/frameworks/base/media/libeffects/testlibs/ |
H A D | AudioBiquadFilter.h | 27 // 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 D | AudioBiquadFilter.cpp | 68 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 D | AudioShelvingFilter.h | 93 // 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
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H A D | AudioPeakingFilter.h | 99 // 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
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H A D | AudioEqualizer.h | 180 // frameCount * nChannels interlaced samples. Processing can be done 184 // frameCount Number of frames to produce on each call to process(). 186 int frameCount);
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/frameworks/base/graphics/java/android/graphics/ |
H A D | Interpolator.java | 29 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
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/frameworks/base/services/audioflinger/ |
H A D | AudioResampler.cpp | 127 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 D | AudioResamplerCubic.cpp | 66 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 D | AudioBufferProvider.h | 37 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer
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H A D | AudioMixer.cpp | 43 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 D | AudioMixer.h | 39 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);
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H A D | AudioResamplerSinc.cpp | 205 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 D | AudioFlinger.cpp | 287 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 D | AudioTrack.cpp | 48 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 D | AudioRecord.cpp | 48 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 D | IAudioFlingerClient.cpp | 57 data.writeInt32(desc->frameCount); 87 desc.frameCount = data.readInt32();
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/frameworks/base/include/private/media/ |
H A D | AudioTrackShared.h | 63 uint32_t frameCount; member in struct:android::audio_track_cblk_t 88 uint32_t stepUser(uint32_t frameCount); 89 bool stepServer(uint32_t frameCount);
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/frameworks/base/include/media/ |
H A D | AudioRecord.h | 70 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,
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H A D | AudioTrack.h | 75 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,
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H A D | IAudioFlinger.h | 53 int frameCount, 66 int frameCount, 77 virtual size_t frameCount(int output) const = 0;
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/frameworks/base/core/jni/android/graphics/ |
H A D | Interpolator.cpp | 8 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);
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/frameworks/base/opengl/tests/angeles/ |
H A D | app-linux.cpp | 205 int frameCount = 0; local 219 frameCount++; 228 printf("totalTime=%f s, frameCount=%d, %.2f fps\n", 229 totalTime, frameCount, frameCount/totalTime);
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/frameworks/base/media/libeffects/lvm/wrapper/Bundle/ |
H A D | EffectBundle.h | 97 int frameCount; member in struct:BundledEffectContext
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/frameworks/base/media/libeffects/visualizer/ |
H A D | EffectVisualizer.cpp | 236 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));
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/frameworks/base/media/libeffects/lvm/wrapper/Reverb/ |
H A D | EffectReverb.cpp | 401 // 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...] |