diff options
| author | Andy Hung <hunga@google.com> | 2015-03-29 00:49:22 -0700 |
|---|---|---|
| committer | Andy Hung <hunga@google.com> | 2015-04-08 14:31:58 -0700 |
| commit | 6b3b7e304e0f8f167241b2c75f1eb04a9ef192ec (patch) | |
| tree | 1cec011ad26676dc9dc3eea778e18136c083e04f /services/audioflinger/AudioResamplerCubic.cpp | |
| parent | 25f82752942b1c78aec8ee303d61afff85cff9d1 (diff) | |
| download | frameworks_av-6b3b7e304e0f8f167241b2c75f1eb04a9ef192ec.zip frameworks_av-6b3b7e304e0f8f167241b2c75f1eb04a9ef192ec.tar.gz frameworks_av-6b3b7e304e0f8f167241b2c75f1eb04a9ef192ec.tar.bz2 | |
Return number of frames output from resample method
Change-Id: Ic297e2ed59839f1788c83e099ef1a9e4af29591f
Diffstat (limited to 'services/audioflinger/AudioResamplerCubic.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 23 |
1 files changed, 13 insertions, 10 deletions
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index d3cbd1c..172c2a5 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -14,7 +14,7 @@ * limitations under the License. */ -#define LOG_TAG "AudioSRC" +#define LOG_TAG "AudioResamplerCubic" #include <stdint.h> #include <string.h> @@ -32,7 +32,7 @@ void AudioResamplerCubic::init() { memset(&right, 0, sizeof(state)); } -void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // should never happen, but we overflow if it does @@ -41,15 +41,16 @@ void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, // select the appropriate resampler switch (mChannelCount) { case 1: - resampleMono16(out, outFrameCount, provider); - break; + return resampleMono16(out, outFrameCount, provider); case 2: - resampleStereo16(out, outFrameCount, provider); - break; + return resampleStereo16(out, outFrameCount, provider); + default: + LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount); + return 0; } } -void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -67,7 +68,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) { - return; + return 0; } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); } @@ -115,9 +116,10 @@ save_state: // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex / 2 /* channels for stereo */; } -void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, +size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { int32_t vl = mVolume[0]; @@ -135,7 +137,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) { - return; + return 0; } // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } @@ -182,6 +184,7 @@ save_state: // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; + return outputIndex; } // ---------------------------------------------------------------------------- |