Return number of frames output from resample method
Change-Id: Ic297e2ed59839f1788c83e099ef1a9e4af29591f
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 46e3d6c..e49b7b1 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -41,7 +41,7 @@
AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
}
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
// number of bits used in interpolation multiply - 15 bits avoids overflow
@@ -51,9 +51,9 @@
static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
void init() {}
- void resampleMono16(int32_t* out, size_t outFrameCount,
+ size_t resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
- void resampleStereo16(int32_t* out, size_t outFrameCount,
+ size_t resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
@@ -329,7 +329,7 @@
// ----------------------------------------------------------------------------
-void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
// should never happen, but we overflow if it does
@@ -338,15 +338,16 @@
// 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 AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -442,9 +443,10 @@
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex / 2 /* channels for stereo */;
}
-void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
@@ -538,6 +540,7 @@
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex;
}
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 863614a..a8e3e6f 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -67,12 +67,18 @@
// Resample int16_t samples from provider and accumulate into 'out'.
// A mono provider delivers a sequence of samples.
// A stereo provider delivers a sequence of interleaved pairs of samples.
- // Multi-channel providers are not supported.
+ //
// In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
// That is, for a mono provider, there is an implicit up-channeling.
// Since this method accumulates, the caller is responsible for clearing 'out' initially.
- // FIXME assumes provider is always successful; it should return the actual frame count.
- virtual void resample(int32_t* out, size_t outFrameCount,
+ //
+ // For a float resampler, 'out' holds interleaved pairs of float samples.
+ //
+ // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
+ // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
+ //
+ // Returns the number of frames resampled into the out buffer.
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
virtual void reset();
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 @@
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 @@
// 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 @@
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 @@
// 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 @@
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 @@
// ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex;
}
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 1ddc5f9..4b45b0b 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -31,7 +31,7 @@
AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
}
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
// number of bits used in interpolation multiply - 14 bits avoids overflow
@@ -43,9 +43,9 @@
int32_t a, b, c, y0, y1, y2, y3;
} state;
void init();
- void resampleMono16(int32_t* out, size_t outFrameCount,
+ size_t resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
- void resampleStereo16(int32_t* out, size_t outFrameCount,
+ size_t resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
static inline int32_t interp(state* p, int32_t x) {
return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1;
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index c21d4ca..6481b85 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -477,15 +477,15 @@
}
template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
- (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
+ return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
}
template<typename TC, typename TI, typename TO>
template<int CHANNELS, bool LOCKED, int STRIDE>
-void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
// TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
@@ -610,6 +610,7 @@
ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
mInBuffer.setImpulse(impulse);
mPhaseFraction = phaseFraction;
+ return outputIndex / OUTPUT_CHANNELS;
}
/* instantiate templates used by AudioResampler::create */
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 238b163..3b1c381 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -52,7 +52,7 @@
virtual void setVolume(float left, float right);
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
@@ -111,10 +111,10 @@
int inSampleRate, int outSampleRate, double tbwCheat);
template<int CHANNELS, bool LOCKED, int STRIDE>
- void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+ size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
// define a pointer to member function type for resample
- typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+ typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
size_t outFrameCount, AudioBufferProvider* provider);
// data - the contiguous storage and layout of these is important.
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index ba9a356..41730ee 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -256,7 +256,7 @@
mVolumeSIMD[1] = u4_28_from_float(clampFloatVol(right));
}
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
// FIXME store current state (up or down sample) and only load the coefs when the state
@@ -272,17 +272,18 @@
// select the appropriate resampler
switch (mChannelCount) {
case 1:
- resample<1>(out, outFrameCount, provider);
- break;
+ return resample<1>(out, outFrameCount, provider);
case 2:
- resample<2>(out, outFrameCount, provider);
- break;
+ return resample<2>(out, outFrameCount, provider);
+ default:
+ LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+ return 0;
}
}
template<int CHANNELS>
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
const Constants& c(*mConstants);
@@ -357,6 +358,7 @@
mImpulse = impulse;
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
+ return outputIndex / CHANNELS;
}
template<int CHANNELS>
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 6d8e85d..0fbeac8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -39,7 +39,7 @@
virtual ~AudioResamplerSinc();
- virtual void resample(int32_t* out, size_t outFrameCount,
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
void init();
@@ -47,7 +47,7 @@
virtual void setVolume(float left, float right);
template<int CHANNELS>
- void resample(int32_t* out, size_t outFrameCount,
+ size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
template<int CHANNELS>
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index d6217ba..9e375db 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -48,7 +48,10 @@
if (thisFrames == 0 || thisFrames > outputFrames - i) {
thisFrames = outputFrames - i;
}
- resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
+ size_t framesResampled = resampler->resample(
+ (int32_t*) output + channels*i, thisFrames, provider);
+ // we should have enough buffer space, so there is no short count.
+ ASSERT_EQ(thisFrames, framesResampled);
i += thisFrames;
}
}
