diff options
author | Glenn Kasten <gkasten@google.com> | 2012-01-27 15:24:38 -0800 |
---|---|---|
committer | Glenn Kasten <gkasten@google.com> | 2012-02-08 17:21:49 -0800 |
commit | 90bebef5669a9385c706b042d146a31dca2e5d9b (patch) | |
tree | a60c6383825eb3ed02493036605391d015732190 | |
parent | 98ec94c5854daccc3474758524e7f4adfe535ce0 (diff) | |
download | frameworks_av-90bebef5669a9385c706b042d146a31dca2e5d9b.zip frameworks_av-90bebef5669a9385c706b042d146a31dca2e5d9b.tar.gz frameworks_av-90bebef5669a9385c706b042d146a31dca2e5d9b.tar.bz2 |
No newline or space at end of ALOG format string
Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32
-rw-r--r-- | media/libmedia/AudioEffect.cpp | 2 | ||||
-rw-r--r-- | media/libmedia/MediaProfiles.cpp | 2 | ||||
-rw-r--r-- | media/libmedia/MediaScanner.cpp | 2 | ||||
-rw-r--r-- | media/libmedia/MediaScannerClient.cpp | 6 | ||||
-rw-r--r-- | media/libmedia/ToneGenerator.cpp | 58 | ||||
-rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 18 | ||||
-rw-r--r-- | services/audioflinger/AudioResampler.cpp | 24 | ||||
-rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 4 |
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; |