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-rw-r--r--media/libmedia/AudioEffect.cpp2
-rw-r--r--media/libmedia/MediaProfiles.cpp2
-rw-r--r--media/libmedia/MediaScanner.cpp2
-rw-r--r--media/libmedia/MediaScannerClient.cpp6
-rw-r--r--media/libmedia/ToneGenerator.cpp58
-rw-r--r--services/audioflinger/AudioFlinger.cpp18
-rw-r--r--services/audioflinger/AudioResampler.cpp24
-rw-r--r--services/audioflinger/AudioResamplerCubic.cpp4
8 files changed, 58 insertions, 58 deletions
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 6549ce6..f9f997f 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -159,7 +159,7 @@ status_t AudioEffect::set(const effect_uuid_t *type,
mCblk->buffer = (uint8_t *)mCblk + bufOffset;
iEffect->asBinder()->linkToDeath(mIEffectClient);
- ALOGV("set() %p OK effect: %s id: %d status %d enabled %d, ", this, mDescriptor.name, mId, mStatus, mEnabled);
+ ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled);
return mStatus;
}
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index c905762..bff251b 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -349,7 +349,7 @@ void MediaProfiles::addImageEncodingQualityLevel(int cameraId, const char** atts
{
CHECK(!strcmp("quality", atts[0]));
int quality = atoi(atts[1]);
- ALOGV("%s: cameraId=%d, quality=%d\n", __func__, cameraId, quality);
+ ALOGV("%s: cameraId=%d, quality=%d", __func__, cameraId, quality);
ImageEncodingQualityLevels *levels = findImageEncodingQualityLevels(cameraId);
if (levels == NULL) {
diff --git a/media/libmedia/MediaScanner.cpp b/media/libmedia/MediaScanner.cpp
index 79cab74..73d4519 100644
--- a/media/libmedia/MediaScanner.cpp
+++ b/media/libmedia/MediaScanner.cpp
@@ -143,7 +143,7 @@ MediaScanResult MediaScanner::doProcessDirectory(
if (pathRemaining >= 8 /* strlen(".nomedia") */ ) {
strcpy(fileSpot, ".nomedia");
if (access(path, F_OK) == 0) {
- ALOGV("found .nomedia, setting noMedia flag\n");
+ ALOGV("found .nomedia, setting noMedia flag");
noMedia = true;
}
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 9fe1820..cdfd477 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -142,12 +142,12 @@ void MediaScannerClient::convertValues(uint32_t encoding)
UConverter *conv = ucnv_open(enc, &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for %s\n", enc);
+ ALOGE("could not create UConverter for %s", enc);
return;
}
UConverter *utf8Conv = ucnv_open("UTF-8", &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for UTF-8\n");
+ ALOGE("could not create UConverter for UTF-8");
ucnv_close(conv);
return;
}
@@ -181,7 +181,7 @@ void MediaScannerClient::convertValues(uint32_t encoding)
ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
&source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
if (U_FAILURE(status)) {
- ALOGE("ucnv_convertEx failed: %d\n", status);
+ ALOGE("ucnv_convertEx failed: %d", status);
mValues->setEntry(i, "???");
} else {
// zero terminate
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index e6e989d..dfa41c4 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -800,7 +800,7 @@ const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava) {
- ALOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
+ ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume);
mState = TONE_IDLE;
@@ -829,9 +829,9 @@ ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool
}
if (initAudioTrack()) {
- ALOGV("ToneGenerator INIT OK, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("ToneGenerator INIT OK, time: %d", (unsigned int)(systemTime()/1000000));
} else {
- ALOGV("!!!ToneGenerator INIT FAILED!!!\n");
+ ALOGV("!!!ToneGenerator INIT FAILED!!!");
}
}
@@ -853,11 +853,11 @@ ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool
//
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::~ToneGenerator() {
- ALOGV("ToneGenerator destructor\n");
+ ALOGV("ToneGenerator destructor");
if (mpAudioTrack != NULL) {
stopTone();
- ALOGV("Delete Track: %p\n", mpAudioTrack);
+ ALOGV("Delete Track: %p", mpAudioTrack);
delete mpAudioTrack;
}
}
@@ -892,7 +892,7 @@ bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
}
}
- ALOGV("startTone\n");
+ ALOGV("startTone");
mLock.lock();
@@ -915,7 +915,7 @@ bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
if (mState == TONE_INIT) {
if (prepareWave()) {
- ALOGV("Immediate start, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("Immediate start, time %d", (unsigned int)(systemTime()/1000000));
lResult = true;
mState = TONE_STARTING;
mLock.unlock();
@@ -934,7 +934,7 @@ bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
mState = TONE_IDLE;
}
} else {
- ALOGV("Delayed start\n");
+ ALOGV("Delayed start");
mState = TONE_RESTARTING;
lStatus = mWaitCbkCond.waitRelative(mLock, seconds(3));
if (lStatus == NO_ERROR) {
@@ -949,8 +949,8 @@ bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
}
mLock.unlock();
- ALOGV_IF(lResult, "Tone started, time %d\n", (unsigned int)(systemTime()/1000000));
- ALOGW_IF(!lResult, "Tone start failed!!!, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV_IF(lResult, "Tone started, time %d", (unsigned int)(systemTime()/1000000));
+ ALOGW_IF(!lResult, "Tone start failed!!!, time %d", (unsigned int)(systemTime()/1000000));
return lResult;
}
@@ -1017,7 +1017,7 @@ bool ToneGenerator::initAudioTrack() {
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- ALOGV("Create Track: %p\n", mpAudioTrack);
+ ALOGV("Create Track: %p", mpAudioTrack);
mpAudioTrack->set(mStreamType,
0,
@@ -1046,7 +1046,7 @@ initAudioTrack_exit:
// Cleanup
if (mpAudioTrack) {
- ALOGV("Delete Track I: %p\n", mpAudioTrack);
+ ALOGV("Delete Track I: %p", mpAudioTrack);
delete mpAudioTrack;
mpAudioTrack = NULL;
}
@@ -1141,7 +1141,7 @@ void ToneGenerator::audioCallback(int event, void* user, void *info) {
if (lpToneGen->mTotalSmp > lpToneGen->mNextSegSmp) {
// Time to go to next sequence segment
- ALOGV("End Segment, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End Segment, time: %d", (unsigned int)(systemTime()/1000000));
lGenSmp = lReqSmp;
@@ -1156,13 +1156,13 @@ void ToneGenerator::audioCallback(int event, void* user, void *info) {
lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd);
lFrequency = lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[++lFreqIdx];
}
- ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
}
// check if we need to loop and loop for the reqd times
if (lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
if (lpToneGen->mLoopCounter < lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
- ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1172,14 +1172,14 @@ void ToneGenerator::audioCallback(int event, void* user, void *info) {
// completed loop. go to next segment
lpToneGen->mLoopCounter = 0;
lpToneGen->mCurSegment++;
- ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
}
} else {
lpToneGen->mCurSegment++;
- ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1188,32 +1188,32 @@ void ToneGenerator::audioCallback(int event, void* user, void *info) {
// Handle loop if last segment reached
if (lpToneDesc->segments[lpToneGen->mCurSegment].duration == 0) {
- ALOGV("Last Seg: %d\n", lpToneGen->mCurSegment);
+ ALOGV("Last Seg: %d", lpToneGen->mCurSegment);
// Pre increment loop count and restart if total count not reached. Stop sequence otherwise
if (++lpToneGen->mCurCount <= lpToneDesc->repeatCnt) {
- ALOGV("Repeating Count: %d\n", lpToneGen->mCurCount);
+ ALOGV("Repeating Count: %d", lpToneGen->mCurCount);
lpToneGen->mCurSegment = lpToneDesc->repeatSegment;
if (lpToneDesc->segments[lpToneDesc->repeatSegment].waveFreq[0] != 0) {
lWaveCmd = WaveGenerator::WAVEGEN_START;
}
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
} else {
lGenSmp = 0;
- ALOGV("End repeat, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End repeat, time: %d", (unsigned int)(systemTime()/1000000));
}
} else {
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
if (lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[0] != 0) {
// If next segment is not silent, OFF -> ON transition : reset wave generator
lWaveCmd = WaveGenerator::WAVEGEN_START;
- ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
} else {
lGenSmp = 0;
}
@@ -1251,13 +1251,13 @@ audioCallback_EndLoop:
switch (lpToneGen->mState) {
case TONE_RESTARTING:
- ALOGV("Cbk restarting track\n");
+ ALOGV("Cbk restarting track");
if (lpToneGen->prepareWave()) {
lpToneGen->mState = TONE_STARTING;
// must reload lpToneDesc as prepareWave() may change mpToneDesc
lpToneDesc = lpToneGen->mpToneDesc;
} else {
- ALOGW("Cbk restarting prepareWave() failed\n");
+ ALOGW("Cbk restarting prepareWave() failed");
lpToneGen->mState = TONE_IDLE;
lpToneGen->mpAudioTrack->stop();
// Force loop exit
@@ -1266,14 +1266,14 @@ audioCallback_EndLoop:
lSignal = true;
break;
case TONE_STOPPING:
- ALOGV("Cbk Stopping\n");
+ ALOGV("Cbk Stopping");
lpToneGen->mState = TONE_STOPPED;
// Force loop exit
lNumSmp = 0;
break;
case TONE_STOPPED:
lpToneGen->mState = TONE_INIT;
- ALOGV("Cbk Stopped track\n");
+ ALOGV("Cbk Stopped track");
lpToneGen->mpAudioTrack->stop();
// Force loop exit
lNumSmp = 0;
@@ -1281,7 +1281,7 @@ audioCallback_EndLoop:
lSignal = true;
break;
case TONE_STARTING:
- ALOGV("Cbk starting track\n");
+ ALOGV("Cbk starting track");
lpToneGen->mState = TONE_PLAYING;
lSignal = true;
break;
@@ -1491,7 +1491,7 @@ ToneGenerator::WaveGenerator::WaveGenerator(unsigned short samplingRate,
d0 = 32767;
mA1_Q14 = (short) d0;
- ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d\n",
+ ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d",
mA1_Q14, mS2_0, mAmplitude_Q15);
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 93c91fb..d5d1b6c 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1920,7 +1920,7 @@ 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 +1934,9 @@ 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;
@@ -2638,7 +2638,7 @@ 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 +2651,9 @@ 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) {
@@ -3046,9 +3046,9 @@ 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;
@@ -6209,7 +6209,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
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);
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 6e17a4a..4eac032 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -184,7 +184,7 @@ 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) {
@@ -197,7 +197,7 @@ 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;
@@ -211,7 +211,7 @@ 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);
@@ -220,7 +220,7 @@ 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) {
@@ -242,7 +242,7 @@ 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) {
@@ -259,7 +259,7 @@ 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
@@ -280,7 +280,7 @@ 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
@@ -292,7 +292,7 @@ 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;
@@ -304,7 +304,7 @@ 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;
@@ -314,7 +314,7 @@ 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) {
@@ -337,7 +337,7 @@ 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) {
@@ -353,7 +353,7 @@ 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/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;
generated by cgit v1.1 at 2024-05-14 23:20:00 +0000