Searched refs:frameCount (Results 1 - 25 of 65) sorted by relevance

123

/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.h32 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
H A DAudioRecord.h60 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
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H A DAudioTrack.h52 // 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
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H A DAudioSystem.h95 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
/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/include/private/media/
H A DAudioTrackShared.h150 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
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/frameworks/av/services/audioflinger/
H A DAudioResampler.cpp228 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
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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 DTracks.cpp68 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
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H A DAudioMixer.cpp64 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
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H A DRecordTracks.h30 size_t frameCount,
H A DAudioMixer.h40 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,
/frameworks/av/media/libmedia/
H A DAudioTrack.cpp39 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
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H A DAudioRecord.cpp35 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
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H A DIAudioFlingerClient.cpp58 data.writeInt32(desc->frameCount);
88 desc.frameCount = data.readInt32();
/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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