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| 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 /services/audioflinger/AudioResampler.cpp | |
| 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
Diffstat (limited to 'services/audioflinger/AudioResampler.cpp')
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 24 |
1 files changed, 12 insertions, 12 deletions
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 |