Searched refs:frameCount (Results 1 - 25 of 110) sorted by relevance

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/frameworks/native/libs/ui/
H A DFrameStats.cpp41 size_t frameCount = desiredPresentTimesNano.size(); local
46 memcpy(timestamps, desiredPresentTimesNano.array(), frameCount * timestampSize);
47 timestamps += frameCount;
49 memcpy(timestamps, actualPresentTimesNano.array(), frameCount * timestampSize);
50 timestamps += frameCount;
52 memcpy(timestamps, frameReadyTimesNano.array(), frameCount * timestampSize);
65 size_t frameCount = (size - timestampSize) / (3 * timestampSize); local
70 desiredPresentTimesNano.resize(frameCount);
71 memcpy(desiredPresentTimesNano.editArray(), timestamps, frameCount * timestampSize);
72 timestamps += frameCount;
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/frameworks/av/media/libeffects/testlibs/
H A DAudioBiquadFilter.h27 // The filter works on fixed sized blocks of data (frameCount multi-channel
72 // Process a buffer of data. Always processes frameCount multi-channel
75 // in The input buffer. Should be of size frameCount * nChannels.
76 // out The output buffer. Should be of size frameCount * nChannels.
77 // frameCount Number of multi-channel samples to process.
79 int frameCount);
98 int frameCount);
154 bool updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount);
158 int frameCount);
161 int frameCount);
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H A DAudioBiquadFilter.cpp66 int frameCount) {
67 (this->*mCurProcessFunc)(in, out, frameCount);
121 int frameCount) {
122 int64_t maxDelta = mMaxDelta * frameCount;
141 int frameCount) {
144 memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
150 int frameCount) {
151 size_t nFrames = frameCount;
184 int frameCount) {
185 if (updateCoefs(mTargetCoefs, frameCount)) {
65 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
120 updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount) argument
139 process_bypass(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
148 process_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
182 process_transition_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
191 process_transition_bypass_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
200 process_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
240 process_transition_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
249 process_transition_bypass_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
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H A DAudioShelvingFilter.h93 // frameCount * nChannels interlaced samples. Processing can be done
97 // frameCount Number of frames to produce.
99 int frameCount) { mBiquad.process(in, out, frameCount); }
98 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
H A DAudioPeakingFilter.h99 // frameCount * nChannels interlaced samples. Processing can be done
103 // frameCount Number of frames to produce.
105 int frameCount) { mBiquad.process(in, out, frameCount); }
104 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
/frameworks/av/media/libnbaio/
H A DSourceAudioBufferProvider.cpp50 ALOG_ASSERT(buffer != NULL && buffer->frameCount > 0 && mGetCount == 0);
54 if (mRemaining < buffer->frameCount) {
55 buffer->frameCount = mRemaining;
58 mGetCount = buffer->frameCount;
62 if (buffer->frameCount > mSize) {
67 mAllocated = calloc(buffer->frameCount, mFrameSize);
72 mSize = buffer->frameCount;
76 ssize_t actual = mSource->read(mAllocated, buffer->frameCount);
78 ALOG_ASSERT((size_t) actual <= buffer->frameCount);
82 buffer->frameCount
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H A DAudioBufferProviderSource.cpp46 return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0;
55 mBuffer.frameCount = count;
63 size_t available = mBuffer.frameCount - mConsumed;
70 if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) {
101 mBuffer.frameCount = count;
104 ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count);
115 size_t available = mBuffer.frameCount - mConsumed;
131 if (CC_LIKELY((mConsumed += ret) < mBuffer.frameCount)) {
/frameworks/av/include/media/
H A DAudioBufferProvider.h32 Buffer() : raw(NULL), frameCount(0) { }
38 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer
46 // buffer->frameCount maximum number of desired frames
49 // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames
50 // buffer->frameCount number of contiguous available frames at buffer->raw,
51 // 0 < buffer->frameCount <= entry value
55 // buffer->frameCount 0
61 // buffer->frameCount number of frames to release, must be <= number of frames
65 // buffer->frameCount 0; implementation MUST set to zero
/frameworks/av/media/libaudioclient/include/media/
H A DAudioBufferProvider.h32 Buffer() : raw(NULL), frameCount(0) { }
38 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer
46 // buffer->frameCount maximum number of desired frames
49 // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames
50 // buffer->frameCount number of contiguous available frames at buffer->raw,
51 // 0 < buffer->frameCount <= entry value
55 // buffer->frameCount 0
61 // buffer->frameCount number of frames to release, must be <= number of frames
65 // buffer->frameCount 0; implementation MUST set to zero
/frameworks/base/graphics/java/android/graphics/
H A DInterpolator.java29 public Interpolator(int valueCount, int frameCount) { argument
31 mFrameCount = frameCount;
32 native_instance = nativeConstructor(valueCount, frameCount);
49 public void reset(int valueCount, int frameCount) { argument
51 mFrameCount = frameCount;
52 nativeReset(native_instance, valueCount, frameCount);
157 private static native long nativeConstructor(int valueCount, int frameCount); argument
159 private static native void nativeReset(long native_instance, int valueCount, int frameCount); argument
/frameworks/av/media/libaaudio/tests/
H A Dtest_block_adapter.cpp38 void fillSequence(int32_t *indexBuffer, int32_t frameCount) { argument
39 ASSERT_LE(frameCount, TEST_BUFFER_SIZE);
40 for (int i = 0; i < frameCount; i++) {
45 int checkSequence(const int32_t *indexBuffer, int32_t frameCount) { argument
47 for (int i = 0; i < frameCount; i++) {
75 int32_t frameCount = numBytes / sizeof(int32_t); variable
76 return checkSequence((int32_t *) buffer, frameCount);
102 int32_t frameCount = numBytes / sizeof(int32_t); variable
103 fillSequence((int32_t *) buffer, frameCount);
/frameworks/av/media/libaudioprocessing/
H A DBufferProviders.cpp61 mBuffer.frameCount = 0;
67 if (mBuffer.frameCount != 0) {
76 // this, pBuffer, pBuffer->frameCount);
80 copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
84 if (mBuffer.frameCount == 0) {
85 mBuffer.frameCount = pBuffer->frameCount;
88 // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
90 // By API spec, if res != OK, then mBuffer.frameCount == 0.
92 ALOG_ASSERT(res == OK || mBuffer.frameCount
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H A DAudioResampler.cpp283 mBuffer.frameCount = 0;
314 mBuffer.frameCount = 0;
356 while (mBuffer.frameCount == 0) {
357 mBuffer.frameCount = inFrameCount;
363 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
364 if (mBuffer.frameCount > inputIndex) break;
366 inputIndex -= mBuffer.frameCount;
367 mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
368 mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
370 // mBuffer.frameCount
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H A DAudioResamplerCubic.cpp67 if (mBuffer.frameCount == 0) {
68 mBuffer.frameCount = inFrameCount;
73 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
95 if (inputIndex == mBuffer.frameCount) {
98 mBuffer.frameCount = inFrameCount;
104 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
134 if (mBuffer.frameCount == 0) {
135 mBuffer.frameCount = inFrameCount;
140 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
163 if (inputIndex == mBuffer.frameCount) {
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H A DAudioMixer.cpp138 // t->frameCount
149 // t->buffer.frameCount
1004 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
1013 // (vl + vlInc*frameCount)/65536.0f, frameCount);
1031 } while (--frameCount);
1039 } while (--frameCount);
1047 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
1063 } while (--frameCount);
1071 } while (--frameCount);
1003 volumeRampStereo( int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
1046 volumeStereo( int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
1075 track__16BitsStereo( int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument
1167 track__16BitsMono( int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument
1308 const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames); local
1526 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument
1570 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument
1747 track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument
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/frameworks/av/services/audioflinger/
H A DFastCapture.cpp92 const size_t frameCount = current->mFrameCount; local
128 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) {
132 if (frameCount > 0 && mSampleRate > 0) {
136 size_t bufferSize = frameCount * Format_frameSize(mFormat);
139 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
140 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
141 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
142 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95
143 mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75
144 mWarmupNsMax = (frameCount * 125000000
164 const size_t frameCount = current->mFrameCount; local
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H A DFastMixer.cpp143 const size_t frameCount = current->mFrameCount; local
176 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) {
184 if (frameCount > 0 && mSampleRate > 0) {
192 mMixer = new AudioMixer(frameCount, mSampleRate);
196 mMixerBufferSize = mixerFrameSize * frameCount;
201 mSinkBufferSize = sinkFrameSize * frameCount;
204 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
205 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
206 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
207 mForceNs = (frameCount * 95000000
339 const size_t frameCount = current->mFrameCount; local
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H A DTracks.cpp70 size_t frameCount,
94 mFrameCount(frameCount),
116 size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
118 if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2
278 (void) mTeeSink->write(buffer->raw, buffer->frameCount);
283 buf.mFrameCount = buffer->frameCount;
285 buffer->frameCount = 0;
381 size_t frameCount,
390 : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
63 TrackBase( ThreadBase *thread, const sp<Client>& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, uid_t clientUid, bool isOut, alloc_type alloc, track_type type, audio_port_handle_t portId) argument
373 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, const sp<IMemory>& sharedBuffer, audio_session_t sessionId, uid_t uid, audio_output_flags_t flags, track_type type, audio_port_handle_t portId) argument
1282 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, uid_t uid) argument
1499 PatchTrack(PlaybackThread *playbackThread, audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, size_t bufferSize, audio_output_flags_t flags) argument
1643 RecordTrack( RecordThread *thread, const sp<Client>& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, uid_t uid, audio_input_flags_t flags, track_type type, audio_port_handle_t portId) argument
1867 PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, size_t bufferSize, audio_input_flags_t flags) argument
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/frameworks/av/media/libaudioprocessing/tests/
H A Dtest_utils.h123 size_t requestedFrames = buffer->frameCount;
125 buffer->frameCount = mNumFrames - mNextFrame;
130 mNextIdx-1, provided, buffer->frameCount);
131 if (provided < buffer->frameCount) {
132 buffer->frameCount = provided;
140 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
141 mUnrel = buffer->frameCount;
142 if (buffer->frameCount > 0) {
153 if (buffer->frameCount > mUnrel) {
155 "to release", buffer->frameCount, mUnre
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/frameworks/av/media/libaudiohal/2.0/
H A DEffectBufferHalHidl.cpp60 mHidlBuffer.frameCount = 0;
106 void EffectBufferHalHidl::setFrameCount(size_t frameCount) { argument
107 mHidlBuffer.frameCount = frameCount;
108 mAudioBuffer.frameCount = frameCount;
/frameworks/av/media/libaudiohal/4.0/
H A DEffectBufferHalHidl.cpp61 mHidlBuffer.frameCount = 0;
107 void EffectBufferHalHidl::setFrameCount(size_t frameCount) { argument
108 mHidlBuffer.frameCount = frameCount;
109 mAudioBuffer.frameCount = frameCount;
/frameworks/av/media/libaudioclient/tests/
H A Dtest_create_audiotrack.cpp64 size_t frameCount; local
82 &frameCount, &notificationFrames, &useSharedBuffer,
92 audio_bytes_per_sample(format) * frameCount;
95 frameCount = 0;
118 frameCount,
/frameworks/av/include/private/media/
H A DAudioTrackShared.h207 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut,
218 size_t frameCount() const { return mFrameCount; } function in class:android::Proxy
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, argument
362 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
417 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
473 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
475 : ClientProxy(cblk, buffers, frameCount, frameSize,
495 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
570 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
688 AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool clientInServer) argument
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/frameworks/base/core/jni/android/graphics/
H A DInterpolator.cpp7 static jlong Interpolator_constructor(JNIEnv* env, jobject clazz, jint valueCount, jint frameCount) argument
9 return reinterpret_cast<jlong>(new SkInterpolator(valueCount, frameCount));
18 static void Interpolator_reset(JNIEnv* env, jobject clazz, jlong interpHandle, jint valueCount, jint frameCount) argument
21 interp->reset(valueCount, frameCount);
/frameworks/native/services/surfaceflinger/tests/fakehwc/
H A DFakeComposerUtils.h109 int frameCount = mComposer.getFrameCount(); local
113 mComposer.waitUntilFrame(frameCount + 1);
114 LOG_ALWAYS_FATAL_IF(frameCount + 1 != mComposer.getFrameCount(),
116 mComposer.getFrameCount() - frameCount);

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