Searched refs:frameCount (Results 1 - 25 of 71) sorted by relevance

123

/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/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/services/audioflinger/
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 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 DAudioResampler.cpp280 mBuffer.frameCount = 0;
311 mBuffer.frameCount = 0;
353 while (mBuffer.frameCount == 0) {
354 mBuffer.frameCount = inFrameCount;
360 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
361 if (mBuffer.frameCount > inputIndex) break;
363 inputIndex -= mBuffer.frameCount;
364 mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
365 mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
367 // 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 DFastMixer.cpp141 const size_t frameCount = current->mFrameCount; local
174 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) {
182 if (frameCount > 0 && mSampleRate > 0) {
190 mMixer = new AudioMixer(frameCount, mSampleRate, FastMixerState::sMaxFastTracks);
193 mMixerBufferSize = mixerFrameSize * frameCount;
198 mSinkBufferSize = sinkFrameSize * frameCount;
201 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
202 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
203 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
204 mForceNs = (frameCount * 95000000
336 const size_t frameCount = current->mFrameCount; local
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H A DTracks.cpp72 size_t frameCount,
94 mFrameCount(frameCount),
116 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize;
251 (void) mTeeSink->write(buffer->raw, buffer->frameCount);
256 buf.mFrameCount = buffer->frameCount;
258 buffer->frameCount = 0;
343 size_t frameCount,
350 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
384 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
66 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int clientUid, IAudioFlinger::track_flags_t flags, bool isOut, alloc_type alloc, track_type type) argument
336 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, const sp<IMemory>& sharedBuffer, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument
1126 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int uid) argument
1325 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, IAudioFlinger::track_flags_t flags) argument
1461 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument
1661 PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) argument
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H A Dtest-resample.cpp276 size_t requestedFrames = buffer->frameCount;
278 buffer->frameCount = mNumFrames - mNextFrame;
282 printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount);
283 if (provided < buffer->frameCount) {
284 buffer->frameCount = provided;
293 requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount);
295 mUnrel = buffer->frameCount;
296 if (buffer->frameCount > 0) {
305 if (buffer->frameCount > mUnrel) {
307 "to release\n", buffer->frameCount, mUnre
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H A DAudioMixer.cpp101 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument
115 mState.frameCount = frameCount;
199 // t->frameCount
210 // t->buffer.frameCount
703 target == RAMP_VOLUME ? mState.frameCount : 0,
714 target == RAMP_VOLUME ? mState.frameCount : 0,
1003 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
1006 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
1114 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_ argument
1157 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
1186 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument
1278 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument
1692 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument
1736 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument
1902 track__NoResample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument
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H A DAudioResamplerDyn.cpp523 ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d",
524 inputIndex, mBuffer.frameCount);
527 while (mBuffer.frameCount == 0 && inFrameCount > 0) {
528 mBuffer.frameCount = inFrameCount;
533 inFrameCount -= mBuffer.frameCount;
541 if (inputIndex >= mBuffer.frameCount) {
555 const size_t frameCount = mBuffer.frameCount; local
582 if (inputIndex >= frameCount) {
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H A DAudioMixer.h45 AudioMixer(size_t frameCount, uint32_t sampleRate,
193 uint16_t frameCount; member in struct:android::AudioMixer::track_t
287 size_t frameCount; member in struct:android::AudioMixer::state_t
326 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
328 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
361 static void track__Resample(track_t* t, TO* out, size_t frameCount,
364 static void track__NoResample(track_t* t, TO* out, size_t frameCount,
/frameworks/av/services/audioflinger/tests/
H A Dtest_utils.h117 size_t requestedFrames = buffer->frameCount;
119 buffer->frameCount = mNumFrames - mNextFrame;
124 mNextIdx-1, provided, buffer->frameCount);
125 if (provided < buffer->frameCount) {
126 buffer->frameCount = provided;
134 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
135 mUnrel = buffer->frameCount;
136 if (buffer->frameCount > 0) {
147 if (buffer->frameCount > mUnrel) {
149 "to release", buffer->frameCount, mUnre
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/frameworks/av/include/private/media/
H A DAudioTrackShared.h205 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut,
216 size_t frameCount() const { return mFrameCount; } function in class:android::Proxy
238 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
356 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
358 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
407 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
457 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
459 : ClientProxy(cblk, buffers, frameCount, frameSize,
479 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
544 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
646 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/av/media/libmedia/
H A DAudioRecord.cpp37 size_t* frameCount,
42 if (frameCount == NULL) {
56 if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
81 size_t frameCount,
100 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
138 size_t frameCount,
150 ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
153 inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
225 mReqFrameCount = frameCount;
594 size_t frameCount local
36 getMinFrameCount( size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument
75 AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& opPackageName, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument
133 set( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, bool threadCanCallJava, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument
[all...]
H A DAudioTrackShared.cpp59 Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, argument
61 : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize),
62 mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer),
69 ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
71 : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer)
75 setBufferSizeInFrames(frameCount);
94 const uint32_t maximum = frameCount();
524 size_t frameCount, size_t frameSize)
525 : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
615 ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument
523 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) argument
829 tallyUnderrunFrames(uint32_t frameCount) argument
861 StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) argument
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/frameworks/native/opengl/tests/angeles/
H A Dapp-linux.cpp206 int frameCount = 0; local
220 frameCount++;
229 printf("totalTime=%f s, frameCount=%d, %.2f fps\n",
230 totalTime, frameCount, frameCount/totalTime);

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