Return number of frames output from resample method (6b3b7e30) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioResampler.cpp

+13 −10

Original line number	Diff line number	Diff line
		@@ -41,7 +41,7 @@ public:
		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 @@ private:
		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 AudioResampler::reset() {

		// ----------------------------------------------------------------------------

		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 @@ void AudioResamplerOrder1::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 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 @@ resampleStereo16_exit:
		// 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 @@ resampleMono16_exit:
		// save state
		mInputIndex = inputIndex;
		mPhaseFraction = phaseFraction;
		return outputIndex;
		}

		#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1

services/audioflinger/AudioResampler.h

+9 −3

Original line number	Diff line number	Diff line
		@@ -67,12 +67,18 @@ public:
		// 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();

services/audioflinger/AudioResamplerCubic.cpp

+13 −10

Original line number	Diff line number	Diff line
		@@ -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;
		}

		// ----------------------------------------------------------------------------

services/audioflinger/AudioResamplerCubic.h

+3 −3

Original line number	Diff line number	Diff line
		@@ -31,7 +31,7 @@ public:
		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 @@ private:
		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;

services/audioflinger/AudioResamplerDyn.cpp

+4 −3

Original line number	Diff line number	Diff line
		@@ -477,15 +477,15 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate)
		}

		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 @@ resample_exit:
		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 */