/frameworks/av/media/img_utils/include/img_utils/ |
H A D | EndianUtils.h | 5 * 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 [all...] |
/frameworks/av/media/libeffects/lvm/wrapper/Bundle/ |
H A D | EffectBundle.cpp | 6 * 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 [all...] |
/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioBiquadFilter.cpp | 6 ** 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 [all...] |
H A D | AudioPeakingFilter.h | 6 ** 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 [all...] |
H A D | AudioShelvingFilter.h | 6 ** 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 [all...] |
H A D | EffectsMath.c | 5 * 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 [all...] |
/frameworks/av/media/libmediaplayerservice/ |
H A D | TestPlayerStub.h | 5 * 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);
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/frameworks/av/media/libmediaplayerservice/nuplayer/ |
H A D | NuPlayer.cpp | 5 * 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 [all...] |
H A D | NuPlayerDriver.cpp | 5 * 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
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | block_switch.c | 5 ** 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 [all...] |
/frameworks/av/media/libstagefright/codecs/amrnb/common/src/ |
H A D | vad1.cpp | 5 * 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 [all...] |
/frameworks/av/media/libstagefright/codecs/amrnb/dec/src/ |
H A D | agc.cpp | 5 * 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 [all...] |
/frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
H A D | wb_vad.c | 5 ** 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 [all...] |
/frameworks/av/media/libstagefright/codecs/avc/enc/src/ |
H A D | motion_comp.cpp | 5 * 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 [all...] |
/frameworks/av/media/libstagefright/codecs/g711/dec/ |
H A D | SoftG711.cpp | 5 * 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
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/frameworks/av/media/libstagefright/codecs/gsm/dec/ |
H A D | SoftGSM.cpp | 5 * 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
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/frameworks/av/media/libstagefright/codecs/mp3dec/src/ |
H A D | pvmp3_imdct_synth.cpp | 5 * 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 [all...] |
/frameworks/av/services/audioflinger/ |
H A D | AudioFlinger.cpp | 6 ** 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 [all...] |
H A D | AudioFlinger.h | 6 ** 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 [all...] |
H A D | AudioMixer.cpp | 6 ** 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 [all...] |
H A D | AudioMixer.h | 6 ** 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 [all...] |
H A D | AudioMixerOps.h | 5 * 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 [all...] |
H A D | AudioResampler.cpp | 5 * 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 [all...] |
H A D | AudioResamplerCubic.cpp | 5 * 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 [all...] |
H A D | AudioResamplerCubic.h | 5 * 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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