Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index cb7678b..c399691 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -33,6 +33,8 @@
#include <system/audio.h>
#include <audio_utils/primitives.h>
+#include <common_time/local_clock.h>
+#include <common_time/cc_helper.h>
#include "AudioMixer.h"
@@ -45,6 +47,9 @@
{
// AudioMixer is not yet capable of multi-channel beyond stereo
assert(2 == MAX_NUM_CHANNELS);
+
+ LocalClock lc;
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
@@ -80,6 +85,7 @@
t->sampleRate = mSampleRate;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->localTimeFreq = lc.getLocalFreq();
t++;
}
}
@@ -289,6 +295,7 @@
if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
+ resampler->setLocalTimeFreq(localTimeFreq);
}
return true;
}
@@ -333,13 +340,13 @@
-void AudioMixer::process()
+void AudioMixer::process(int64_t pts)
{
- mState.hook(&mState);
+ mState.hook(&mState, pts);
}
-void AudioMixer::process__validate(state_t* state)
+void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -443,7 +450,7 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
+ state->hook(state, pts);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
@@ -757,7 +764,7 @@
}
// no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
@@ -787,7 +794,9 @@
size_t outFrames = state->frameCount;
while (outFrames) {
t1.buffer.frameCount = outFrames;
- t1.bufferProvider->getNextBuffer(&t1.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t1, pts, state->frameCount - outFrames);
+ t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS);
if (t1.buffer.raw == NULL) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
@@ -797,7 +806,7 @@
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
@@ -809,7 +818,7 @@
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
@@ -863,7 +872,9 @@
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t, pts, numFrames + (BLOCKSIZE - outFrames));
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
@@ -892,7 +903,7 @@
// generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+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;
@@ -932,6 +943,7 @@
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+ t.resampler->setPTS(pts);
(t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
@@ -939,7 +951,8 @@
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ 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.
@@ -959,7 +972,8 @@
}
// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
// This method is only called when state->enabledTracks has exactly
// one bit set. The asserts below would verify this, but are commented out
@@ -979,7 +993,8 @@
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
- t.bufferProvider->getNextBuffer(&b);
+ int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
+ t.bufferProvider->getNextBuffer(&b, outputPTS);
const int16_t *in = b.i16;
// in == NULL can happen if the track was flushed just after having
@@ -1023,7 +1038,8 @@
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
int i;
uint32_t en = state->enabledTracks;
@@ -1057,7 +1073,9 @@
if (frameCount0 == 0) {
b0.frameCount = numFrames;
- t0.bufferProvider->getNextBuffer(&b0);
+ int64_t outputPTS = calculateOutputPTS(t0, pts,
+ out - t0.mainBuffer);
+ t0.bufferProvider->getNextBuffer(&b0, outputPTS);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1071,7 +1089,9 @@
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
- t1.bufferProvider->getNextBuffer(&b1);
+ int64_t outputPTS = calculateOutputPTS(t1, pts,
+ out - t0.mainBuffer);
+ t1.bufferProvider->getNextBuffer(&b1, outputPTS);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1117,5 +1137,14 @@
}
#endif
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex)
+{
+ if (AudioBufferProvider::kInvalidPTS == basePTS)
+ return AudioBufferProvider::kInvalidPTS;
+
+ return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+}
+
// ----------------------------------------------------------------------------
}; // namespace android