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-rw-r--r--services/audioflinger/AudioFlinger.cpp94
-rw-r--r--services/audioflinger/AudioFlinger.h108
-rw-r--r--services/audioflinger/AudioPolicyService.cpp6
-rw-r--r--services/audioflinger/AudioResampler.cpp24
-rw-r--r--services/audioflinger/AudioResampler.h6
-rw-r--r--services/audioflinger/AudioResamplerCubic.cpp4
-rw-r--r--services/audioflinger/AudioResamplerSinc.cpp23
7 files changed, 122 insertions, 143 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 93c91fb..285f7a1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -748,7 +748,7 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
{
status_t result;
- ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+ ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
@@ -821,7 +821,7 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
-// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
@@ -1524,12 +1524,11 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
- uint32_t strategy =
- AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
+ uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
- uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
+ uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
@@ -1551,7 +1550,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
+ chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
@@ -1781,7 +1780,7 @@ uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
- return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
+ return AudioSystem::getStrategyForStream(track->streamType());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
@@ -1920,7 +1919,7 @@ bool AudioFlinger::MixerThread::threadLoop()
if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended)) {
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+ ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -1934,9 +1933,9 @@ bool AudioFlinger::MixerThread::threadLoop()
releaseWakeLock_l();
// wait until we have something to do...
- ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("MixerThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
+ ALOGV("MixerThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
@@ -2162,7 +2161,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
@@ -2170,7 +2169,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
} else {
// read original volumes with volume control
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
vl = vlr & 0xFFFF;
@@ -2331,7 +2330,7 @@ void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
- if (t->type() == streamType) {
+ if (t->streamType() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
@@ -2505,21 +2504,6 @@ AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
@@ -2638,7 +2622,7 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
mSuspended)) {
// wait until we have something to do...
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p\n", this);
+ ALOGV("Audio hardware entering standby, mixer %p", this);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -2651,9 +2635,9 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+ ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid());
acquireWakeLock_l();
if (!mMasterMute) {
@@ -2704,13 +2688,13 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
@@ -3046,9 +3030,9 @@ bool AudioFlinger::DuplicatingThread::threadLoop()
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ ALOGV("DuplicatingThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
@@ -3391,7 +3375,7 @@ AudioFlinger::PlaybackThread::Track::Track(
mName = playbackThread->getTrackName_l();
mMainBuffer = playbackThread->mixBuffer();
}
- ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
ALOGE("no more track names available");
}
@@ -3428,9 +3412,7 @@ void AudioFlinger::PlaybackThread::Track::destroy()
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
@@ -3523,7 +3505,7 @@ bool AudioFlinger::PlaybackThread::Track::isReady() const {
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
- ALOGV("start(%d), calling thread %d session %d",
+ ALOGV("start(%d), calling pid %d session %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
@@ -3541,9 +3523,7 @@ status_t AudioFlinger::PlaybackThread::Track::start()
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
- status = AudioSystem::startOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3565,7 +3545,7 @@ status_t AudioFlinger::PlaybackThread::Track::start()
void AudioFlinger::PlaybackThread::Track::stop()
{
- ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3581,9 +3561,7 @@ void AudioFlinger::PlaybackThread::Track::stop()
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3594,7 +3572,7 @@ void AudioFlinger::PlaybackThread::Track::stop()
void AudioFlinger::PlaybackThread::Track::pause()
{
- ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3603,9 +3581,7 @@ void AudioFlinger::PlaybackThread::Track::pause()
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -5885,13 +5861,13 @@ void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast) {
+ bool unpinIfLast) {
Mutex::Autolock _l(mLock);
ALOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
- if (!effect->isPinned() || unpiniflast) {
+ if (!effect->isPinned() || unpinIfLast) {
removeEffect_l(effect);
AudioSystem::unregisterEffect(effect->id());
}
@@ -6207,16 +6183,16 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
return mHandles.size() != 0 ? mHandles[0].promote() : 0;
}
-void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast)
{
- ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
+ ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- thread->disconnectEffect(keep, handle, unpiniflast);
+ thread->disconnectEffect(keep, handle, unpinIfLast);
}
}
}
@@ -6888,13 +6864,13 @@ void AudioFlinger::EffectHandle::disconnect()
disconnect(true);
}
-void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
+void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
{
- ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
+ ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
if (mEffect == 0) {
return;
}
- mEffect->disconnect(this, unpiniflast);
+ mEffect->disconnect(this, unpinIfLast);
if (mHasControl && mEnabled) {
sp<ThreadBase> thread = mEffect->thread().promote();
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 97103c4..45df893 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -67,7 +67,7 @@ public:
virtual status_t dump(int fd, const Vector<String16>& args);
- // IAudioFlinger interface
+ // IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
pid_t pid,
audio_stream_type_t streamType,
@@ -81,6 +81,17 @@ public:
int *sessionId,
status_t *status);
+ virtual sp<IAudioRecord> openRecord(
+ pid_t pid,
+ audio_io_handle_t input,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ uint32_t flags,
+ int *sessionId,
+ status_t *status);
+
virtual uint32_t sampleRate(audio_io_handle_t output) const;
virtual int channelCount(audio_io_handle_t output) const;
virtual audio_format_t format(audio_io_handle_t output) const;
@@ -112,7 +123,6 @@ public:
virtual void registerClient(const sp<IAudioFlingerClient>& client);
virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
- virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
@@ -145,6 +155,8 @@ public:
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const;
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
+
virtual int newAudioSessionId();
virtual void acquireAudioSessionId(int audioSession);
@@ -171,49 +183,19 @@ public:
virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput);
- enum hardware_call_state {
- AUDIO_HW_IDLE = 0,
- AUDIO_HW_INIT,
- AUDIO_HW_OUTPUT_OPEN,
- AUDIO_HW_OUTPUT_CLOSE,
- AUDIO_HW_INPUT_OPEN,
- AUDIO_HW_INPUT_CLOSE,
- AUDIO_HW_STANDBY,
- AUDIO_HW_SET_MASTER_VOLUME,
- AUDIO_HW_GET_ROUTING,
- AUDIO_HW_SET_ROUTING,
- AUDIO_HW_GET_MODE,
- AUDIO_HW_SET_MODE,
- AUDIO_HW_GET_MIC_MUTE,
- AUDIO_HW_SET_MIC_MUTE,
- AUDIO_SET_VOICE_VOLUME,
- AUDIO_SET_PARAMETER,
- };
-
- // record interface
- virtual sp<IAudioRecord> openRecord(
- pid_t pid,
- audio_io_handle_t input,
- uint32_t sampleRate,
- audio_format_t format,
- uint32_t channelMask,
- int frameCount,
- uint32_t flags,
- int *sessionId,
- status_t *status);
-
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
+ // end of IAudioFlinger interface
+
+private:
audio_mode_t getMode() const { return mMode; }
bool btNrecIsOff() const { return mBtNrecIsOff; }
-private:
-
AudioFlinger();
virtual ~AudioFlinger();
@@ -442,7 +424,7 @@ private:
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast);
+ bool unpinIfLast);
// return values for hasAudioSession (bit field)
enum effect_state {
@@ -564,6 +546,18 @@ private:
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
};
+ struct stream_type_t {
+ stream_type_t()
+ : volume(1.0f),
+ mute(false),
+ valid(true)
+ {
+ }
+ float volume;
+ bool mute;
+ bool valid;
+ };
+
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
@@ -600,7 +594,7 @@ private:
return mName;
}
- audio_stream_type_t type() const {
+ audio_stream_type_t streamType() const {
return mStreamType;
}
status_t attachAuxEffect(int EffectId);
@@ -758,18 +752,6 @@ private:
void setStreamValid(audio_stream_type_t streamType, bool valid);
- struct stream_type_t {
- stream_type_t()
- : volume(1.0f),
- mute(false),
- valid(true)
- {
- }
- float volume;
- bool mute;
- bool valid;
- };
-
protected:
int16_t* mMixBuffer;
int mSuspended;
@@ -1120,7 +1102,7 @@ private:
const wp<ThreadBase>& thread() { return mThread; }
status_t addHandle(const sp<EffectHandle>& handle);
- void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
+ void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
size_t removeHandle (const wp<EffectHandle>& handle);
effect_descriptor_t& desc() { return mDescriptor; }
@@ -1196,7 +1178,9 @@ mutable Mutex mLock; // mutex for process, commands and handl
uint32_t *replySize,
void *pReplyData);
virtual void disconnect();
- virtual void disconnect(bool unpiniflast);
+ private:
+ void disconnect(bool unpinIfLast);
+ public:
virtual sp<IMemory> getCblk() const { return mCblkMemory; }
virtual status_t onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags);
@@ -1408,11 +1392,31 @@ mutable Mutex mLock; // mutex for process, commands and handl
mutable Mutex mHardwareLock;
audio_hw_device_t* mPrimaryHardwareDev;
Vector<audio_hw_device_t*> mAudioHwDevs;
+
+ enum hardware_call_state {
+ AUDIO_HW_IDLE = 0,
+ AUDIO_HW_INIT,
+ AUDIO_HW_OUTPUT_OPEN,
+ AUDIO_HW_OUTPUT_CLOSE,
+ AUDIO_HW_INPUT_OPEN,
+ AUDIO_HW_INPUT_CLOSE,
+ AUDIO_HW_STANDBY,
+ AUDIO_HW_SET_MASTER_VOLUME,
+ AUDIO_HW_GET_ROUTING,
+ AUDIO_HW_SET_ROUTING,
+ AUDIO_HW_GET_MODE,
+ AUDIO_HW_SET_MODE,
+ AUDIO_HW_GET_MIC_MUTE,
+ AUDIO_HW_SET_MIC_MUTE,
+ AUDIO_SET_VOICE_VOLUME,
+ AUDIO_SET_PARAMETER,
+ };
+
mutable hardware_call_state mHardwareStatus; // for dump only
DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
- PlaybackThread::stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+ stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
// both are protected by mLock
float mMasterVolume;
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 10efd85..f8b430e 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -542,7 +542,7 @@ status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
}
void AudioPolicyService::binderDied(const wp<IBinder>& who) {
- ALOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(),
+ ALOGW("binderDied() %p, tid %d, calling pid %d", who.unsafe_get(), gettid(),
IPCThreadState::self()->getCallingPid());
}
@@ -588,10 +588,10 @@ status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
}
dumpInternals(fd);
- if (mAudioCommandThread != NULL) {
+ if (mAudioCommandThread != 0) {
mAudioCommandThread->dump(fd);
}
- if (mTonePlaybackThread != NULL) {
+ if (mTonePlaybackThread != 0) {
mTonePlaybackThread->dump(fd);
}
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 0a52939..9486b9c 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -188,7 +188,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
@@ -201,7 +201,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
goto resampleStereo16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -215,7 +215,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -224,7 +224,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -246,7 +246,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -263,7 +263,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleStereo16_exit:
// save state
@@ -284,7 +284,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
@@ -296,7 +296,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -308,7 +308,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
int32_t sample = Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
@@ -318,7 +318,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -341,7 +341,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -357,7 +357,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleMono16_exit:
// save state
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index ee171ff..c23016e 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -72,9 +72,9 @@ protected:
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
- int32_t mBitDepth;
- int32_t mChannelCount;
- int32_t mSampleRate;
+ const int32_t mBitDepth;
+ const int32_t mChannelCount;
+ const int32_t mSampleRate;
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 47205ba..c0e760e 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -99,7 +99,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
in = mBuffer.i16;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
// advance sample state
@@ -133,7 +133,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL)
return;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index d012433..7a27b81 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -199,33 +199,32 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- AudioBufferProvider::Buffer& buffer(mBuffer);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
- while (buffer.frameCount == 0) {
- buffer.frameCount = inFrameCount;
- provider->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL) {
goto resample_exit;
}
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
if (phaseIndex == 1) {
// read one frame
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} else if (phaseIndex == 2) {
// read 2 frames
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
inputIndex++;
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
} else {
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
}
}
}
- int16_t *in = buffer.i16;
- const size_t frameCount = buffer.frameCount;
+ int16_t *in = mBuffer.i16;
+ const size_t frameCount = mBuffer.frameCount;
// Always read-in the first samples from the input buffer
int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +263,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
// if done with buffer, save samples
if (inputIndex >= frameCount) {
inputIndex -= frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
}
}
generated by cgit v1.1 at 2024-12-23 23:21:07 +0000