate。然后调用invalidateState(1 << name);表明需要调用刷新函数。调用过程如下：
mAudioMixer->setParameter(
+ name,
| AudioMixer::RESAMPLE,
| AudioMixer::SAMPLE_RATE,
| (void *)reqSampleRate);
|
+--> track.setResampler(uint32_t(valueInt), mSampleRate)
+
+--> if (sampleRate != value) { //只有输入采样率跟输出采样率不同的时候才会进行重采样
+ if (resampler == NULL) {
| quality = AudioResampler::VERY_HIGH_QUALITY; //高级重采样
| resampler = AudioResampler::create(...); //创建resampler
| }
|}
+--> switch (quality) {
| default:
| case DEFAULT_QUALITY:
| case LOW_QUALITY:
| ALOGV("Create linear Resampler");
| resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
| break;
| case MED_QUALITY:
| ALOGV("Create cubic Resampler");
| resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
| break;
| case HIGH_QUALITY:
| ALOGV("Create HIGH_QUALITY sinc Resampler");
| resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
| break;
| case VERY_HIGH_QUALITY: //由于我们选择的是VERY_HIGH_QUALITY，所以resampler创建的是AudioResamplerSinc
| ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
| resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
| break;
| }
|
+--> // initialize resampler
resampler->init();
设置目的buffer。然后调用invalidateState(1 << name);表明需要调用刷新函数。
我们追踪一下目的buffer是在哪里创建的：
track->mainBuffer()
+
+--> int16_t *mainBuffer() const { return mMainBuffer; }
mMainBuffer是在track创建的时候就被赋值了
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(...)
+
+--> track = new Track(...)
+
+--> AudioFlinger::PlaybackThread::Track::Track(...)
+:mMainBuffer(thread->mixBuffer())
|
+--> int16_t *mixBuffer() const { return mMixBuffer; };
thread就是MixerThread，在MixerThread创建的同时，PlaybackThread也一同被创建。在PlaybackThread的构造函数内，申请了一块buffer，并赋值给mMixerBuffer
AudioFlinger::MixerThread::MixerThread
+
+--> AudioFlinger::PlaybackThread::PlaybackThread
+
+--> void AudioFlinger::PlaybackThread::readOutputParameters()
+
+--> mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1];
|
+--> mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align);
这表明了一个AudioMixer对应一个mMixBuffer，经过某个AudioMixer的音频数据最后会汇聚到一个buffer内进行输出
我们上面大量提到了invalidateState可以用来表明需要调用刷新函数，现在来分析一下。
void AudioMixer::invalidateState(uint32_t mask)
{
if (mask) {
mState.needsChanged |= mask; //mask即track->name，表明该track需要被刷新
mState.hook = process__validate;
}
}
由于AudioMixer进行混音处理的时候会调用process方法，而process调用的是mState.hook，所以调用invalidateState，会使得下一次的process函数会调用process__validate进行参数的刷新。process__validate分析如下：
void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing‘s invalid");
uint32_t changed = state->needsChanged; //所有需要invalidate的track都在这里面
state->needsChanged = 0; // clear the validation flag
// recompute which tracks are enabled / disabled
uint32_t enabled = 0;
uint32_t disabled = 0;
while (changed) { //对于所有需要invalidate的track，取出来
const int i = 31 - __builtin_clz(changed);
const uint32_t mask = 1<<i;
changed &= ~mask;
track_t& t = state->tracks[i];
(t.enabled ? enabled : disabled) |= mask; //通过track.enabled或者track.disabled来判断该track是否需要混音
}
state->enabledTracks &= ~disabled; //disabled mask
state->enabledTracks |= enabled; //enabled mask
// compute everything we need...
int countActiveTracks = 0;
bool all16BitsStereoNoResample = true;
bool resampling = false;
bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) { //对所有需要进行混音的track
const int i = 31 - __builtin_clz(en); //取出最高位为1的bit
en &= ~(1<<i); //把这一位置为0
countActiveTracks++;
track_t& t = state->tracks[i]; //取出来track
uint32_t n = 0;
n |= NEEDS_CHANNEL_1 + t.channelCount - 1; //至少有一个channel需要混音
n |= NEEDS_FORMAT_16; //必须为16bit PCM
n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED; //是否需要重采样
if (t.auxLevel != 0 && t.auxBuffer != NULL) {
n |= NEEDS_AUX_ENABLED;
}
if (t.volumeInc[0]|t.volumeInc[1]) {
volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
t.needs = n; //更新track flag
//下面为设置track的混音方法
if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) { //mute
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { //重采样
all16BitsStereoNoResample = false;
resampling = true;
t.hook = track__genericResample;
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix + resample", i);
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ //单声道
t.hook = track__16BitsMono;
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){ //双声道
t.hook = track__16BitsStereo;
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix", i);
}
}
}
}
// select the processing hooks //下面为设置整体的混音方法，一个process__xxx内会循环调用track_xxx
state->hook = process__nop;
if (countActiveTracks) {
if (resampling) { //重采样，需要多一块重采样buffer
if (!state->outputTemp) {
state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
if (!state->resampleTemp) {
state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
state->hook = process__genericResampling;
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling; //双声道process
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling; //单声道process
}
}
}
}
ALOGV("mixer configuration change: %d activeTracks (%08x) "
"all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
state->hook(state, pts); //这里调用一次进行混音，后续会在MixerThread的threadLoop_mix内调用
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
if (countActiveTracks) {
bool allMuted = true;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
track_t& t = state->tracks[i];
if (!t.doesResample() && t.volumeRL == 0)
{
t.needs |= NEEDS_MUTE_ENABLED;
t.hook = track__nop;
} else {
allMuted = false;
}
}
if (allMuted) {
state->hook = process__nop;
} else if (all16BitsStereoNoResample) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
在分析MixerThread时说过，我们调用AudioMixer的process方法进行混音的，实际上混音的方法是调用AudioMixer内的process_xxx方法，各个process方法大同小异。下面来分析process__genericResampling这个方法。
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
// this const just means that local variable outTemp doesn‘t change
int32_t* const outTemp = state->outputTemp; //重采样缓存
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
size_t numFrames = state->frameCount;
uint32_t e0 = state->enabledTracks;
while (e0) {
// process by group of tracks with same output buffer
// to optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j]; //取出第一个track t1
e2 &= ~(1<<j); //除了t1之外，其余的track的索引都在e2内
//对于其他的track，通过循环取出来，赋值为t2，如果t2的目标buffer与t1的不同，则把t2从e1的集合中去掉
//这么做就是为了把相同目标buffer的track取出来，一起进行混音，因为不同目标buffer的track是要混音输出到不同buffer的
//不过实际上一般都会有相同的目标buffer，如MixerThread设定了mMixBuffer作为track的目标buffer
//如果设定了eq(AudioEffect)那就有可能会出现不同目标buffer的情况？
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
int32_t *out = t1.mainBuffer;
memset(outTemp, 0, size);
while (e1) { //对于e1内的所有track，调用t.hook进行混音
const int i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer;
}
// this is a little goofy, on the resampling case we don‘t
// acquire/release the buffers because it‘s done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
ALOGE("[%s:%d]", __FUNCTION__, __LINE__);
t.resampler->setPTS(pts);
t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); //实际上重采样会走这里，然后输出到重采样buffer,outTemp
} else {
size_t outFrames = 0;
ALOGE("[%s:%d]", __FUNCTION__, __LINE__);
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount,
state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
}
ditherAndClamp(out, outTemp, numFrames); //把重采样buffer内的数据输出到out，即目标buffer
}
}
在process__invalidate时，设置了重采样时track.hook函数为track__genericResample，下面看一下这个函数做了什么
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
int32_t* temp, int32_t* aux)
{
//设置输入采样率
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
if (aux != NULL) {
// always resample with unity gain when sending to auxiliary buffer to be able
// to apply send level after resampling
// TODO: modify each resampler to support aux channel?
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
volumeRampStereo(t, out, outFrameCount, temp, aux);
}
// constant gain
else {
//设置音量
t->resampler->setVolume(t->volume[0], t->volume[1]);
//进行重采样
t->resampler->resample(out, outFrameCount, t->bufferProvider);
}
}
}
最终调用了resampler的resample方法进行重采样
那么，下一篇我们来分析重采样
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原文地址：http://www.cnblogs.com/TaigaCon/p/4844919.html
