From 90bebef5669a9385c706b042d146a31dca2e5d9b Mon Sep 17 00:00:00 2001 From: Glenn Kasten Date: Fri, 27 Jan 2012 15:24:38 -0800 Subject: No newline or space at end of ALOG format string Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32 --- services/audioflinger/AudioFlinger.cpp | 18 +++++++++--------- services/audioflinger/AudioResampler.cpp | 24 ++++++++++++------------ services/audioflinger/AudioResamplerCubic.cpp | 4 ++-- 3 files changed, 23 insertions(+), 23 deletions(-) (limited to 'services') 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::EffectModule::controlHandle() void AudioFlinger::EffectModule::disconnect(const wp& 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 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; -- cgit v1.1