/frameworks/native/libs/ui/ |
D | FrameStats.cpp | 41 size_t frameCount = desiredPresentTimesNano.size(); in flatten() local 46 memcpy(timestamps, desiredPresentTimesNano.array(), frameCount * timestampSize); in flatten() 47 timestamps += frameCount; in flatten() 49 memcpy(timestamps, actualPresentTimesNano.array(), frameCount * timestampSize); in flatten() 50 timestamps += frameCount; in flatten() 52 memcpy(timestamps, frameReadyTimesNano.array(), frameCount * timestampSize); in flatten() 65 size_t frameCount = (size - timestampSize) / (3 * timestampSize); in unflatten() local 70 desiredPresentTimesNano.resize(frameCount); in unflatten() 71 memcpy(desiredPresentTimesNano.editArray(), timestamps, frameCount * timestampSize); in unflatten() 72 timestamps += frameCount; in unflatten() [all …]
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/frameworks/av/services/audioflinger/ |
D | FastCapture.cpp | 93 const size_t frameCount = current->mFrameCount; in onStateChange() local 129 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { in onStateChange() 133 if (frameCount > 0 && mSampleRate > 0) { in onStateChange() 137 size_t bufferSize = frameCount * Format_frameSize(mFormat); in onStateChange() 140 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 in onStateChange() 141 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 in onStateChange() 142 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 in onStateChange() 143 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95 in onStateChange() 144 mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75 in onStateChange() 145 mWarmupNsMax = (frameCount * 1250000000LL) / mSampleRate; // 1.25 in onStateChange() [all …]
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D | FastMixer.cpp | 212 const size_t frameCount = current->mFrameCount; in onStateChange() local 250 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { in onStateChange() 258 if (frameCount > 0 && mSampleRate > 0) { in onStateChange() 262 mMixer = new AudioMixer(frameCount, mSampleRate); in onStateChange() 265 params.frameCount = frameCount; in onStateChange() 270 mMixerBufferSize = mixerFrameSize * frameCount; in onStateChange() 275 mSinkBufferSize = sinkFrameSize * frameCount; in onStateChange() 278 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 in onStateChange() 279 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 in onStateChange() 280 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 in onStateChange() [all …]
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D | Tracks.cpp | 74 size_t frameCount, in TrackBase() argument 99 mFrameCount(frameCount), in TrackBase() 123 size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount; in TrackBase() 125 if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2 in TrackBase() 267 mTee.write(buffer->raw, buffer->frameCount); in releaseBuffer() 271 buf.mFrameCount = buffer->frameCount; in releaseBuffer() 273 buffer->frameCount = 0; in releaseBuffer() 293 uint64_t mixBufferNs = ((uint64_t)2 * thread.frameCount() * 1000000000) / in PatchTrackBase() 507 size_t frameCount, in Track() argument 518 : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount, in Track() [all …]
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/frameworks/av/media/libaudioprocessing/ |
D | BufferProviders.cpp | 57 mBuffer.frameCount = 0; in CopyBufferProvider() 63 __func__, this, mBuffer.frameCount, mTrackBufferProvider, mLocalBufferData); in ~CopyBufferProvider() 64 if (mBuffer.frameCount != 0) { in ~CopyBufferProvider() 77 copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount); in getNextBuffer() 81 if (mBuffer.frameCount == 0) { in getNextBuffer() 82 mBuffer.frameCount = pBuffer->frameCount; in getNextBuffer() 89 ALOG_ASSERT(res == OK || mBuffer.frameCount == 0); in getNextBuffer() 90 if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe. in getNextBuffer() 92 pBuffer->frameCount = 0; in getNextBuffer() 97 ALOG_ASSERT(mConsumed < mBuffer.frameCount); in getNextBuffer() [all …]
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D | AudioResampler.cpp | 283 mBuffer.frameCount = 0; in AudioResampler() 314 mBuffer.frameCount = 0; in reset() 356 while (mBuffer.frameCount == 0) { in resampleStereo16() 357 mBuffer.frameCount = inFrameCount; in resampleStereo16() 364 if (mBuffer.frameCount > inputIndex) break; in resampleStereo16() 366 inputIndex -= mBuffer.frameCount; in resampleStereo16() 367 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; in resampleStereo16() 368 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; in resampleStereo16() 390 if (inputIndex + 2 < mBuffer.frameCount) { in resampleStereo16() 395 maxInIdx = mBuffer.frameCount - 2; in resampleStereo16() [all …]
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D | AudioMixerBase.cpp | 775 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) in volumeRampStereo() argument 802 } while (--frameCount); in volumeRampStereo() 810 } while (--frameCount); in volumeRampStereo() 818 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) in volumeStereo() argument 834 } while (--frameCount); in volumeStereo() 842 } while (--frameCount); in volumeStereo() 847 int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) in track__16BitsStereo() argument 876 } while (--frameCount); in track__16BitsStereo() 897 } while (--frameCount); in track__16BitsStereo() 916 } while (--frameCount); in track__16BitsStereo() [all …]
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D | AudioResamplerCubic.cpp | 67 if (mBuffer.frameCount == 0) { in resampleStereo16() 68 mBuffer.frameCount = inFrameCount; in resampleStereo16() 95 if (inputIndex == mBuffer.frameCount) { in resampleStereo16() 98 mBuffer.frameCount = inFrameCount; in resampleStereo16() 134 if (mBuffer.frameCount == 0) { in resampleMono16() 135 mBuffer.frameCount = inFrameCount; in resampleMono16() 163 if (inputIndex == mBuffer.frameCount) { in resampleMono16() 166 mBuffer.frameCount = inFrameCount; in resampleMono16()
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D | AudioResamplerDyn.cpp | 659 ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%zu > frameCount%zu", in resample() 660 inputIndex, mBuffer.frameCount); in resample() 663 while (mBuffer.frameCount == 0 && inFrameCount > 0) { in resample() 664 mBuffer.frameCount = inFrameCount; in resample() 672 inFrameCount -= mBuffer.frameCount; in resample() 680 if (inputIndex >= mBuffer.frameCount) { in resample() 694 const size_t frameCount = mBuffer.frameCount; in resample() local 721 if (inputIndex >= frameCount) { in resample() 733 ALOG_ASSERT(inputIndex == frameCount, "inputIndex(%zu) != frameCount(%zu)", in resample() 734 inputIndex, frameCount); // must have been fully read. in resample() [all …]
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/frameworks/av/media/libnbaio/ |
D | SourceAudioBufferProvider.cpp | 50 ALOG_ASSERT(buffer != NULL && buffer->frameCount > 0 && mGetCount == 0); in getNextBuffer() 54 if (mRemaining < buffer->frameCount) { in getNextBuffer() 55 buffer->frameCount = mRemaining; in getNextBuffer() 58 mGetCount = buffer->frameCount; in getNextBuffer() 62 if (buffer->frameCount > mSize) { in getNextBuffer() 67 mAllocated = calloc(buffer->frameCount, mFrameSize); in getNextBuffer() 72 mSize = buffer->frameCount; in getNextBuffer() 76 ssize_t actual = mSource->read(mAllocated, buffer->frameCount); in getNextBuffer() 78 ALOG_ASSERT((size_t) actual <= buffer->frameCount); in getNextBuffer() 82 buffer->frameCount = actual; in getNextBuffer() [all …]
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D | AudioBufferProviderSource.cpp | 46 return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0; in availableToRead() 55 mBuffer.frameCount = count; in read() 63 size_t available = mBuffer.frameCount - mConsumed; in read() 70 if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) { in read() 101 mBuffer.frameCount = count; in readVia() 104 ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count); in readVia() 115 size_t available = mBuffer.frameCount - mConsumed; in readVia() 131 if (CC_LIKELY((mConsumed += ret) < mBuffer.frameCount)) { in readVia()
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/frameworks/av/media/libeffects/testlibs/ |
D | AudioBiquadFilter.cpp | 66 int frameCount) { in process() argument 67 (this->*mCurProcessFunc)(in, out, frameCount); in process() 121 int frameCount) { in updateCoefs() argument 122 int64_t maxDelta = mMaxDelta * frameCount; in updateCoefs() 141 int frameCount) { in process_bypass() argument 144 memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t)); in process_bypass() 150 int frameCount) { in process_normal_mono() argument 151 size_t nFrames = frameCount; in process_normal_mono() 184 int frameCount) { in process_transition_normal_mono() argument 185 if (updateCoefs(mTargetCoefs, frameCount)) { in process_transition_normal_mono() [all …]
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D | AudioBiquadFilter.h | 79 int frameCount); 98 int frameCount); 154 bool updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount); 158 int frameCount); 161 int frameCount); 164 audio_sample_t * out, int frameCount); 167 audio_sample_t * out, int frameCount); 170 int frameCount); 173 audio_sample_t * out, int frameCount); 176 audio_sample_t * out, int frameCount);
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/frameworks/base/graphics/java/android/graphics/ |
D | Interpolator.java | 29 public Interpolator(int valueCount, int frameCount) { in Interpolator() argument 31 mFrameCount = frameCount; in Interpolator() 32 native_instance = nativeConstructor(valueCount, frameCount); in Interpolator() 49 public void reset(int valueCount, int frameCount) { in reset() argument 51 mFrameCount = frameCount; in reset() 52 nativeReset(native_instance, valueCount, frameCount); in reset() 157 private static native long nativeConstructor(int valueCount, int frameCount); in nativeConstructor() argument 159 private static native void nativeReset(long native_instance, int valueCount, int frameCount); in nativeReset() argument
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/frameworks/av/media/libaaudio/tests/ |
D | test_block_adapter.cpp | 38 void fillSequence(int32_t *indexBuffer, int32_t frameCount) { in fillSequence() argument 39 ASSERT_LE(frameCount, TEST_BUFFER_SIZE); in fillSequence() 40 for (int i = 0; i < frameCount; i++) { in fillSequence() 45 int checkSequence(const int32_t *indexBuffer, int32_t frameCount) { in checkSequence() argument 47 for (int i = 0; i < frameCount; i++) { in checkSequence() 75 int32_t frameCount = numBytes / sizeof(int32_t); in onProcessFixedBlock() local 76 return checkSequence((int32_t *) buffer, frameCount); in onProcessFixedBlock() 102 int32_t frameCount = numBytes / sizeof(int32_t); in onProcessFixedBlock() local 103 fillSequence((int32_t *) buffer, frameCount); in onProcessFixedBlock()
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/frameworks/av/media/libaudioprocessing/tests/ |
D | test_utils.h | 124 size_t requestedFrames = buffer->frameCount; 126 buffer->frameCount = mNumFrames - mNextFrame; 131 mNextIdx-1, provided, buffer->frameCount); 132 if (provided < buffer->frameCount) { 133 buffer->frameCount = provided; 141 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount); 142 mUnrel = buffer->frameCount; 143 if (buffer->frameCount > 0) { 154 if (buffer->frameCount > mUnrel) { 156 "to release", buffer->frameCount, mUnrel); [all …]
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D | test-resampler.cpp | 277 size_t requestedFrames = buffer->frameCount; in main() 279 buffer->frameCount = mNumFrames - mNextFrame; in main() 283 printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount); in main() 284 if (provided < buffer->frameCount) { in main() 285 buffer->frameCount = provided; in main() 294 requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount); in main() 296 mUnrel = buffer->frameCount; in main() 297 if (buffer->frameCount > 0) { in main() 306 if (buffer->frameCount > mUnrel) { in main() 308 "to release\n", buffer->frameCount, mUnrel); in main() [all …]
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/frameworks/av/include/private/media/ |
D | AudioTrackShared.h | 207 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, 218 size_t frameCount() const { return mFrameCount; } in frameCount() function 240 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 360 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 362 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, in ClientProxy() argument 417 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 473 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, in AudioRecordClientProxy() argument 475 : ClientProxy(cblk, buffers, frameCount, frameSize, in AudioRecordClientProxy() 495 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 577 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, [all …]
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/frameworks/av/media/libaudioprocessing/tests/fuzzer/ |
D | libaudioprocessing_resampler_fuzzer.cpp | 70 if (buffer->frameCount > mNumFrames - mNextFrame) { in getNextBuffer() 71 buffer->frameCount = mNumFrames - mNextFrame; in getNextBuffer() 73 mUnrel = buffer->frameCount; in getNextBuffer() 74 if (buffer->frameCount > 0) { in getNextBuffer() 83 if (buffer->frameCount > mUnrel) { in releaseBuffer() 87 mNextFrame += buffer->frameCount; in releaseBuffer() 88 mUnrel -= buffer->frameCount; in releaseBuffer() 90 buffer->frameCount = 0; in releaseBuffer()
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/frameworks/av/media/libaudiohal/impl/ |
D | EffectBufferHalHidl.cpp | 62 mHidlBuffer.frameCount = 0; 108 void EffectBufferHalHidl::setFrameCount(size_t frameCount) { in setFrameCount() argument 109 mHidlBuffer.frameCount = frameCount; in setFrameCount() 110 mAudioBuffer.frameCount = frameCount; in setFrameCount()
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/frameworks/av/media/libaudioclient/tests/ |
D | test_create_audiotrack.cpp | 64 size_t frameCount; in testTrack() local 82 &frameCount, ¬ificationFrames, &useSharedBuffer, in testTrack() 92 audio_bytes_per_sample(format) * frameCount; in testTrack() 95 frameCount = 0; in testTrack() 118 frameCount, in testTrack()
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/frameworks/av/media/libaudioclient/include/media/ |
D | IAudioFlinger.h | 84 frameCount = parcel->readInt64(); in readFromParcel() 107 (void)parcel->writeInt64(frameCount); in writeToParcel() 124 size_t frameCount; variable 139 frameCount = parcel->readInt64(); in readFromParcel() 157 (void)parcel->writeInt64(frameCount); in writeToParcel() 174 size_t frameCount; variable 215 frameCount = parcel->readInt64(); in readFromParcel() 232 (void)parcel->writeInt64(frameCount); in writeToParcel() 248 size_t frameCount; variable 263 frameCount = parcel->readInt64(); in readFromParcel() [all …]
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/frameworks/native/services/surfaceflinger/tests/fakehwc/ |
D | FakeComposerUtils.h | 106 int frameCount = mComposer.getFrameCount(); in ~TransactionScope() local 110 mComposer.waitUntilFrame(frameCount + 1); in ~TransactionScope() 111 LOG_ALWAYS_FATAL_IF(frameCount + 1 != mComposer.getFrameCount(), in ~TransactionScope() 113 mComposer.getFrameCount() - frameCount); in ~TransactionScope()
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/frameworks/av/media/libaudioclient/ |
D | AudioRecord.cpp | 45 size_t* frameCount, in getMinFrameCount() argument 50 if (frameCount == NULL) { in getMinFrameCount() 65 if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) * in getMinFrameCount() 136 size_t frameCount, in AudioRecord() argument 157 (void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, in AudioRecord() 198 size_t frameCount, in set() argument 223 inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, in set() 306 mReqFrameCount = frameCount; in set() 761 input.frameCount = mReqFrameCount; in createRecord_l() 785 mReqFrameCount, output.frameCount); in createRecord_l() [all …]
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/frameworks/av/media/libaudioprocessing/include/media/ |
D | AudioMixerBase.h | 91 AudioMixerBase(size_t frameCount, uint32_t sampleRate) in AudioMixerBase() argument 93 , mFrameCount(frameCount) { in AudioMixerBase() 245 uint16_t frameCount; member 290 void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 291 void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 295 void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux); 297 void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
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