Searched refs:frameCount (Results 1 - 25 of 41) sorted by relevance

12

/frameworks/base/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);
[all...]
H A DAudioBiquadFilter.cpp68 int frameCount) {
69 (this->*mCurProcessFunc)(in, out, frameCount);
123 int frameCount) {
124 int64_t maxDelta = mMaxDelta * frameCount;
143 int frameCount) {
146 memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
152 int frameCount) {
153 size_t nFrames = frameCount;
186 int frameCount) {
187 if (updateCoefs(mTargetCoefs, frameCount)) {
67 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
122 updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount) argument
141 process_bypass(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
150 process_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
184 process_transition_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
193 process_transition_bypass_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
202 process_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
242 process_transition_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
251 process_transition_bypass_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument
[all...]
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
H A DAudioEqualizer.h180 // frameCount * nChannels interlaced samples. Processing can be done
184 // frameCount Number of frames to produce on each call to process().
186 int frameCount);
/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);
156 private static native int nativeConstructor(int valueCount, int frameCount); argument
158 private static native void nativeReset(int native_instance, int valueCount, int frameCount); argument
/frameworks/base/services/audioflinger/
H A DAudioResampler.cpp127 mBuffer.frameCount = 0;
189 while (mBuffer.frameCount == 0) {
190 mBuffer.frameCount = inFrameCount;
196 // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
197 if (mBuffer.frameCount > inputIndex) break;
199 inputIndex -= mBuffer.frameCount;
200 mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
201 mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
203 // mBuffer.frameCount == 0 now so we reload a new buffer
222 if (inputIndex + 2 < mBuffer.frameCount) {
[all...]
H A DAudioResamplerCubic.cpp66 if (mBuffer.frameCount == 0) {
67 mBuffer.frameCount = inFrameCount;
71 // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
94 if (inputIndex == mBuffer.frameCount) {
97 mBuffer.frameCount = inFrameCount;
102 // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
131 if (mBuffer.frameCount == 0) {
132 mBuffer.frameCount = inFrameCount;
136 // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
159 if (inputIndex == mBuffer.frameCount) {
[all...]
H A DAudioBufferProvider.h37 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer
H A DAudioMixer.cpp43 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) argument
48 mState.frameCount = frameCount;
237 int32_t volInc = d / int32_t(mState.frameCount);
257 int32_t volInc = d / int32_t(mState.frameCount);
401 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
404 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
579 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
588 // (vl + vlInc*frameCount)/65536.0f, frameCount);
621 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
649 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
739 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
[all...]
H A DAudioMixer.h39 AudioMixer(size_t frameCount, uint32_t sampleRate);
146 uint16_t frameCount; member in struct:android::AudioMixer::track_t
173 size_t frameCount; member in struct:android::AudioMixer::state_t
193 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
194 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
H A DAudioResamplerSinc.cpp205 while (buffer.frameCount == 0) {
206 buffer.frameCount = inFrameCount;
219 if (inputIndex >= mBuffer.frameCount) {
220 inputIndex -= mBuffer.frameCount;
228 const size_t frameCount = buffer.frameCount; local
247 if (inputIndex >= frameCount)
252 if (inputIndex >= frameCount)
257 if (inputIndex >= frameCount)
265 if (inputIndex >= frameCount) {
[all...]
H A DAudioFlinger.cpp287 int frameCount,
354 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);
413 size_t AudioFlinger::frameCount(int output) const function in class:android::AudioFlinger
418 LOGW("frameCount() unknown thread %d", output);
421 return thread->frameCount();
882 size_t AudioFlinger::ThreadBase::frameCount() const function in class:android::AudioFlinger::ThreadBase
1135 int frameCount,
1185 channelCount, frameCount, sharedBuffer, sessionId);
1359 desc.frameCount = mFrameCount;
2092 size_t frameCount local
281 createTrack( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int output, int *sessionId, status_t *status) argument
1129 createTrack_l( const sp<AudioFlinger::Client>& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId, status_t *status) argument
2354 size_t frameCount = mFrameCount; local
2682 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); local
2751 TrackBase( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int sessionId) argument
2905 Track( const wp<ThreadBase>& thread, const sp<Client>& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument
3207 RecordTrack( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, int sessionId) argument
3320 OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, int format, int channelCount, int frameCount) argument
3655 openRecord( pid_t pid, int input, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, int *sessionId, status_t *status) argument
[all...]
/frameworks/base/media/libmedia/
H A DAudioTrack.cpp48 int* frameCount,
69 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
86 int frameCount,
95 frameCount, flags, cbf, user, notificationFrames,
140 int frameCount,
209 mFrameCount = frameCount;
216 frameCount, flags, sharedBuffer, output, true);
278 uint32_t AudioTrack::frameCount() const function in class:android::AudioTrack
280 return mCblk->frameCount;
518 loopEnd - loopStart > cblk->frameCount) {
47 getMinFrameCount( int* frameCount, int streamType, uint32_t sampleRate) argument
81 AudioTrack( int streamType, uint32_t sampleRate, int format, int channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument
135 set( int streamType, uint32_t sampleRate, int format, int channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId) argument
653 createTrack( int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, bool enforceFrameCount) argument
1118 stepUser(uint32_t frameCount) argument
1146 stepServer(uint32_t frameCount) argument
[all...]
H A DAudioRecord.cpp48 int* frameCount,
73 *frameCount = size;
89 int frameCount,
98 frameCount, flags, cbf, user, notificationFrames, sessionId);
122 int frameCount,
131 LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
174 if (frameCount == 0) {
175 frameCount = minFrameCount;
176 } else if (frameCount < minFrameCoun
47 getMinFrameCount( int* frameCount, uint32_t sampleRate, int format, int channelCount) argument
84 AudioRecord( int inputSource, uint32_t sampleRate, int format, uint32_t channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument
117 set( int inputSource, uint32_t sampleRate, int format, uint32_t channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) argument
247 uint32_t AudioRecord::frameCount() const function in class:android::AudioRecord
416 openRecord( uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, audio_io_handle_t input) argument
[all...]
H A DIAudioFlingerClient.cpp57 data.writeInt32(desc->frameCount);
87 desc.frameCount = data.readInt32();
/frameworks/base/include/private/media/
H A DAudioTrackShared.h63 uint32_t frameCount; member in struct:android::audio_track_cblk_t
88 uint32_t stepUser(uint32_t frameCount);
89 bool stepServer(uint32_t frameCount);
/frameworks/base/include/media/
H A DAudioRecord.h70 size_t frameCount; member in class:android::AudioRecord::Buffer
111 static status_t getMinFrameCount(int* frameCount,
133 * frameCount: Total size of track PCM buffer in frames. This defines the
154 int frameCount = 0,
180 int frameCount = 0,
207 uint32_t frameCount() const;
298 /* obtains a buffer of "frameCount" frames. The buffer must be
352 int frameCount,
H A DAudioTrack.h75 size_t frameCount; member in class:android::AudioTrack::Buffer
112 static status_t getMinFrameCount(int* frameCount,
134 * frameCount: Total size of track PCM buffer in frames. This defines the
148 int frameCount = 0,
192 int frameCount = 0,
220 uint32_t frameCount() const;
389 /* obtains a buffer of "frameCount" frames. The buffer must be
444 int frameCount,
H A DIAudioFlinger.h53 int frameCount,
66 int frameCount,
77 virtual size_t frameCount(int output) const = 0;
/frameworks/base/core/jni/android/graphics/
H A DInterpolator.cpp8 static SkInterpolator* Interpolator_constructor(JNIEnv* env, jobject clazz, int valueCount, int frameCount) argument
10 return new SkInterpolator(valueCount, frameCount);
18 static void Interpolator_reset(JNIEnv* env, jobject clazz, SkInterpolator* interp, int valueCount, int frameCount) argument
20 interp->reset(valueCount, frameCount);
/frameworks/base/opengl/tests/angeles/
H A Dapp-linux.cpp205 int frameCount = 0; local
219 frameCount++;
228 printf("totalTime=%f s, frameCount=%d, %.2f fps\n",
229 totalTime, frameCount, frameCount/totalTime);
/frameworks/base/media/libeffects/lvm/wrapper/Bundle/
H A DEffectBundle.h97 int frameCount; member in struct:BundledEffectContext
/frameworks/base/media/libeffects/visualizer/
H A DEffectVisualizer.cpp236 inBuffer->frameCount != outBuffer->frameCount ||
237 inBuffer->frameCount == 0) {
246 int len = inBuffer->frameCount * 2;
267 inIdx < inBuffer->frameCount && captIdx < pContext->mCaptureSize;
283 for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
287 memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(int16_t));
/frameworks/base/media/libeffects/lvm/wrapper/Reverb/
H A DEffectReverb.cpp401 // frameCount: Frames to process
412 int frameCount,
437 fwrite(pIn, frameCount*sizeof(LVM_INT16)*samplesPerFrame, 1, pContext->PcmInPtr);
449 for(int i=0; i<frameCount*samplesPerFrame; i++){
454 for (int i = 0; i < frameCount; i++) {
461 memset(pContext->OutFrames32, 0, frameCount * sizeof(LVM_INT32) * 2); //always stereo here
464 memset(pContext->InFrames32,0,frameCount * sizeof(LVM_INT32) * samplesPerFrame);
472 frameCount); /* Number of samples to read */
480 for (int i=0; i < frameCount*2; i++) { //always stereo here
484 for (int i=0; i < frameCount*
410 process( LVM_INT16 *pIn, LVM_INT16 *pOut, int frameCount, ReverbContext *pContext) argument
[all...]

Completed in 317 milliseconds

12