diff options
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 0a52939..9486b9c 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -188,7 +188,7 @@ 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) { @@ -201,7 +201,7 @@ 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; @@ -215,7 +215,7 @@ 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); @@ -224,7 +224,7 @@ 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) { @@ -246,7 +246,7 @@ 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) { @@ -263,7 +263,7 @@ 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 @@ -284,7 +284,7 @@ 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 @@ -296,7 +296,7 @@ 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; @@ -308,7 +308,7 @@ 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; @@ -318,7 +318,7 @@ 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) { @@ -341,7 +341,7 @@ 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) { @@ -357,7 +357,7 @@ 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 |