Merge "Replace int bitDepth with audio_format_t in Resampler" (3d638b30) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioMixer.cpp

+1 −13

Original line number	Diff line number	Diff line
		@@ -763,21 +763,9 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
		quality = AudioResampler::DEFAULT_QUALITY;
		}

		int bits;
		switch (mMixerInFormat) {
		case AUDIO_FORMAT_PCM_16_BIT:
		bits = 16;
		break;
		case AUDIO_FORMAT_PCM_FLOAT:
		bits = 32; // 32 bits to the AudioResampler::create() indicates float.
		break;
		default:
		LOG_ALWAYS_FATAL("Invalid mMixerInFormat: %#x", mMixerInFormat);
		break;
		}
		ALOGVV("Creating resampler with %d bits\n", bits);
		resampler = AudioResampler::create(
		bits,
		mMixerInFormat,
		// the resampler sees the number of channels after the downmixer, if any
		(int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
		devSampleRate, quality);

services/audioflinger/AudioResampler.cpp

+30 −25

Original line number	Diff line number	Diff line
		@@ -40,8 +40,8 @@ namespace android {

		class AudioResamplerOrder1 : public AudioResampler {
		public:
		AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
		AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
		AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
		AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
		}
		virtual void resample(int32_t* out, size_t outFrameCount,
		AudioBufferProvider* provider);
		@@ -145,7 +145,7 @@ static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, s
		static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
		static uint32_t currentMHz = 0;

		AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
		AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount,
		int32_t sampleRate, src_quality quality) {

		bool atFinalQuality;
		@@ -216,34 +216,41 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
		default:
		case LOW_QUALITY:
		ALOGV("Create linear Resampler");
		resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
		LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
		resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
		break;
		case MED_QUALITY:
		ALOGV("Create cubic Resampler");
		resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
		LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
		resampler = new AudioResamplerCubic(inChannelCount, sampleRate);
		break;
		case HIGH_QUALITY:
		ALOGV("Create HIGH_QUALITY sinc Resampler");
		resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
		LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
		resampler = new AudioResamplerSinc(inChannelCount, sampleRate);
		break;
		case VERY_HIGH_QUALITY:
		ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
		resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
		LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
		resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);
		break;
		case DYN_LOW_QUALITY:
		case DYN_MED_QUALITY:
		case DYN_HIGH_QUALITY:
		ALOGV("Create dynamic Resampler = %d", quality);
		if (bitDepth == 32) { /* bitDepth == 32 signals float precision */
		resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,
		if (format == AUDIO_FORMAT_PCM_FLOAT) {
		resampler = new AudioResamplerDyn<float, float, float>(inChannelCount,
		sampleRate, quality);
		} else if (quality == DYN_HIGH_QUALITY) {
		resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,
		} else {
		LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
		if (quality == DYN_HIGH_QUALITY) {
		resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount,
		sampleRate, quality);
		} else {
		resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,
		resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount,
		sampleRate, quality);
		}
		}
		break;
		}

		@@ -252,18 +259,17 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
		return resampler;
		}

		AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
		AudioResampler::AudioResampler(int inChannelCount,
		int32_t sampleRate, src_quality quality) :
		mBitDepth(bitDepth), mChannelCount(inChannelCount),
		mChannelCount(inChannelCount),
		mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
		mPhaseFraction(0), mLocalTimeFreq(0),
		mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
		// sanity check on format
		if ((bitDepth != 16 && (quality < DYN_LOW_QUALITY \|\| bitDepth != 32))
		\|\| inChannelCount < 1

		if (inChannelCount < 1
		\|\| inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) {
		LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d bits, %d channels",
		quality, bitDepth, inChannelCount);
		LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
		quality, inChannelCount);
		}
		if (sampleRate <= 0) {
		LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
		@@ -272,7 +278,6 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
		// initialize common members
		mVolume[0] = mVolume[1] = 0;
		mBuffer.frameCount = 0;

		}

		AudioResampler::~AudioResampler() {

services/audioflinger/AudioResampler.h

+3 −3

Original line number	Diff line number	Diff line
		@@ -22,6 +22,7 @@
		#include <cutils/compiler.h>

		#include <media/AudioBufferProvider.h>
		#include <system/audio.h>

		namespace android {
		// ----------------------------------------------------------------------------
		@@ -46,7 +47,7 @@ public:
		DYN_HIGH_QUALITY=7,
		};

		static AudioResampler* create(int bitDepth, int inChannelCount,
		static AudioResampler* create(audio_format_t format, int inChannelCount,
		int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);

		virtual ~AudioResampler();
		@@ -86,7 +87,7 @@ protected:
		// multiplier to calculate fixed point phase increment
		static const double kPhaseMultiplier;

		AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
		AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);

		// prevent copying
		AudioResampler(const AudioResampler&);
		@@ -94,7 +95,6 @@ protected:

		int64_t calculateOutputPTS(int outputFrameIndex);

		const int32_t mBitDepth;
		const int32_t mChannelCount;
		const int32_t mSampleRate;
		int32_t mInSampleRate;

services/audioflinger/AudioResamplerCubic.h

+2 −2

Original line number	Diff line number	Diff line
		@@ -28,8 +28,8 @@ namespace android {

		class AudioResamplerCubic : public AudioResampler {
		public:
		AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :
		AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {
		AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
		AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
		}
		virtual void resample(int32_t* out, size_t outFrameCount,
		AudioBufferProvider* provider);

services/audioflinger/AudioResamplerDyn.cpp

+2 −2

Original line number	Diff line number	Diff line
		@@ -162,9 +162,9 @@ void AudioResamplerDyn<TC, TI, TO>::Constants::set(
		}

		template<typename TC, typename TI, typename TO>
		AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(int bitDepth,
		AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(
		int inChannelCount, int32_t sampleRate, src_quality quality)
		: AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
		: AudioResampler(inChannelCount, sampleRate, quality),
		mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
		mCoefBuffer(NULL)
		{