/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 | 131 mBuffer.frameCount = 0; 158 mBuffer.frameCount = 0; 199 while (mBuffer.frameCount == 0) { 200 mBuffer.frameCount = inFrameCount; 206 // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); 207 if (mBuffer.frameCount > inputIndex) break; 209 inputIndex -= mBuffer.frameCount; 210 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; 211 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; 213 // 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 | 47 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) argument 52 mState.frameCount = frameCount; 253 int32_t volInc = d / int32_t(mState.frameCount); 273 int32_t volInc = d / int32_t(mState.frameCount); 441 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 444 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 619 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 628 // (vl + vlInc*frameCount)/65536.0f, frameCount); 661 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 689 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 779 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); 149 uint16_t frameCount; member in struct:android::AudioMixer::track_t 179 size_t frameCount; member in struct:android::AudioMixer::state_t 199 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); 200 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 | 383 int frameCount, 450 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 509 size_t AudioFlinger::frameCount(int output) const function in class:android::AudioFlinger 514 LOGW("frameCount() unknown thread %d", output); 517 return thread->frameCount(); 1031 size_t AudioFlinger::ThreadBase::frameCount() const function in class:android::AudioFlinger::ThreadBase 1485 int frameCount, 1538 channelMask, frameCount, sharedBuffer, sessionId); 1703 desc.frameCount = mFrameCount; 2126 LOG_ASSERT(minFrames <= cblk->frameCount); 377 createTrack( pid_t pid, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int output, int *sessionId, status_t *status) argument 1479 createTrack_l( const sp<AudioFlinger::Client>& client, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId, status_t *status) argument 2521 size_t frameCount = mFrameCount; local 2786 size_t frameCount = mFrameCount; local 3128 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); local 3197 TrackBase( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int sessionId) argument 3358 Track( const wp<ThreadBase>& thread, const sp<Client>& client, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument 3672 RecordTrack( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, int sessionId) argument 3785 OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount) argument 4122 openRecord( pid_t pid, int input, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, int *sessionId, status_t *status) argument 4468 createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, int format, int channelMask, int frameCount, uint32_t flags, int sessionId, status_t *status) argument [all...] |
/frameworks/base/include/private/media/ |
H A D | AudioTrackShared.h | 69 uint32_t frameCount; member in struct:android::audio_track_cblk_t 94 uint32_t stepUser(uint32_t frameCount); 95 bool stepServer(uint32_t frameCount);
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/frameworks/base/media/libmedia/ |
H A D | AudioTrack.cpp | 53 int* frameCount, 74 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 91 int frameCount, 100 frameCount, flags, cbf, user, notificationFrames, 146 int frameCount, 218 mFrameCount = frameCount; 228 frameCount, 296 uint32_t AudioTrack::frameCount() const function in class:android::AudioTrack 298 return mCblk->frameCount; 562 loopEnd - loopStart > cblk->frameCount || 52 getMinFrameCount( int* frameCount, int streamType, uint32_t sampleRate) argument 86 AudioTrack( int streamType, uint32_t sampleRate, int format, int channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 141 set( int streamType, uint32_t sampleRate, int format, int channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId) argument 710 createTrack_l( int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, bool enforceFrameCount) argument 1313 stepUser(uint32_t frameCount) argument 1343 stepServer(uint32_t frameCount) argument [all...] |
H A D | AudioRecord.cpp | 51 int* frameCount, 76 *frameCount = size; 92 int frameCount, 101 frameCount, flags, cbf, user, notificationFrames, sessionId); 126 int frameCount, 135 LOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount); 192 if (frameCount == 0) { 193 frameCount = minFrameCount; 194 } else if (frameCount < minFrameCoun 50 getMinFrameCount( int* frameCount, uint32_t sampleRate, int format, int channelCount) argument 87 AudioRecord( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 121 set( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) argument 264 uint32_t AudioRecord::frameCount() const function in class:android::AudioRecord 449 openRecord_l( uint32_t sampleRate, uint32_t format, uint32_t channelMask, 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/media/ |
H A D | AudioRecord.h | 71 size_t frameCount; member in class:android::AudioRecord::Buffer 112 static status_t getMinFrameCount(int* frameCount, 134 * frameCount: Total size of track PCM buffer in frames. This defines the 155 int frameCount = 0, 181 int frameCount = 0, 208 uint32_t frameCount() const; 299 /* obtains a buffer of "frameCount" frames. The buffer must be 353 int frameCount,
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H A D | AudioTrack.h | 74 size_t frameCount; member in class:android::AudioTrack::Buffer 111 static status_t getMinFrameCount(int* frameCount, 133 * frameCount: Minimum size of track PCM buffer in frames. This defines the 149 int frameCount = 0, 193 int frameCount = 0, 221 uint32_t frameCount() const; 390 /* obtains a buffer of "frameCount" frames. The buffer must be 445 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/media/libvideoeditor/lvpp/ |
H A D | VideoEditorSRC.cpp | 189 LOGV("Requesting %d, chan = %d", pBuffer->frameCount, mChannelCnt); 191 uint32_t want = pBuffer->frameCount * mChannelCnt * 2; 213 pBuffer->frameCount = 0; 267 pBuffer->frameCount = done / (mChannelCnt * 2); 268 LOGV("getNextBuffer done %d", pBuffer->frameCount); 276 pBuffer->frameCount = 0;
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/frameworks/base/media/libeffects/preprocessing/ |
H A D | PreProcessing.cpp | 101 size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount) member in struct:preproc_session_s 751 session->frameCount = session->apmFrameCount; 873 session->frameCount = session->apmFrameCount; 875 session->frameCount = (session->apmFrameCount * session->samplingRate) / 1091 // inBuffer->frameCount, session->enabledMsk, session->processedMsk); 1095 size_t framesRq = outBuffer->frameCount; 1099 if (outBuffer->frameCount < fr) { 1100 fr = outBuffer->frameCount; 1111 outBuffer->frameCount = framesWr; 1113 inBuffer->frameCount [all...] |
/frameworks/base/media/libeffects/lvm/wrapper/Bundle/ |
H A D | EffectBundle.h | 97 int frameCount; member in struct:BundledEffectContext
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