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-rw-r--r--services/audioflinger/AudioMixer.cpp61
1 files changed, 44 insertions, 17 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 1200f75..98415b1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <assert.h>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -30,6 +31,9 @@
#include <system/audio.h>
+#include <common_time/local_clock.h>
+#include <common_time/cc_helper.h>
+
#include "AudioMixer.h"
namespace android {
@@ -47,6 +51,9 @@ static inline int16_t clamp16(int32_t sample)
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
: mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
@@ -74,6 +81,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->in = 0;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->localTimeFreq = mLocalTimeFreq;
t++;
}
}
@@ -293,6 +301,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
if (resampler == 0) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
+ resampler->setLocalTimeFreq(localTimeFreq);
}
return true;
}
@@ -357,13 +366,13 @@ status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
-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)
{
LOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -467,7 +476,7 @@ void AudioMixer::process__validate(state_t* state)
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
@@ -876,7 +885,7 @@ void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
}
// 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;
@@ -906,7 +915,9 @@ void AudioMixer::process__nop(state_t* state)
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) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
@@ -916,7 +927,7 @@ void AudioMixer::process__nop(state_t* state)
}
// 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)));
@@ -928,7 +939,7 @@ void AudioMixer::process__genericNoResampling(state_t* state)
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
@@ -982,7 +993,9 @@ void AudioMixer::process__genericNoResampling(state_t* state)
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);
@@ -1011,7 +1024,7 @@ void AudioMixer::process__genericNoResampling(state_t* state)
// generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1050,6 +1063,7 @@ void AudioMixer::process__genericResampling(state_t* state)
// 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 {
@@ -1057,7 +1071,8 @@ void AudioMixer::process__genericResampling(state_t* state)
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.
@@ -1077,7 +1092,8 @@ void AudioMixer::process__genericResampling(state_t* state)
}
// 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)
{
const int i = 31 - __builtin_clz(state->enabledTracks);
const track_t& t = state->tracks[i];
@@ -1092,7 +1108,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
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);
int16_t const *in = b.i16;
// in == NULL can happen if the track was flushed just after having
@@ -1135,7 +1152,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
// 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;
@@ -1169,7 +1187,9 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
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];
@@ -1183,7 +1203,9 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
}
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];
@@ -1230,6 +1252,11 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
}
}
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex)
+{
+ return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+}
+
// ----------------------------------------------------------------------------
}; // namespace android
-
generated by cgit v1.1 at 2025-11-13 12:24:11 +0000