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author | Glenn Kasten <gkasten@google.com> | 2012-11-01 11:11:38 -0700 |
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committer | Glenn Kasten <gkasten@google.com> | 2012-11-01 12:19:25 -0700 |
commit | 85ab62c4b433df3f1a9826bed1c9bec07a86c750 (patch) | |
tree | 0c5443a20824924cb7403f4cabfee84062489793 /services/audioflinger/AudioMixer.cpp | |
parent | 5fe6138bd839297a1eed16885102b3bdfc98c040 (diff) | |
download | frameworks_av-85ab62c4b433df3f1a9826bed1c9bec07a86c750.zip frameworks_av-85ab62c4b433df3f1a9826bed1c9bec07a86c750.tar.gz frameworks_av-85ab62c4b433df3f1a9826bed1c9bec07a86c750.tar.bz2 |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
Diffstat (limited to 'services/audioflinger/AudioMixer.cpp')
-rw-r--r-- | services/audioflinger/AudioMixer.cpp | 30 |
1 files changed, 20 insertions, 10 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index a4ed445..b3ca877 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -765,7 +765,8 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) } -void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, + int32_t* temp, int32_t* aux) { t->resampler->setSampleRate(t->sampleRate); @@ -798,11 +799,13 @@ void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFram } } -void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, + int32_t* aux) { } -void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { int32_t vl = t->prevVolume[0]; int32_t vr = t->prevVolume[1]; @@ -844,7 +847,8 @@ void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, i t->adjustVolumeRamp(aux != NULL); } -void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t vl = t->volume[0]; const int16_t vr = t->volume[1]; @@ -872,7 +876,8 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32 } } -void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<const int16_t *>(t->in); @@ -962,7 +967,8 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount t->in = in; } -void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<int16_t const *>(t->in); @@ -1147,7 +1153,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) while (outFrames) { size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; if (inFrames) { - t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); + t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, + state->resampleTemp, aux); t.frameCount -= inFrames; outFrames -= inFrames; if (CC_UNLIKELY(aux != NULL)) { @@ -1156,7 +1163,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) } if (t.frameCount == 0 && outFrames) { t.bufferProvider->releaseBuffer(&t.buffer); - t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames); + t.buffer.frameCount = (state->frameCount - numFrames) - + (BLOCKSIZE - outFrames); int64_t outputPTS = calculateOutputPTS( t, pts, numFrames + (BLOCKSIZE - outFrames)); t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); @@ -1246,7 +1254,8 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) if (CC_UNLIKELY(aux != NULL)) { aux += outFrames; } - t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux); + t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, + state->resampleTemp, aux); outFrames += t.buffer.frameCount; t.bufferProvider->releaseBuffer(&t.buffer); } @@ -1286,7 +1295,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, // been enabled for mixing. if (in == NULL || ((unsigned long)in & 3)) { memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x", + ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " + "buffer %p track %d, channels %d, needs %08x", in, i, t.channelCount, t.needs); return; } |