/frameworks/av/media/libstagefright/ |
H A D | AudioPlayer.cpp | 620 uint32_t sampleRate; local 623 sampleRate = mAudioSink->getSampleRate(); 626 sampleRate = mAudioTrack->getSampleRate(); 628 if (sampleRate != 0) { 629 mSampleRate = sampleRate;
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H A D | AudioSource.cpp | 54 uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate, 57 mSampleRate(sampleRate), 58 mOutSampleRate(outSampleRate > 0 ? outSampleRate : sampleRate), 67 ALOGV("sampleRate: %u, outSampleRate: %u, channelCount: %u", 68 sampleRate, outSampleRate, channelCount); 70 CHECK(sampleRate > 0); 74 sampleRate, 89 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 52 AudioSource( audio_source_t inputSource, const String16 &opPackageName, uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate, uid_t uid, pid_t pid) argument
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/frameworks/av/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 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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H A D | SoftAACEncoder.cpp | 352 params.sampleRate = mSampleRate; 365 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) { argument 374 if (sampleRate == kSampleRateTable[i]) {
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | block_switch.c | 108 Word32 sampleRate, 135 if(sampleRate >= 16000) { 106 BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, Word16 *timeSignal, Word32 sampleRate, Word16 chIncrement) argument
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H A D | psy_main.c | 191 Word32 sampleRate, 201 sampleRate, 207 err = InitTnsConfigurationLong(bitRate, sampleRate, channels, 213 sampleRate, 217 err = InitTnsConfigurationShort(bitRate, sampleRate, channels, 256 Word32 sampleRate) 274 sampleRate, 190 psyMainInit(PSY_KERNEL *hPsy, Word32 sampleRate, Word32 bitRate, Word16 channels, Word16 tnsMask, Word16 bandwidth) argument 246 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 | psy_configuration.c | 39 Word32 sampleRate; member in struct:__anon379 69 Word32 GetSRIndex(Word32 sampleRate) argument 71 if (92017 <= sampleRate) return 0; 72 if (75132 <= sampleRate) return 1; 73 if (55426 <= sampleRate) return 2; 74 if (46009 <= sampleRate) return 3; 75 if (37566 <= sampleRate) return 4; 76 if (27713 <= sampleRate) return 5; 77 if (23004 <= sampleRate) return 6; 78 if (18783 <= sampleRate) retur [all...] |
H A D | qc_main.c | 62 Word32 sampleRate, 70 quot = result / sampleRate; 74 result -= quot * sampleRate; 91 Word32 sampleRate, 100 sampleRate, 107 *paddingRest = *paddingRest + sampleRate; 544 Word32 sampleRate) /* output sampling rate */ 553 sampleRate, 558 sampleRate, 61 calcFrameLen(Word32 bitRate, Word32 sampleRate, FRAME_LEN_RESULT_MODE mode) argument 90 framePadding(Word32 bitRate, Word32 sampleRate, Word32 *paddingRest) argument 542 AdjustBitrate(QC_STATE *hQC, Word32 bitRate, Word32 sampleRate) argument
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | AMPEG4ElementaryAssembler.cpp | 89 static bool GetSampleRateIndex(int32_t sampleRate, size_t *tableIndex) { argument 99 if (sampleRate == kSampleRateTable[index]) { 189 int32_t sampleRate, numChannels; local 191 desc.c_str(), &sampleRate, &numChannels); 194 CHECK(GetSampleRateIndex(sampleRate, &mSampleRateIndex));
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H A D | APacketSource.cpp | 471 int32_t sampleRate, numChannels; local 473 desc.c_str(), &sampleRate, &numChannels); 475 mFormat->setInt32(kKeySampleRate, sampleRate); 487 int32_t sampleRate, numChannels; local 489 desc.c_str(), &sampleRate, &numChannels); 491 mFormat->setInt32(kKeySampleRate, sampleRate); 494 if (sampleRate != 8000 || numChannels != 1) { 500 int32_t sampleRate, numChannels; local 502 desc.c_str(), &sampleRate, &numChannels); 504 mFormat->setInt32(kKeySampleRate, sampleRate); 551 int32_t sampleRate, numChannels; local [all...] |
H A D | ARTPWriter.cpp | 475 int32_t sampleRate, numChannels; local 476 CHECK(mSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); 480 CHECK_EQ(sampleRate, (mMode == AMR_NB) ? 8000 : 16000); 483 sdp.append(AStringPrintf("/%d/%d", sampleRate, numChannels));
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/frameworks/av/services/audioflinger/ |
H A D | AudioResampler.cpp | 42 AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) : argument 43 AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { 148 int32_t sampleRate, src_quality quality) { 219 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate); 224 resampler = new AudioResamplerCubic(inChannelCount, sampleRate); 229 resampler = new AudioResamplerSinc(inChannelCount, sampleRate); 234 resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality); 242 sampleRate, quality); 247 sampleRate, quality); 250 sampleRate, qualit 147 create(audio_format_t format, int inChannelCount, int32_t sampleRate, src_quality quality) argument 261 AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality) argument [all...] |
H A D | BufferProviders.cpp | 139 uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) : 147 sampleRate, sessionId); 162 mDownmixConfig.inputCfg.samplingRate = sampleRate; 163 mDownmixConfig.outputCfg.samplingRate = sampleRate; 325 audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) : 328 mSampleRate(sampleRate), 333 mSonicStream(sonicCreateStream(sampleRate, mChannelCount)), 342 this, channelCount, format, sampleRate, playbackRate.mSpeed, 136 DownmixerBufferProvider( audio_channel_mask_t inputChannelMask, audio_channel_mask_t outputChannelMask, audio_format_t format, uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) argument 324 TimestretchBufferProvider(int32_t channelCount, audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) argument
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H A D | PatchPanel.cpp | 265 config.sample_rate = newPatch->mPlaybackThread->sampleRate(); 430 uint32_t sampleRate = patch->mPlaybackThread->sampleRate(); local 435 sampleRate, 455 sampleRate,
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H A D | TrackBase.h | 59 uint32_t sampleRate, 107 virtual uint32_t sampleRate() const { return mSampleRate; } function in class:TrackBase
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H A D | AudioMixer.h | 45 AudioMixer(size_t frameCount, uint32_t sampleRate, 214 uint32_t sampleRate; member in struct:android::AudioMixer::track_t
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/frameworks/av/services/audioflinger/tests/ |
H A D | test-mixer.cpp | 55 uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) { 62 info.samplerate = sampleRate; 54 writeFile(const char *filename, const void *buffer, uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) argument
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H A D | test_utils.h | 188 size_t channels, double sampleRate, double freq) 190 double tscale = 1. / sampleRate; 212 size_t channels, double sampleRate, double minfreq, double maxfreq) 214 double tscale = 1. / sampleRate; 250 void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time) argument 252 createBufferByFrames<T>(channels, sampleRate, sampleRate*time); 258 double freq, double sampleRate, double time) 260 createBufferByFrames<T>(channels, sampleRate, sampleRate*tim 187 createSine(void *vbuffer, size_t frames, size_t channels, double sampleRate, double freq) argument 211 createChirp(void *vbuffer, size_t frames, size_t channels, double sampleRate, double minfreq, double maxfreq) argument 257 setSine(size_t channels, double freq, double sampleRate, double time) argument 284 createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) argument [all...] |
/frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
H A D | AudioSession.cpp | 32 uint32_t sampleRate, 40 mConfig({ .format = format, .sample_rate = sampleRate, .channel_mask = channelMask}), 104 other->sampleRate() == mConfig.sample_rate && 29 AudioSession(audio_session_t session, audio_source_t inputSource, audio_format_t format, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_input_flags_t flags, uid_t uid, bool isSoundTrigger, AudioMix* policyMix, AudioPolicyClientInterface *clientInterface) argument
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/frameworks/base/core/java/android/speech/tts/ |
H A D | BlockingAudioTrack.java | 78 BlockingAudioTrack(AudioOutputParams audioParams, int sampleRate, argument 81 mSampleRateInHz = sampleRate;
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/frameworks/wilhelm/src/android/ |
H A D | AudioRecorder_to_android.cpp | 434 uint32_t sampleRate = sles_to_android_sampleRate(df_pcm->samplesPerSec); local 464 sampleRate, // sample rate in Hertz
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/frameworks/wilhelm/tests/examples/ |
H A D | slesTestRecBuffQueue.cpp | 42 uint32_t sampleRate = 48000; variable 164 info.samplerate = sampleRate; 250 pcm.sampleRate = sampleRate * 1000; // milliHz 435 sampleRate = atoi(&arg[2]);
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H A D | slesTestFeedback.cpp | 41 static SLuint32 sampleRate = 48000; // -s# variable 296 sampleRate = atoi(&arg[2]); 297 switch (sampleRate) { 310 (unsigned) sampleRate); 373 info.samplerate = sampleRate; 419 pcm.samplesPerSec = sampleRate * 1000;
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/frameworks/av/cmds/stagefright/ |
H A D | sf2.cpp | 307 int32_t numChannels, sampleRate; local 309 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 312 msg->setInt32("sample-rate", sampleRate);
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/frameworks/av/media/libstagefright/wifi-display/source/ |
H A D | TSPacketizer.cpp | 318 int32_t sampleRate; local 319 CHECK(mFormat->findInt32("sample-rate", &sampleRate)); 320 CHECK(sampleRate == 44100 || sampleRate == 48000); 327 unsigned sampling_frequency = (sampleRate == 44100) ? 1 : 2;
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