From 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 Mon Sep 17 00:00:00 2001 From: Steve Block Date: Fri, 6 Jan 2012 19:20:56 +0000 Subject: Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE See https://android-git.corp.google.com/g/#/c/157220 Bug: 5449033 Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c --- services/audioflinger/AudioResampler.cpp | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'services/audioflinger/AudioResampler.cpp') diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 7205045..fbdcb62 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -121,7 +121,7 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, mPhaseFraction(0) { // sanity check on format if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { - LOGE("Unsupported sample format, %d bits, %d channels", bitDepth, + ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth, inChannelCount); // LOG_ASSERT(0); } @@ -190,7 +190,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, size_t outputSampleCount = outFrameCount * 2; size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", + // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", // outFrameCount, inputIndex, phaseFraction, phaseIncrement); while (outputIndex < outputSampleCount) { @@ -203,7 +203,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, goto resampleStereo16_exit; } - // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); + // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); if (mBuffer.frameCount > inputIndex) break; inputIndex -= mBuffer.frameCount; @@ -217,7 +217,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, // handle boundary case while (inputIndex == 0) { - // LOGE("boundary case\n"); + // ALOGE("boundary case\n"); out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction); out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction); Advance(&inputIndex, &phaseFraction, phaseIncrement); @@ -226,7 +226,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, } // process input samples - // LOGE("general case\n"); + // ALOGE("general case\n"); #ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 if (inputIndex + 2 < mBuffer.frameCount) { @@ -248,13 +248,13 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, Advance(&inputIndex, &phaseFraction, phaseIncrement); } - // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); + // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); // if done with buffer, save samples if (inputIndex >= mBuffer.frameCount) { inputIndex -= mBuffer.frameCount; - // LOGE("buffer done, new input index %d", inputIndex); + // ALOGE("buffer done, new input index %d", inputIndex); mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; @@ -265,7 +265,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, } } - // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); + // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); resampleStereo16_exit: // save state @@ -286,7 +286,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, size_t outputSampleCount = outFrameCount * 2; size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", + // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", // outFrameCount, inputIndex, phaseFraction, phaseIncrement); while (outputIndex < outputSampleCount) { // buffer is empty, fetch a new one @@ -298,7 +298,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, mPhaseFraction = phaseFraction; goto resampleMono16_exit; } - // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); + // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); if (mBuffer.frameCount > inputIndex) break; inputIndex -= mBuffer.frameCount; @@ -310,7 +310,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, // handle boundary case while (inputIndex == 0) { - // LOGE("boundary case\n"); + // ALOGE("boundary case\n"); int32_t sample = Interp(mX0L, in[0], phaseFraction); out[outputIndex++] += vl * sample; out[outputIndex++] += vr * sample; @@ -320,7 +320,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, } // process input samples - // LOGE("general case\n"); + // ALOGE("general case\n"); #ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 if (inputIndex + 2 < mBuffer.frameCount) { @@ -343,13 +343,13 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, } - // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); + // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); // if done with buffer, save samples if (inputIndex >= mBuffer.frameCount) { inputIndex -= mBuffer.frameCount; - // LOGE("buffer done, new input index %d", inputIndex); + // ALOGE("buffer done, new input index %d", inputIndex); mX0L = mBuffer.i16[mBuffer.frameCount-1]; provider->releaseBuffer(&mBuffer); @@ -359,7 +359,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, } } - // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); + // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); resampleMono16_exit: // save state -- cgit v1.1