Searched refs:frameCount (Results 1 - 25 of 54) sorted by relevance

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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) {
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
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H A DAudioBufferProviderSource.cpp46 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)) {
/frameworks/av/include/media/
H A DAudioBufferProvider.h30 Buffer() : raw(NULL), frameCount(0) { }
36 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer
H A DAudioRecord.h66 size_t frameCount; member in class:android::AudioRecord::Buffer
67 size_t size; // total size in bytes == frameCount * frameSize
101 static status_t getMinFrameCount(int* frameCount,
123 * frameCount: Total size of track PCM buffer in frames. This defines the
137 int frameCount = 0,
162 int frameCount = 0,
187 uint32_t frameCount() const;
281 /* obtains a buffer of "frameCount" frames. The buffer must be
347 int frameCount,
H A DAudioTrack.h77 size_t frameCount; // number of sample frames corresponding to size; member in class:android::AudioTrack::Buffer
117 static status_t getMinFrameCount(int* frameCount,
139 * frameCount: Minimum size of track PCM buffer in frames. This defines the
158 int frameCount = 0,
170 int frameCount = 0,
214 int frameCount = 0,
242 uint32_t frameCount() const;
411 /* Obtains a buffer of "frameCount" frames. The buffer must be
434 /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
489 int frameCount,
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H A DAudioSystem.h91 static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
100 int* frameCount);
111 static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
155 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {}
160 size_t frameCount; member in class:android::AudioSystem::OutputDescriptor
/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/av/services/audioflinger/
H A DAudioResampler.cpp232 mBuffer.frameCount = 0;
279 mBuffer.frameCount = 0;
320 while (mBuffer.frameCount == 0) {
321 mBuffer.frameCount = inFrameCount;
328 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
329 if (mBuffer.frameCount > inputIndex) break;
331 inputIndex -= mBuffer.frameCount;
332 mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
333 mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
335 // mBuffer.frameCount
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H A DAudioResamplerCubic.cpp66 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) {
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H A DAudioMixer.cpp63 mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount;
65 mDownmixConfig.outputCfg.buffer.frameCount = pBuffer->frameCount;
99 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument
113 mState.frameCount = frameCount;
187 // t->frameCount
198 // t->buffer.frameCount
497 int32_t volInc = d / int32_t(mState.frameCount);
800 volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
842 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
870 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
960 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument
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H A DAudioMixer.h39 AudioMixer(size_t frameCount, uint32_t sampleRate,
164 uint16_t frameCount; member in struct:android::AudioMixer::track_t
211 size_t frameCount; member in struct:android::AudioMixer::state_t
261 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
262 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
H A DAudioResamplerSinc.cpp293 while (mBuffer.frameCount == 0) {
294 mBuffer.frameCount = inFrameCount;
308 if (inputIndex >= mBuffer.frameCount) {
309 inputIndex -= mBuffer.frameCount;
317 const size_t frameCount = mBuffer.frameCount; local
336 if (inputIndex >= frameCount)
341 if (inputIndex >= frameCount)
346 if (inputIndex >= frameCount)
354 if (inputIndex >= frameCount) {
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/frameworks/av/media/libmedia/
H A DAudioTrack.cpp53 int* frameCount,
57 if (frameCount == NULL) return BAD_VALUE;
60 *frameCount = 0;
85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
88 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
107 int frameCount,
119 frameCount, flags, cbf, user, notificationFrames,
129 int frameCount,
141 frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
162 0 /*frameCount*/, flag
52 getMinFrameCount( int* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) argument
102 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) argument
124 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
186 set( 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, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId) argument
352 uint32_t AudioTrack::frameCount() const function in class:android::AudioTrack
749 createTrack_l( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output) argument
1514 stepUser(uint32_t frameCount) argument
1552 stepServer(uint32_t frameCount) argument
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H A DAudioRecord.cpp39 int* frameCount,
44 if (frameCount == NULL) return BAD_VALUE;
47 *frameCount = 0;
70 *frameCount = size;
87 int frameCount,
96 frameCount, cbf, user, notificationFrames, sessionId);
122 int frameCount,
130 ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount);
186 if (frameCount
38 getMinFrameCount( int* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument
82 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) argument
117 set( 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, bool threadCanCallJava, int sessionId) argument
255 uint32_t AudioRecord::frameCount() const function in class:android::AudioRecord
425 openRecord_l( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_io_handle_t input) argument
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H A DIAudioFlingerClient.cpp57 data.writeInt32(desc->frameCount);
87 desc.frameCount = data.readInt32();
/frameworks/av/include/private/media/
H A DAudioTrackShared.h76 uint32_t frameCount; member in struct:android::audio_track_cblk_t
117 uint32_t stepUser(uint32_t frameCount); // called by client only, where
119 bool stepServer(uint32_t frameCount); // called by server only
/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/native/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/av/libvideoeditor/lvpp/
H A DVideoEditorSRC.cpp202 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;
/frameworks/av/media/libeffects/preprocessing/
H A DPreProcessing.cpp107 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
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/frameworks/av/media/libeffects/lvm/wrapper/Bundle/
H A DEffectBundle.h97 int frameCount; member in struct:BundledEffectContext

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