/frameworks/av/media/libstagefright/httplive/ |
H A D | HTTPDownloader.cpp | 70 * | | `source` handle => `out` buffer | | | | 81 const char *url, sp<ABuffer> *out, 136 sp<ABuffer> buffer = *out != NULL ? *out : new ABuffer(size); 137 if (*out == NULL) { 182 // to help us break out of the loop. 203 *out = buffer; 215 const char *url, sp<ABuffer> *out, String8 *actualUrl) { 216 ssize_t err = fetchBlock(url, out, 0, -1, 0, actualUrl, true /* reconnect */); 80 fetchBlock( const char *url, sp<ABuffer> *out, int64_t range_offset, int64_t range_length, uint32_t block_size, String8 *actualUrl, bool reconnect ) argument 214 fetchFile( const char *url, sp<ABuffer> *out, String8 *actualUrl) argument
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H A D | M3UParser.cpp | 477 static bool MakeURL(const char *baseURL, const char *url, AString *out) { argument 478 out->clear(); 491 out->setTo(url); 493 ALOGV("base:'%s', url:'%s' => '%s'", baseURL, url, out->c_str()); 505 out->setTo(baseURL, pathStart - baseURL); 507 out->setTo(baseURL); 510 out->append(url); 533 out->setTo(baseURL, end); 535 out->setTo(baseURL); 538 out [all...] |
/frameworks/av/media/libstagefright/matroska/ |
H A D | MatroskaExtractor.cpp | 185 // If we do detect out-of-order cues, return NULL. 638 MediaBuffer **out, const ReadOptions *options) { 639 *out = NULL; 679 *out = frame; 784 *out = buffer; 637 read( MediaBuffer **out, const ReadOptions *options) argument
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/frameworks/av/media/libstagefright/mpeg2ts/ |
H A D | AnotherPacketSource.cpp | 166 MediaBuffer **out, const ReadOptions *) { 167 *out = NULL; 225 *out = mediaBuffer; 165 read( MediaBuffer **out, const ReadOptions *) argument
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H A D | ESQueue.cpp | 849 AString out; local 873 out.append(", "); 875 out.append(tmp); 888 ALOGV("accessUnit contains nal types %s", out.c_str());
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H A D | MPEG2TSExtractor.cpp | 90 MediaBuffer **out, const ReadOptions *options) { 91 *out = NULL; 107 return mImpl->read(out, options); 89 read( MediaBuffer **out, const ReadOptions *options) argument
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/frameworks/av/media/libstagefright/omx/ |
H A D | OMXNodeInstance.cpp | 1454 const void *data, size_t size, T *out) { 1458 *out = *(T*)data; 1453 getInternalOption( const void *data, size_t size, T *out) argument
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | AMPEG4AudioAssembler.cpp | 70 uint8_t *out = buffer->data(); local 89 *out++ = accum; 365 sp<ABuffer> out = new ABuffer(buffer->size()); local 366 out->setRange(0, 0); 384 return out; 414 memcpy(out->data() + out->size(), &ptr[offset], payloadLength); 415 out->setRange(0, out->size() + payloadLength); 433 return out; [all...] |
H A D | APacketSource.cpp | 79 uint8_t *out = buffer->data(); local 98 *out++ = accum; 174 uint8_t *out = csd->data(); local 176 *out++ = 0x01; // configurationVersion 178 memcpy(out, profileLevelID->data(), 3); 179 out += 3; 181 *out++ = 0x42; // Baseline profile 182 *out++ = 0xE0; // Common subset for all profiles 183 *out++ = 0x0A; // Level 1 186 *out [all...] |
H A D | ARTSPConnection.cpp | 374 // Timed out. Not yet connected. 917 static void H(const AString &s, AString *out) { argument 918 out->clear(); 934 out->append(&nibble, 1); 942 out->append(&nibble, 1); 973 AString out; local 974 encodeBase64(tmp.c_str(), tmp.size(), &out); 978 fragment.append(out);
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H A D | MyHandler.h | 1734 static bool MakeURL(const char *baseURL, const char *url, AString *out) { argument 1735 out->clear(); 1744 out->setTo(url); 1749 out->setTo(baseURL); 1751 out->append("/"); 1753 out->append(url);
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H A D | MyTransmitter.h | 239 void H(const AString &s, AString *out) { argument 240 out->clear(); 256 out->append(&nibble, 1); 264 out->append(&nibble, 1);
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/frameworks/av/media/libstagefright/wifi-display/rtp/ |
H A D | RTPSender.cpp | 318 sp<ABuffer> out = new ABuffer(kMaxUDPPacketSize); local 333 if (outBytesUsed + bytesNeeded > out->capacity()) { 337 && outBytesUsed + nalSize <= out->capacity()) { 342 memcpy(out->data() + outBytesUsed, nalStart, nalSize); 349 out->setRange(0, outBytesUsed); 350 packets.push_back(out); 351 out = new ABuffer(kMaxUDPPacketSize); 360 if (outBytesUsed + bytesNeeded <= out->capacity()) { 361 uint8_t *dst = out->data() + outBytesUsed; 385 size_t copy = out 420 sp<ABuffer> out = *packets.begin(); local [all...] |
/frameworks/av/media/ndk/ |
H A D | NdkMediaFormat.cpp | 153 bool AMediaFormat_getInt32(AMediaFormat* format, const char *name, int32_t *out) { argument 154 return format->mFormat->findInt32(name, out); 158 bool AMediaFormat_getInt64(AMediaFormat* format, const char *name, int64_t *out) { argument 159 return format->mFormat->findInt64(name, out); 163 bool AMediaFormat_getFloat(AMediaFormat* format, const char *name, float *out) { argument 164 return format->mFormat->findFloat(name, out); 168 bool AMediaFormat_getSize(AMediaFormat* format, const char *name, size_t *out) { argument 169 return format->mFormat->findSize(name, out); 184 bool AMediaFormat_getString(AMediaFormat* mData, const char *name, const char **out) { argument 197 *out [all...] |
/frameworks/av/services/audioflinger/ |
H A D | AudioFlinger.cpp | 150 goto out; 156 goto out; 161 goto out; 165 out: 1252 break; // retries failed, break out of loop with frames == 0. 2020 AudioStreamOut *out = thread->clearOutput(); local 2021 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2023 out->hwDev()->close_output_stream(out [all...] |
H A D | AudioMixer.cpp | 243 ALOGE("AudioMixer::getTrackName out of available tracks"); 1075 void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, argument 1089 volumeRampStereo(t, out, outFrameCount, temp, aux); 1091 volumeStereo(t, out, outFrameCount, temp, aux); 1098 volumeRampStereo(t, out, outFrameCount, temp, aux); 1104 t->resampler->resample(out, outFrameCount, t->bufferProvider); 1109 void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused, 1114 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, argument 1136 *out++ += (vl >> 16) * l; 1137 *out 1157 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 1186 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1278 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1448 int32_t *out = t1.mainBuffer; local 1542 int32_t *out = t1.mainBuffer; local 1599 int32_t* out = t.mainBuffer; local 1692 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument 1736 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument 1776 volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t) argument 1822 TO* out = reinterpret_cast<TO*>(t->mainBuffer); local 1871 track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux) argument 1902 track__NoResample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument 1921 convertMixerFormat(void *out, audio_format_t mixerOutFormat, void *in, audio_format_t mixerInFormat, size_t sampleCount) argument [all...] |
H A D | AudioMixerOps.h | 244 * This accumulates into the out pointer. 254 * This accumulates into the out pointer. 263 * MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer. 275 inline void volumeRampMulti(TO* out, size_t frameCount, argument 287 *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum); 293 *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum); 300 *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum); 306 *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[0], &auxaccum); 312 *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[0], &auxaccum); 329 *out 368 volumeMulti(TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument [all...] |
H A D | AudioResamplerCubic.cpp | 35 size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, argument 44 return resampleMono16(out, outFrameCount, provider); 46 return resampleStereo16(out, outFrameCount, provider); 53 size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, argument 82 out[outputIndex++] += vl * interp(&left, x); 83 out[outputIndex++] += vr * interp(&right, x); 84 // out[outputIndex++] += vr * in[inputIndex*2]; 120 size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, argument 151 out[outputIndex++] += vl * sample; 152 out[outputInde [all...] |
H A D | AudioResamplerDyn.cpp | 479 size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount, 482 return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider); 487 size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, argument 490 // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out. 573 &out[outputIndex], 591 // We arrive here when we're finished or when the input buffer runs out. 571 ALOG_ASSERT(phaseFraction < phaseWrapLimit); fir<CHANNELS, LOCKED, STRIDE>( &out[outputIndex], phaseFraction, phaseWrapLimit, coefShift, halfNumCoefs, coefs, impulse, volumeSimd); outputIndex += OUTPUT_CHANNELS; phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) argument
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H A D | AudioResamplerFirOps.h | 55 int32_t out; local 57 asm( "smultb %[out], %[in], %[vRL] \n" 58 : [out]"=r"(out) 62 asm( "smultt %[out], %[in], %[vRL] \n" 63 : [out]"=r"(out) 67 return out; 78 int32_t out; local 79 asm( "smlabb %[out], 93 int32_t out; local 108 int32_t out; local 123 int32_t out; local 146 int32_t out; local [all...] |
H A D | AudioResamplerFirProcess.h | 92 inline void volume(TO*& out, TO gain) { argument 93 *out++ = volumeAdjust(value, gain); 94 Accumulator<CHANNELS-1, TO>::volume(out, gain); 108 inline void volume(TO*& out __unused, TO gain __unused) { 180 void ProcessBase(TO* const out, argument 215 // for (int j = 0; j < CHANNELS; ++j) out[j] += volumeAdjust(accum[j], volumeLR[0]); 216 TO *tmp_out = out; // may remove if const out definition changes. 229 out[0] += volumeAdjust(l, volumeLR[0]); 230 out[ 251 ProcessL(TO* const out, int count, const TC* coefsP, const TC* coefsN, const TI* sP, const TI* sN, const TO* const volumeLR) argument 297 Process(TO* const out, int count, const TC* coefsP, const TC* coefsN, const TC* coefsP1 __unused, const TC* coefsN1 __unused, const TI* sP, const TI* sN, TINTERP lerpP, const TO* const volumeLR) argument 376 fir(TO* const out, const uint32_t phase, const uint32_t phaseWrapLimit, const int coefShift, const int halfNumCoefs, const TC* const coefs, const TI* const samples, const TO* const volumeLR) argument [all...] |
H A D | AudioResamplerFirProcessNeon.h | 52 // Macros to save a mono/stereo accumulator sample in q0 (and q4) as stereo out. 56 "vld1.s32 {d3}, %[out] \n"/* (2) unaligned load the output */\ 61 "vst1.s32 {d3}, %[out] \n"/* (2+2d) store result */ 65 "vld1.s32 {d3}, %[out] \n"/* (2) unaligned load the output*/\ 71 "vst1.s32 {d3}, %[out] \n"/* (2+2d)store result*/ 74 static inline void ProcessNeonIntrinsic(int32_t* out, argument 165 int32x2_t outSamp = vld1_s32(out); 178 vst1_s32(out, outSamp); 182 static inline void ProcessNeonIntrinsic(int32_t* out, argument 340 int32x2_t outSamp = vld1_s32(out); 357 ProcessNeonIntrinsic(float* out, int count, const float* coefsP, const float* coefsN, const float* sP, const float* sN, const float* volumeLR, float lerpP, const float* coefsP1, const float* coefsN1) argument 526 ProcessL(int32_t* const out, int count, const int16_t* coefsP, const int16_t* coefsN, const int16_t* sP, const int16_t* sN, const int32_t* const volumeLR) argument 583 ProcessL(int32_t* const out, int count, const int16_t* coefsP, const int16_t* coefsN, const int16_t* sP, const int16_t* sN, const int32_t* const volumeLR) argument 646 Process(int32_t* const out, int count, const int16_t* coefsP, const int16_t* coefsN, const int16_t* coefsP1, const int16_t* coefsN1, const int16_t* sP, const int16_t* sN, uint32_t lerpP, const int32_t* const volumeLR) argument 722 Process(int32_t* const out, int count, const int16_t* coefsP, const int16_t* coefsN, const int16_t* coefsP1, const int16_t* coefsN1, const int16_t* sP, const int16_t* sN, uint32_t lerpP, const int32_t* const volumeLR) argument 803 ProcessL(int32_t* const out, int count, const int32_t* coefsP, const int32_t* coefsN, const int16_t* sP, const int16_t* sN, const int32_t* const volumeLR) argument 869 ProcessL(int32_t* const out, int count, const int32_t* coefsP, const int32_t* coefsN, const int16_t* sP, const int16_t* sN, const int32_t* const volumeLR) argument 954 Process(int32_t* const out, int count, const int32_t* coefsP, const int32_t* coefsN, const int32_t* coefsP1, const int32_t* coefsN1, const int16_t* sP, const int16_t* sN, uint32_t lerpP, const int32_t* const volumeLR) argument 1044 Process(int32_t* const out, int count, const int32_t* coefsP, const int32_t* coefsN, const int32_t* coefsP1, const int32_t* coefsN1, const int16_t* sP, const int16_t* sN, uint32_t lerpP, const int32_t* const volumeLR) argument 1153 ProcessL(float* const out, int count, const float* coefsP, const float* coefsN, const float* sP, const float* sN, const float* const volumeLR) argument 1166 ProcessL(float* const out, int count, const float* coefsP, const float* coefsN, const float* sP, const float* sN, const float* const volumeLR) argument 1179 Process(float* const out, int count, const float* coefsP, const float* coefsN, const float* coefsP1, const float* coefsN1, const float* sP, const float* sN, float lerpP, const float* const volumeLR) argument 1195 Process(float* const out, int count, const float* coefsP, const float* coefsN, const float* coefsP1, const float* coefsN1, const float* sP, const float* sN, float lerpP, const float* const volumeLR) argument [all...] |
H A D | AudioResamplerSinc.cpp | 150 int32_t out; local 152 asm( "smultb %[out], %[in], %[vRL] \n" 153 : [out]"=r"(out) 157 asm( "smultt %[out], %[in], %[vRL] \n" 158 : [out]"=r"(out) 162 return out; 173 int32_t out; local 174 asm( "smlawb %[out], 188 int32_t out; local 267 resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) argument 294 resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) argument 400 filterCoefficient(int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL) argument [all...] |
H A D | Effects.cpp | 302 int16_t *out = mConfig.outputCfg.buffer.s16; local 304 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
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H A D | test-resample.cpp | 291 printf("getNextBuffer() requested %zu frames out of %zu frames available," 312 printf("releaseBuffer() released %zu frames out of %zu frames available " 421 time = diff_ns; // save the best out of our trials. 473 // mono takes left channel only (out of stereo output pair) 475 int32_t* out = (int32_t*) output_vaddr; local 484 int32_t s = out[i * output_channels + j] + roundVal; // add offset here
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