AudioFlinger分析(6)

process__genericResampling

process__OneTrack16BitsStereoNoResampling process__TwoTracks16BitsStereoNoResampling

AudioMixer构造的时候，hook是process__nop，有几个地方会改变这个函数指针的指向。 这部分涉及到数字音频技术，我就无力讲解了。我们看看最接近的函数 process__OneTrack16BitsStereoNoResampling

void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, void* output) {

单track，16bit双声道，不需要重采样,大部分是这种情况了 const int i = 31 - __builtin_clz(state->enabledTracks); const track_t& t = state->tracks[i];

AudioBufferProvider::Buffer& b(t.buffer);

int32_t* out = static_cast(output); size_t numFrames = state->frameCount;

const int16_t vl = t.volume[0]; const int16_t vr = t.volume[1]; const uint32_t vrl = t.volumeRL; while (numFrames) {

b.frameCount = numFrames; //获得buffer

t.bufferProvider->getNextBuffer(&b); int16_t const *in = b.i16;

size_t outFrames = b.frameCount; if UNLIKELY--->不走这. else { do {

//计算音量等数据，和数字音频技术有关。这里不说了

uint32_t rl = *reinterpret_cast(in);

in += 2;

int32_t l = mulRL(1, rl, vrl) >> 12; int32_t r = mulRL(0, rl, vrl) >> 12; *out++ = (r<<16) | (l & 0xFFFF); } while (--outFrames); }

numFrames -= b.frameCount; //释放buffer。

t.bufferProvider->releaseBuffer(&b); } }

好像挺简单的啊，不就是把数据处理下嘛。这里注意下buffer。到现在，我们还没看到取共享内存里AT端write的数据呐。 那只能到bufferProvider去看了。

注意，这里用的是AudioBufferProvider基类，实际的对象是Track。它从AudioBufferProvider派生。

我们用得是PlaybackThread的这个Track status_t

AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) {

//一阵暗喜吧。千呼万唤始出来，终于见到cblk了 audio_track_cblk_t* cblk = this->cblk(); uint32_t framesReady;

uint32_t framesReq = buffer->frameCount; //哈哈，看看数据准备好了没，

framesReady = cblk->framesReady();

if (LIKELY(framesReady)) { uint32_t s = cblk->server;

uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;

bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; if (framesReq > framesReady) { framesReq = framesReady;

}

if (s + framesReq > bufferEnd) { framesReq = bufferEnd - s; } 获得真实的数据地址

buffer->raw = getBuffer(s, framesReq); if (buffer->raw == 0) goto getNextBuffer_exit;

buffer->frameCount = framesReq; return NO_ERROR; }

getNextBuffer_exit: buffer->raw = 0; buffer->frameCount = 0; return NOT_ENOUGH_DATA; }

再看看释放缓冲的地方：releaseBuffer，这个直接在ThreadBase中实现了 void

AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) {

buffer->raw = 0;

mFrameCount = buffer->frameCount; step();

buffer->frameCount = 0; }

看看step吧。mFrameCount表示我已经用完了这么多帧。 bool AudioFlinger::ThreadBase::TrackBase::step() { bool result;

audio_track_cblk_t* cblk = this->cblk();

result = cblk->stepServer(mFrameCount);//哼哼，调用cblk的stepServer，更新 服务端的使用位置 return result;

}

到这里，大伙应该都明白了吧。原来AudioTrack中write的数据，最终是这么被使用的呀！！！ 恩，看一个process__OneTrack16BitsStereoNoResampling不过瘾，再看看 process__TwoTracks16BitsStereoNoResampling。

void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output)

int i;

uint32_t en = state->enabledTracks;

i = 31 - __builtin_clz(en);

const track_t& t0 = state->tracks[i]; AudioBufferProvider::Buffer& b0(t0.buffer);

en &= ~(1<

i = 31 - __builtin_clz(en);

const track_t& t1 = state->tracks[i]; AudioBufferProvider::Buffer& b1(t1.buffer);

int16_t const *in0;

const int16_t vl0 = t0.volume[0]; const int16_t vr0 = t0.volume[1]; size_t frameCount0 = 0;

int16_t const *in1;

const int16_t vl1 = t1.volume[0]; const int16_t vr1 = t1.volume[1]; size_t frameCount1 = 0;

int32_t* out = static_cast(output); size_t numFrames = state->frameCount; int16_t const *buff = NULL;

while (numFrames) {

if (frameCount0 == 0) { b0.frameCount = numFrames;

t0.bufferProvider->getNextBuffer(&b0); if (b0.i16 == NULL) { if (buff == NULL) {

buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; }

in0 = buff;

b0.frameCount = numFrames; } else { in0 = b0.i16; }

frameCount0 = b0.frameCount; }

if (frameCount1 == 0) { b1.frameCount = numFrames;

t1.bufferProvider->getNextBuffer(&b1); if (b1.i16 == NULL) { if (buff == NULL) {

buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; }

in1 = buff;

b1.frameCount = numFrames; } else { in1 = b1.i16; }

frameCount1 = b1.frameCount; }