/frameworks/base/media/jni/soundpool/ |
H A D | SoundPool.cpp | 497 uint32_t sampleRate; local 503 p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format); 505 p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format); 514 LOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d", 515 p->pointer(), p->size(), sampleRate, numChannels); 517 if (sampleRate > kMaxSampleRate) { 518 LOGE("Sample rate (%u) out of range", sampleRate); 533 mSampleRate = sampleRate; 581 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rat local 844 uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5); local [all...] |
H A D | SoundPool.h | 58 int sampleRate() { return mSampleRate; } function in class:android::Sample 68 void init(int numChannels, int sampleRate, int format, size_t size, sp<IMemory> data ) { argument 69 mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; mData = data; }
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/frameworks/media/libvideoeditor/lvpp/ |
H A D | VideoEditorPlayer.cpp | 382 uint32_t sampleRate, int channelCount, int format, int bufferCount, 395 LOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount); 410 frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate; 416 sampleRate, 427 sampleRate, 442 mMsecsPerFrame = 1.e3 / (float) sampleRate; 381 open( uint32_t sampleRate, int channelCount, int format, int bufferCount, AudioCallback cb, void *cookie) argument
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H A D | VideoEditorPlayer.h | 52 uint32_t sampleRate, int channelCount,
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/frameworks/base/include/media/stagefright/ |
H A D | ACodec.h | 153 status_t setupAACDecoder(int32_t numChannels, int32_t sampleRate); 158 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
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H A D | AudioSource.h | 37 int inputSource, uint32_t sampleRate,
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H A D | OMXCodec.h | 231 status_t setAACFormat(int32_t numChannels, int32_t sampleRate, int32_t bitRate); 271 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
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/frameworks/base/media/libstagefright/ |
H A D | AudioSource.cpp | 50 int inputSource, uint32_t sampleRate, uint32_t channels) 52 mSampleRate(sampleRate), 57 LOGV("sampleRate: %d, channels: %d", sampleRate, channels); 63 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 49 AudioSource( int inputSource, uint32_t sampleRate, uint32_t channels) argument
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H A D | AMRWriter.cpp | 91 int32_t sampleRate; local 94 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 95 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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H A D | AVIExtractor.cpp | 339 int sampleRate; local 342 header, &frameSize, &sampleRate, NULL, NULL, &numSamples)) { 353 int64_t timeUs = mBaseTimeUs + (mNumSamplesRead * 1000000ll) / sampleRate; 712 uint32_t sampleRate = U32LE_AT(&data[4]); local 715 track->mMeta->setInt32(kKeySampleRate, sampleRate);
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H A D | OMXCodec.cpp | 662 int32_t numChannels, sampleRate; local 664 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 666 status_t err = setAACFormat(numChannels, sampleRate, bitRate); 3437 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) { 3464 pcmParams.nSamplingRate = sampleRate; 3548 int32_t sampleRate; local 3550 CHECK(format->findInt32(kKeySampleRate, &sampleRate)); 3553 setRawAudioFormat(kPortIndexInput, sampleRate, numChannels); 3557 status_t OMXCodec::setAACFormat(int32_t numChannels, int32_t sampleRate, int32_t bitRate) { argument 3563 setRawAudioFormat(kPortIndexInput, sampleRate, numChannel 3436 setRawAudioFormat( OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) argument 4464 int32_t numChannels, sampleRate; local 4523 int32_t numChannels, sampleRate, bitRate; local [all...] |
/frameworks/base/media/libstagefright/codecs/aacenc/ |
H A D | AACEncoder.cpp | 84 params.sampleRate = mSampleRate; 96 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) { argument 103 if (sampleRate == kSampleRateTable[i]) { 109 LOGE("Sampling rate %d bps is not supported", sampleRate);
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/frameworks/base/media/libmediaplayerservice/nuplayer/ |
H A D | NuPlayerDecoder.cpp | 124 int32_t numChannels, sampleRate; local 126 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 129 msg->setInt32("sample-rate", sampleRate);
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H A D | NuPlayer.cpp | 330 int32_t sampleRate; local 331 CHECK(codecRequest->findInt32("sample-rate", &sampleRate)); 334 sampleRate, numChannels); 338 sampleRate,
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/frameworks/base/media/libmedia/ |
H A D | IAudioFlinger.cpp | 86 uint32_t sampleRate, 101 data.writeInt32(sampleRate); 133 uint32_t sampleRate, 146 data.writeInt32(sampleRate); 173 virtual uint32_t sampleRate(int output) const function in class:android::BpAudioFlinger 346 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 350 data.writeInt32(sampleRate); 680 uint32_t sampleRate = data.readInt32(); local 690 streamType, sampleRate, format, 701 uint32_t sampleRate local 83 createTrack( pid_t pid, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int output, int *sessionId, status_t *status) argument 130 openRecord( pid_t pid, int input, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, int *sessionId, status_t *status) argument 827 uint32_t sampleRate = data.readInt32(); local [all...] |
H A D | AudioSystem.cpp | 227 *samplingRate = af->sampleRate(output); 301 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount, argument 305 if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat) 308 gPrevInSamplingRate = sampleRate; 317 gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
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/frameworks/base/services/audioflinger/ |
H A D | AudioFlinger.h | 80 uint32_t sampleRate, 90 virtual uint32_t sampleRate(int output) const; 118 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); 197 uint32_t sampleRate, 323 uint32_t sampleRate, 360 int sampleRate() const; 417 uint32_t sampleRate() const; 583 uint32_t sampleRate, 670 uint32_t sampleRate, 725 uint32_t sampleRate, [all...] |
H A D | AudioFlinger.cpp | 380 uint32_t sampleRate, 449 track = thread->createTrack_l(client, streamType, sampleRate, format, 476 uint32_t AudioFlinger::sampleRate(int output) const function in class:android::AudioFlinger 481 LOGW("sampleRate() unknown thread %d", output); 484 return thread->sampleRate(); 841 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 848 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 1016 uint32_t AudioFlinger::ThreadBase::sampleRate() const function in class:android::AudioFlinger::ThreadBase 1482 uint32_t sampleRate, 1495 if (sampleRate ! 377 createTrack( pid_t pid, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int output, int *sessionId, status_t *status) argument 1479 createTrack_l( const sp<AudioFlinger::Client>& client, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId, status_t *status) argument 3197 TrackBase( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, int sessionId) argument 3325 int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { function in class:android::AudioFlinger::ThreadBase::TrackBase 3358 Track( const wp<ThreadBase>& thread, const sp<Client>& client, int streamType, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument 3672 RecordTrack( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, int sessionId) argument 3785 OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount) argument 4122 openRecord( pid_t pid, int input, uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, int *sessionId, status_t *status) argument 4237 RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id, uint32_t device) argument 4468 createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, int format, int channelMask, int frameCount, uint32_t flags, int sessionId, status_t *status) argument [all...] |
H A D | AudioMixer.cpp | 47 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) argument 48 : mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate) 73 t->sampleRate = mSampleRate; 131 track.sampleRate = mSampleRate; 291 if (sampleRate != value) { 292 sampleRate = value; 584 t->resampler->setSampleRate(t->sampleRate);
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/frameworks/base/media/libstagefright/codecs/aacenc/src/ |
H A D | tns.c | 134 Word32 sampleRate, /*!< Sampling frequency */ 162 tC->tnsStartBand = FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 166 sampleRate, 171 sampleRate, 201 Word32 sampleRate, /*!< Sampling frequency */ 228 tC->tnsStartBand=FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 232 sampleRate, 237 sampleRate, 133 InitTnsConfigurationLong(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_LONG *pC, Word16 active) argument 200 InitTnsConfigurationShort(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_SHORT *pC, Word16 active) argument
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H A D | psy_main.c | 187 Word32 sampleRate, 197 sampleRate, 203 err = InitTnsConfigurationLong(bitRate, sampleRate, channels, 209 sampleRate, 213 err = InitTnsConfigurationShort(bitRate, sampleRate, channels, 252 Word32 sampleRate) 270 sampleRate, 186 psyMainInit(PSY_KERNEL *hPsy, Word32 sampleRate, Word32 bitRate, Word16 channels, Word16 tnsMask, Word16 bandwidth) argument 242 psyMain(Word16 nChannels, ELEMENT_INFO *elemInfo, Word16 *timeSignal, PSY_DATA psyData[MAX_CHANNELS], TNS_DATA tnsData[MAX_CHANNELS], PSY_CONFIGURATION_LONG *hPsyConfLong, PSY_CONFIGURATION_SHORT *hPsyConfShort, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT *psyOutElement, Word32 *pScratchTns, Word32 sampleRate) argument
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H A D | block_switch.c | 111 Word32 sampleRate, 138 if(sampleRate >= 16000) { 109 BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, Word16 *timeSignal, Word32 sampleRate, Word16 chIncrement) argument
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/frameworks/base/media/libmediaplayerservice/ |
H A D | MediaPlayerService.cpp | 1198 *pSampleRate = cache->sampleRate(); 1201 LOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat); 1245 *pSampleRate = cache->sampleRate(); 1248 LOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat); 1342 uint32_t sampleRate, int channelCount, int format, int bufferCount, 1354 LOGV("open(%u, %d, %d, %d, %d)", sampleRate, channelCount, format, bufferCount,mSessionId); 1367 frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate; 1373 sampleRate, 1385 sampleRate, 1405 mMsecsPerFrame = 1.e3 / (float) sampleRate; 1341 open( uint32_t sampleRate, int channelCount, int format, int bufferCount, AudioCallback cb, void *cookie) argument 1612 open( uint32_t sampleRate, int channelCount, int format, int bufferCount, AudioCallback cb, void *cookie) argument [all...] |
H A D | MediaPlayerService.h | 85 uint32_t sampleRate, int channelCount, 141 uint32_t sampleRate, int channelCount, int format, 153 uint32_t sampleRate() const { return mSampleRate; } function in class:android::MediaPlayerService::AudioCache
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/frameworks/base/include/media/ |
H A D | MediaProfiles.h | 242 AudioCodec(audio_encoder codec, int bitRate, int sampleRate, int channels) argument 245 mSampleRate(sampleRate),
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