/frameworks/av/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 | 66 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 [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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/frameworks/av/media/libnbaio/ |
H A D | SourceAudioBufferProvider.cpp | 50 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) { 64 mAllocated = malloc(buffer->frameCount << mFrameBitShift); 65 mSize = buffer->frameCount; 68 ssize_t actual = mSource->read(mAllocated, buffer->frameCount, pts); 70 ALOG_ASSERT((size_t) actual <= buffer->frameCount); 74 buffer->frameCount [all...] |
H A D | AudioBufferProviderSource.cpp | 46 return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0; 57 mBuffer.frameCount = count; 65 size_t available = mBuffer.frameCount - mConsumed; 72 if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) { 104 mBuffer.frameCount = count; 107 ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count); 118 size_t available = mBuffer.frameCount - mConsumed; 134 if (CC_LIKELY((mConsumed += ret) < mBuffer.frameCount)) {
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/frameworks/av/include/media/ |
H A D | AudioBufferProvider.h | 32 Buffer() : raw(NULL), frameCount(0) { } 38 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer 52 // buffer->frameCount maximum number of desired frames 55 // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames 56 // buffer->frameCount number of contiguous available frames at buffer->raw, 57 // 0 < buffer->frameCount <= entry value 61 // buffer->frameCount 0
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H A D | AudioRecord.h | 60 size_t frameCount; // number of sample frames corresponding to size; member in class:android::AudioRecord::Buffer 65 size_t size; // input/output in bytes == frameCount * frameSize 66 // FIXME this is redundant with respect to frameCount, 104 static status_t getMinFrameCount(size_t* frameCount, 134 * frameCount: Minimum size of track PCM buffer in frames. This defines the 154 int frameCount = 0, 186 int frameCount = 0, 212 size_t frameCount() const { return mFrameCount; } function in class:android::AudioRecord 310 /* Obtains a buffer of up to "audioBuffer->frameCount" full frames. 333 * frameCount numbe [all...] |
H A D | AudioTrack.h | 52 // ignore the event by setting frameCount to zero. 79 size_t frameCount; // number of sample frames corresponding to size; member in class:android::AudioTrack::Buffer 84 size_t size; // input/output in bytes == frameCount * frameSize 86 // FIXME this is redundant with respect to frameCount, 128 static status_t getMinFrameCount(size_t* frameCount, 162 * frameCount: Minimum size of track PCM buffer in frames. This defines the 183 int frameCount = 0, 233 * If sharedBuffer is non-0, the frameCount parameter is ignored and 244 int frameCount = 0, 282 uint32_t frameCount() cons function in class:android::AudioTrack [all...] |
H A D | AudioSystem.h | 95 static status_t getOutputFrameCount(size_t* frameCount, 106 size_t* frameCount); 158 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {} 163 size_t frameCount; member in class:android::AudioSystem::OutputDescriptor
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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/av/include/private/media/ |
H A D | AudioTrackShared.h | 150 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, 181 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 270 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 272 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, 308 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 336 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 338 : ClientProxy(cblk, buffers, frameCount, frameSize, 348 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 394 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 396 : ServerProxy(cblk, buffers, frameCount, frameSiz 458 AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) argument [all...] |
/frameworks/av/services/audioflinger/ |
H A D | AudioResampler.cpp | 228 mBuffer.frameCount = 0; 275 mBuffer.frameCount = 0; 316 while (mBuffer.frameCount == 0) { 317 mBuffer.frameCount = inFrameCount; 324 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount); 325 if (mBuffer.frameCount > inputIndex) break; 327 inputIndex -= mBuffer.frameCount; 328 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; 329 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; 331 // mBuffer.frameCount [all...] |
H A D | AudioResamplerCubic.cpp | 66 if (mBuffer.frameCount == 0) { 67 mBuffer.frameCount = inFrameCount; 71 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); 94 if (inputIndex == mBuffer.frameCount) { 97 mBuffer.frameCount = inFrameCount; 103 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 132 if (mBuffer.frameCount == 0) { 133 mBuffer.frameCount = inFrameCount; 137 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 160 if (inputIndex == mBuffer.frameCount) { [all...] |
H A D | Tracks.cpp | 68 size_t frameCount, 85 mFrameCount(frameCount), 112 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 137 mCblk->frameCount_ = frameCount; 202 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 207 buf.mFrameCount = buffer->frameCount; 209 buffer->frameCount = 0; 326 size_t frameCount, 331 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffe 62 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, int clientUid, bool isOut) argument 319 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, const sp<IMemory>& sharedBuffer, int sessionId, int uid, IAudioFlinger::track_flags_t flags) argument 972 create( 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, const sp<IMemory>& sharedBuffer, int sessionId, int uid) argument 991 TimedTrack( 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, const sp<IMemory>& sharedBuffer, int sessionId, int uid) argument 1104 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() local 1490 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int uid) argument 1752 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int sessionId, int uid) argument [all...] |
H A D | AudioMixer.cpp | 64 mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount; 66 mDownmixConfig.outputCfg.buffer.frameCount = pBuffer->frameCount; 100 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument 122 mState.frameCount = frameCount; 200 // t->frameCount 210 // t->buffer.frameCount 509 int32_t volInc = d / int32_t(mState.frameCount); 813 volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 856 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 885 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 976 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument [all...] |
H A D | RecordTracks.h | 30 size_t frameCount,
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H A D | AudioMixer.h | 40 AudioMixer(size_t frameCount, uint32_t sampleRate, 173 uint16_t frameCount; member in struct:android::AudioMixer::track_t 220 size_t frameCount; member in struct:android::AudioMixer::state_t 278 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 280 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
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/frameworks/av/media/libmedia/ |
H A D | AudioTrack.cpp | 39 size_t* frameCount, 43 if (frameCount == NULL) { 48 *frameCount = 0; 75 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 78 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 98 int frameCount, 114 frameCount, flags, cbf, user, notificationFrames, 140 0 /*frameCount*/, flags, cbf, user, notificationFrames, 221 size_t frameCount = frameCountInt; local 226 ALOGV("set() streamType %d frameCount 38 getMinFrameCount( size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) argument 93 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_output_flags_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid) argument 845 createTrack_l( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, size_t frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, size_t epoch) argument [all...] |
H A D | AudioRecord.cpp | 35 size_t* frameCount, 40 if (frameCount == NULL) { 45 *frameCount = 0; 67 *frameCount = size; 84 int frameCount, 96 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 164 size_t frameCount = frameCountInt; local 166 ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, 167 frameCount); 225 if (frameCount 34 getMinFrameCount( size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument 79 AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, callback_t cbf, void* user, int notificationFrames, int sessionId, transfer_type transferType, audio_input_flags_t flags) argument [all...] |
H A D | IAudioFlingerClient.cpp | 58 data.writeInt32(desc->frameCount); 88 desc.frameCount = data.readInt32();
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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/native/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/av/libvideoeditor/lvpp/ |
H A D | VideoEditorSRC.cpp | 202 ALOGV("getNextBuffer %d, chan = %d", pBuffer->frameCount, mChannelCnt); 204 uint32_t want = pBuffer->frameCount * mChannelCnt * 2; 226 pBuffer->frameCount = 0; 280 pBuffer->frameCount = done / (mChannelCnt * 2); 281 ALOGV("getNextBuffer done %d", pBuffer->frameCount); 290 pBuffer->frameCount = 0;
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/frameworks/av/media/libeffects/preprocessing/ |
H A D | PreProcessing.cpp | 107 size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount) member in struct:preproc_session_s 798 session->frameCount = session->apmFrameCount; 931 session->frameCount = session->apmFrameCount; 933 session->frameCount = (session->apmFrameCount * session->samplingRate) / 1183 // inBuffer->frameCount, session->enabledMsk, session->processedMsk); 1187 size_t framesRq = outBuffer->frameCount; 1191 if (outBuffer->frameCount < fr) { 1192 fr = outBuffer->frameCount; 1203 outBuffer->frameCount = framesWr; 1205 inBuffer->frameCount [all...] |