Searched defs:in (Results 1 - 25 of 354) sorted by path

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/frameworks/av/media/img_utils/include/img_utils/
H A DEndianUtils.h5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
44 T convertToBigEndian(T in);
50 T convertToLittleEndian(T in);
58 * Wrap the given Output. Calling write methods will result in
96 * Count elements in the buffer will be written with the endianness set for this
98 * be skipped in the buffer before writing.
167 inline uint8_t convertToBigEndian(uint8_t in) { argument
168 return in;
172 inline int8_t convertToBigEndian(int8_t in) { argument
177 convertToBigEndian(uint16_t in) argument
182 convertToBigEndian(int16_t in) argument
187 convertToBigEndian(uint32_t in) argument
192 convertToBigEndian(int32_t in) argument
197 convertToBigEndian(uint64_t in) argument
202 convertToBigEndian(int64_t in) argument
207 convertToLittleEndian(uint8_t in) argument
212 convertToLittleEndian(int8_t in) argument
217 convertToLittleEndian(uint16_t in) argument
222 convertToLittleEndian(int16_t in) argument
227 convertToLittleEndian(uint32_t in) argument
232 convertToLittleEndian(int32_t in) argument
237 convertToLittleEndian(uint64_t in) argument
242 convertToLittleEndian(int64_t in) argument
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/frameworks/av/media/libeffects/lvm/wrapper/Bundle/
H A DEffectBundle.cpp6 * you may not use this file except in compliance with the License.
11 * Unless required by applicable law or agreed to in writing, software
38 "null pointer returned by %s in %s\n\n\n\n", callingFunc, calledFunc);\
42 "bad alignment returned by %s in %s\n\n\n\n", callingFunc, calledFunc);\
46 "bad number of samples returned by %s in %s\n\n\n\n", callingFunc, calledFunc);\
50 "out of range returned by %s in %s\n", callingFunc, calledFunc);\
207 // If this is the first create in this session
209 ALOGV("\tEffectCreate - This is the first effect in current sessionId %d sessionNo %d",
254 /* Saved strength is used to return the exact strength that was used in the set to the get
395 // Disable effect, in thi
2838 LVM_INT16 *in = (LVM_INT16 *)inBuffer->raw; local
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/frameworks/av/media/libeffects/testlibs/
H A DAudioBiquadFilter.cpp6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
65 void AudioBiquadFilter::process(const audio_sample_t in[], audio_sample_t out[], argument
67 (this->*mCurProcessFunc)(in, out, frameCount);
139 void AudioBiquadFilter::process_bypass(const audio_sample_t * in, argument
142 // The common case is in-place processing, because this is what the EQ does.
143 if (CC_UNLIKELY(in != out)) {
144 memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
148 void AudioBiquadFilter::process_normal_mono(const audio_sample_t * in, argument
162 audio_sample_t x0 = *(in
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 DAudioPeakingFilter.h6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
29 // in the output, even when changing parameters abruptly.
43 // sampleRate The input/output sample rate, in Hz.
49 // sampleRate The input/output sample rate, in Hz.
63 // This value will be remembered even if the filter is in disabled() state.
64 // millibel Gain value in millibel (1/100 of decibel).
67 // Gets the gain, in millibel, as set.
71 // This value will be remembered even if the filter is in disabled() state.
72 // cents Bandwidth value in cent
104 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
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H A DAudioShelvingFilter.h6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
29 // in the output, even when changing parameters abruptly.
50 // sampleRate The input/output sample rate, in Hz.
56 // sampleRate The input/output sample rate, in Hz.
69 // This value will be remembered even if the filter is in disabled() state.
70 // millibel Gain value in millibel (1/100 of decibel).
73 // Gets the gain, in millibel, as set.
78 // This value will be remembered even if the filter is in disabled() state.
79 // millihertz Frequency value in mH
98 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
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H A DEffectsMath.c5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
24 // for integers in the range 0 to 63 (i = ai5*2^5 + ai4*2^4 + ai3*2^3 + ai2*2^2 + ai1*2^1 + ai0*2^0)
111 int32_t Effects_Sqrt(int32_t in) argument
119 if (in == 0) return 0;
121 if (in >= 0x10000000)
124 in -= 0x10000000;
127 j = 32 - __builtin_clz(in);
134 if (in >= tmp)
137 in
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/frameworks/av/media/libmediaplayerservice/
H A DTestPlayerStub.h5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
24 class MediaPlayerBase; // in media/MediaPlayerInterface.h
36 // TestPlayerStub::setDataSource loads the library in the test url. 2
46 // typical usage in a java test:
98 virtual status_t invoke(const android::Parcel& in, android::Parcel *out) { argument
99 return mPlayer->invoke(in, out);
/frameworks/av/media/libmediaplayerservice/nuplayer/
H A DNuPlayer.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
145 // Use this if there's no state necessary to save in order to execute
253 // The correct flags will be updated in Source::kWhatFlagsChanged
531 msg->post(1000000ll); // poll again in a second.
559 // If the video decoder is not set (perhaps audio only in this case)
1110 // Audio decoder is no longer needed if it's in shut/shutting down status.
1147 // decoder flush completes only occur in a flushing state.
1148 LOG_ALWAYS_FATAL_IF(isDecoder, "decoder flush in invalid state %d", *state);
1187 // Note: This is called early in NuPlaye
1358 notifyListener(int msg, int ext1, int ext2, const Parcel *in) argument
1922 Parcel in; local
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H A DNuPlayerDriver.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
636 int msg, int ext1, int ext2, const Parcel *in) {
638 notifyListener_l(msg, ext1, ext2, in);
642 int msg, int ext1, int ext2, const Parcel *in) {
661 // The renderer has stopped the sink at the end in order to play out
685 sendEvent(msg, ext1, ext2, in);
716 // in response, NuPlayerDriver has the right state
635 notifyListener( int msg, int ext1, int ext2, const Parcel *in) argument
641 notifyListener_l( int msg, int ext1, int ext2, const Parcel *in) argument
/frameworks/av/media/libstagefright/codecs/aacenc/src/
H A Dblock_switch.c5 ** you may not use this file except in compliance with the License.
10 ** Unless required by applicable law or agreed to in writing, software
34 IIRFilter(const Word16 in, const Word32 coeff[], Word32 states[]);
37 SrchMaxWithIndex(const Word32 *in, Word16 *index, Word16 n);
92 /* Attack in Window 0 */ {1, 3, 3, 1},
93 /* Attack in Window 1 */ {1, 1, 3, 3},
94 /* Attack in Window 2 */ {2, 1, 3, 2},
95 /* Attack in Window 3 */ {3, 1, 3, 1},
96 /* Attack in Window 4 */ {3, 1, 1, 3},
97 /* Attack in Windo
249 SrchMaxWithIndex(const Word32 in[], Word16 *index, Word16 n) argument
339 IIRFilter(const Word16 in, const Word32 coeff[], Word32 states[]) argument
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/frameworks/av/media/libstagefright/codecs/amrnb/common/src/
H A Dvad1.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
50 round function in C standard library.
104 in -- array of type Word16 -- input signal
167 Word16 in[], /* i : input signal */
189 temp1 = shr(in[4*i+0], 2, pOverflow);
196 temp2 = shr(in[4*i+1], 2, pOverflow);
207 temp2 = shr(in[4*i+2], 2, pOverflow);
214 temp2 = shr(in[4*i+3], 2, pOverflow);
465 Purpose : Calculate signal level in
166 first_filter_stage( Word16 in[], Word16 out[], Word16 data[], Flag *pOverflow ) argument
613 filter_bank( vadState1 *st, Word16 in[], Word16 level[], Flag *pOverflow ) argument
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/frameworks/av/media/libstagefright/codecs/amrnb/dec/src/
H A Dagc.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
91 in = input signal (Word16)
126 Word16 in[], // i : input signal (length l_trm)
133 temp = shr (in[0], 2);
138 temp = shr (in[i], 2);
169 Word16 in[], /* i : input signal (length l_trm) */
181 temp = in[i] >> 2;
196 in = input signal (Word16)
229 CALL energy_old ( in
168 energy_old( Word16 in[], Word16 l_trm, Flag *pOverflow ) argument
258 energy_old_Wrapper(Word16 in[], Word16 l_trm, Flag *pOverflow) argument
370 energy_new( Word16 in[], Word16 l_trm, Flag *pOverflow ) argument
478 energy_new_Wrapper(Word16 in[], Word16 l_trm, Flag *pOverflow) argument
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/frameworks/av/media/libstagefright/codecs/amrwbenc/src/
H A Dwb_vad.c5 ** you may not use this file except in compliance with the License.
10 ** Unless required by applicable law or agreed to in writing, software
37 * ilog2(Word32 in) = -1024*log10(in * 2^-31)/log10(2), where in = [1, 2^31-1]
43 * When input is in the range of [1,2^16], max error is 0.0380%.
128 * Purpose : Calculate signal level in a sub-band. Level is calculated
174 * the signal in each band
180 Word16 in[], /* i : input frame */
190 tmp_buf[i] = in[
178 filter_bank( VadVars * st, Word16 in[], Word16 level[] ) argument
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/frameworks/av/media/libstagefright/codecs/avc/enc/src/
H A Dmotion_comp.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
268 void eHorzInterp1MC(uint8 *in, int inpitch, uint8 *out, int outpitch, argument
281 p_ref = in;
286 dx = ((dx >> 1) ? -3 : -4); /* use in 3/4 pel */
558 void eHorzInterp2MC(int *in, int inpitch, uint8 *out, int outpitch, argument
569 p_ref = in;
574 dx = ((dx >> 1) ? -3 : -4); /* use in 3/4 pel */
717 void eHorzInterp3MC(uint8 *in, int inpitch, int *out, int outpitch, argument
727 p_ref = in;
781 eVertInterp1MC(uint8 *in, int inpitch, uint8 *out, int outpitch, int blkwidth, int blkheight, int dy) argument
1083 eVertInterp2MC(uint8 *in, int inpitch, int *out, int outpitch, int blkwidth, int blkheight) argument
1148 eVertInterp3MC(int *in, int inpitch, uint8 *out, int outpitch, int blkwidth, int blkheight, int dy) argument
1649 eFullPelMC(uint8 *in, int inpitch, uint8 *out, int outpitch, int blkwidth, int blkheight) argument
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/frameworks/av/media/libstagefright/codecs/g711/dec/
H A DSoftG711.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
261 int16_t *out, const uint8_t *in, size_t inSize) {
263 int32_t x = *in++;
287 int16_t *out, const uint8_t *in, size_t inSize) {
289 int32_t x = *in++;
260 DecodeALaw( int16_t *out, const uint8_t *in, size_t inSize) argument
286 DecodeMLaw( int16_t *out, const uint8_t *in, size_t inSize) argument
/frameworks/av/media/libstagefright/codecs/gsm/dec/
H A DSoftGSM.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
243 int16_t *out, uint8_t *in, size_t inSize) {
247 gsm_decode(handle, in, out);
248 in += 33;
252 gsm_decode(handle, in, out);
253 in += 32;
242 DecodeGSM(gsm handle, int16_t *out, uint8_t *in, size_t inSize) argument
/frameworks/av/media/libstagefright/codecs/mp3dec/src/
H A Dpvmp3_imdct_synth.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
38 int32 in[], Pointer to spec values of current channel
46 int32 in[],
204 ; Declare functions defined elsewhere and referenced in this module
209 ; Declare variables used in this module but defined elsewhere
216 void pvmp3_imdct_synth(int32 in[SUBBANDS_NUMBER*FILTERBANK_BANDS], argument
234 * in case of mx_poly_band> 0, do
243 int32 * out = in + (band * FILTERBANK_BANDS);
334 int32 * out = in
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/frameworks/av/services/audioflinger/
H A DAudioFlinger.cpp6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
462 // If pid is already in the mClients wp<> map, then use that entry
738 // Set master volume in the HALs which support it.
750 // Now set the master volume in each playback thread. Playback threads
850 // Set master mute in the HALs which support it.
862 // Now set the master mute in each playback thread. Playback threads
1064 // hold a strong ref on thread in case closeOutput() or closeInput() is called
1305 // FIXME should be a "k" constant not hard-coded, in
2160 AudioStreamIn *in = thread->clearInput(); local
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H A DAudioFlinger.h6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
80 // this #define. There is an independent hard-coded upper limit in AudioMixer;
100 // IAudioFlinger interface, in binder opcode order
323 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
339 // Returns true if channel mask is permitted for the PCM sink in the MixerThread
359 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
362 // Returns true if format is permitted for the PCM sink in the MixerThread
492 virtual void signal(); // signal playback thread for a change in control block
575 // effect chain and same instances in th
646 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) argument
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H A DAudioMixer.cpp6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
81 // Set to default copy buffer size in frames for input processing.
193 bufferFrameCount) // set bufferFrameCount to 0 to do in-place
291 // may be in-place if src == dst.
297 /* call once in a pthread_once handler. */
344 // TODO: consider channel representation in index array formulation
379 // The value of 1 << x is undefined in C when x >= 32.
486 t->in = NULL;
731 * even if there is a nonzero floating point increment (in tha
1325 const int16_t *in = static_cast<const int16_t *>(t->in); local
1417 const int16_t *in = static_cast<int16_t const *>(t->in); local
1752 const int16_t *in = b.i16; local
1849 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument
1893 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument
1933 volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t) argument
1989 const TI *in = reinterpret_cast<TI*>(b.raw); local
2064 const TI *in = static_cast<const TI *>(t->in); local
2079 convertMixerFormat(void *out, audio_format_t mixerOutFormat, void *in, audio_format_t mixerInFormat, size_t sampleCount) argument
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H A DAudioMixer.h6 ** you may not use this file except in compliance with the License.
11 ** Unless required by applicable law or agreed to in writing, software
33 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12
87 // parameter 'value' is the new sample rate in Hz.
198 const void* in; // current location in buffer member in struct:android::AudioMixer::track_t
259 // It handles a private buffer for use in converting format or channel masks from the
266 // If bufferFrameCount is 0, no private buffer is created and in-place modification of
283 // #frames in th
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H A DAudioMixerOps.h5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
44 * There are 27 variants, of which 14 are actually defined in an
140 * are not needed in execution and should be removed from the final build by
225 /* MIXTYPE is used to determine how the samples in the input frame
280 const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
291 *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
297 *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum);
300 in++;
304 *out++ = MixMulAux<TO, TI, TV, TA>(*in
279 volumeRampMulti(TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument
372 volumeMulti(TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument
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H A DAudioResampler.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
50 // number of bits used in interpolation multiply - 15 bits avoids overflow
62 void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
65 void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
389 int16_t *in = mBuffer.i16;
394 out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
395 out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
412 AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
418 out[outputIndex++] += vl * Interp(in[inputInde
484 int16_t *in = mBuffer.i16; member in namespace:android
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H A DAudioResamplerCubic.cpp5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
74 int16_t *in = mBuffer.i16; local
84 // out[outputIndex++] += vr * in[inputIndex*2];
104 in = mBuffer.i16;
109 advance(&left, in[inputIndex*2]);
110 advance(&right, in[inputIndex*2+1]);
142 int16_t *in = mBuffer.i16; local
173 in = mBuffer.i16;
177 advance(&left, in[inputInde
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H A DAudioResamplerCubic.h5 * you may not use this file except in compliance with the License.
10 * Unless required by applicable law or agreed to in writing, software
37 // number of bits used in interpolation multiply - 14 bits avoids overflow
53 static inline void advance(state* p, int16_t in) { argument
57 p->y3 = in;

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