/frameworks/base/tests/Camera2Tests/SmartCamera/SimpleCamera/src/androidx/media/filterfw/decoder/ |
H A D | AudioSample.java | 21 public final int sampleRate; field in class:AudioSample 25 public AudioSample(int sampleRate, int channelCount, byte[] bytes) { argument 26 this.sampleRate = sampleRate;
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/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate) argument 45 : mBiquad(nChannels, sampleRate) { 46 configure(nChannels, sampleRate); 50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) { argument 51 mNiquistFreq = sampleRate * 500; 53 mBiquad.configure(nChannels, sampleRate);
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H A D | AudioShelvingFilter.cpp | 50 int sampleRate) 52 mBiquad(nChannels, sampleRate) { 53 configure(nChannels, sampleRate); 56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) { argument 57 mNiquistFreq = sampleRate * 500; 59 mBiquad.configure(nChannels, sampleRate); 49 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate) argument
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H A D | AudioShelvingFilter.h | 50 // sampleRate The input/output sample rate, in Hz. 51 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate); 56 // sampleRate The input/output sample rate, in Hz. 57 void configure(int nChannels, int sampleRate);
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H A D | AudioEqualizer.h | 70 // sampleRate The input/output sample rate, in Hz. 81 int sampleRate, 88 // sampleRate The input/output sample rate, in Hz. 89 void configure(int nChannels, int sampleRate); 232 // sampleRate The input/output sample rate, in Hz. 240 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate,
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H A D | AudioEqualizer.cpp | 39 int nChannels, int sampleRate, 43 "sampleRate=%d, nPresets=%d)", 44 pMem, nBands, nChannels, sampleRate, nPresets); 54 return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate, 58 void AudioEqualizer::configure(int nChannels, int sampleRate) { argument 59 ALOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels, 60 sampleRate); 61 mpLowShelf->configure(nChannels, sampleRate); 63 mpPeakingFilters[i].configure(nChannels, sampleRate); 65 mpHighShelf->configure(nChannels, sampleRate); 38 CreateInstance(void * pMem, int nBands, int nChannels, int sampleRate, const PresetConfig * presets, int nPresets) argument 287 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate, bool ownMem, const PresetConfig * presets, int nPresets) argument [all...] |
H A D | AudioBiquadFilter.h | 44 // sampleRate Sample rate, in Hz. 45 AudioBiquadFilter(int nChannels, int sampleRate); 49 // sampleRate Sample rate, in Hz. 50 void configure(int nChannels, int sampleRate);
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H A D | AudioPeakingFilter.h | 43 // sampleRate The input/output sample rate, in Hz. 44 AudioPeakingFilter(int nChannels, int sampleRate); 49 // sampleRate The input/output sample rate, in Hz. 50 void configure(int nChannels, int sampleRate);
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/frameworks/av/services/oboeservice/ |
H A D | TimestampScheduler.h | 52 int32_t sampleRate) { 53 mBurstPeriod = AAUDIO_NANOS_PER_SECOND * framesPerBurst / sampleRate; 51 setBurstPeriod(int32_t framesPerBurst, int32_t sampleRate) argument
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
H A D | bitenc.h | 35 Word32 sampleRate; member in struct:BITSTREAMENCODER_INIT
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/frameworks/av/media/libstagefright/codecs/common/include/ |
H A D | voAAC.h | 45 int sampleRate; /*! audio file sample rate */ member in struct:__anon561
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/frameworks/base/core/java/android/bluetooth/ |
H A D | BluetoothAudioConfig.java | 35 public BluetoothAudioConfig(int sampleRate, int channelConfig, int audioFormat) { argument 36 mSampleRate = sampleRate; 70 int sampleRate = in.readInt(); 73 return new BluetoothAudioConfig(sampleRate, channelConfig, audioFormat);
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/frameworks/opt/net/voip/src/jni/rtp/ |
H A D | AudioCodec.h | 29 virtual int set(int sampleRate, const char *fmtp) = 0;
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H A D | GsmCodec.cpp | 42 int set(int sampleRate, const char */* fmtp */) { argument 43 return (sampleRate == 8000 && mEncode && mDecode) ? 160 : -1;
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H A D | G711Codec.cpp | 37 int set(int sampleRate, const char */* fmtp */) { argument 38 mSampleCount = sampleRate / 50; 88 int set(int sampleRate, const char */* fmtp */) { argument 89 mSampleCount = sampleRate / 50;
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | aacenc_core.c | 89 config.sampleRate, 110 qcInit.averageBits = (Word16) ((config.bitRate * FRAME_LEN_LONG) / config.sampleRate); 112 qcInit.padding.paddingRest = config.sampleRate; 115 (config.sampleRate>>1)); 129 hAacEnc->bseInit.sampleRate = config.sampleRate; 171 aacEnc->config.sampleRate); 176 aacEnc->config.sampleRate);
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H A D | aacenc.c | 148 config.sampleRate = 44100; 280 pOutInfo->Format.SampleRate = hAacEnc->config.sampleRate; 340 config.sampleRate = pAAC_param->sampleRate; 351 if(config.sampleRate == sampRateTab[i]) 363 if(config.sampleRate%8000 == 0) 368 (config.bitRate > config.sampleRate*6*config.nChannelsOut))) 370 config.bitRate = 640*config.sampleRate/tmp*config.nChannelsOut; 374 else if(config.bitRate > config.sampleRate*6*config.nChannelsOut) 375 config.bitRate = config.sampleRate* [all...] |
H A D | psy_configuration.c | 39 Word32 sampleRate; member in struct:__anon501 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...] |
/frameworks/av/media/libaudiohal/ |
H A D | StreamPowerLog.h | 42 void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) { argument 50 (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000; 52 sampleRate,
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/frameworks/av/media/libaudioprocessing/tests/ |
H A D | test_utils.h | 194 size_t channels, double sampleRate, double freq) 196 double tscale = 1. / sampleRate; 218 size_t channels, double sampleRate, double minfreq, double maxfreq) 220 double tscale = 1. / sampleRate; 256 void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time) argument 258 createBufferByFrames<T>(channels, sampleRate, sampleRate*time); 264 double freq, double sampleRate, double time) 266 createBufferByFrames<T>(channels, sampleRate, sampleRate*tim 193 createSine(void *vbuffer, size_t frames, size_t channels, double sampleRate, double freq) argument 217 createChirp(void *vbuffer, size_t frames, size_t channels, double sampleRate, double minfreq, double maxfreq) argument 263 setSine(size_t channels, double freq, double sampleRate, double time) argument 290 createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) argument [all...] |
/frameworks/av/media/libaaudio/tests/ |
H A D | test_open_params.cpp | 59 int32_t sampleRate, 77 direction, channelCount, sampleRate, format); 84 AAudioStreamBuilder_setSampleRate(aaudioBuilder, sampleRate); 113 if (sampleRate != AAUDIO_UNSPECIFIED) { 114 EXPECT_EQ(sampleRate, actualSampleRate); 57 testOpenOptions(aaudio_direction_t direction, int32_t channelCount, int32_t sampleRate, aaudio_format_t format) argument
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/frameworks/av/media/libnbaio/ |
H A D | AudioStreamInSource.cpp | 50 uint32_t sampleRate; local 52 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat); 54 mFormat = Format_from_SR_C(sampleRate,
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H A D | AudioStreamOutSink.cpp | 48 uint32_t sampleRate; local 50 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat); 52 mFormat = Format_from_SR_C(sampleRate,
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/frameworks/av/media/libaudioprocessing/ |
H A D | AudioResampler.cpp | 45 AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) : argument 46 AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { 151 int32_t sampleRate, src_quality quality) { 222 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate); 227 resampler = new AudioResamplerCubic(inChannelCount, sampleRate); 232 resampler = new AudioResamplerSinc(inChannelCount, sampleRate); 237 resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality); 245 sampleRate, quality); 250 sampleRate, quality); 253 sampleRate, qualit 150 create(audio_format_t format, int inChannelCount, int32_t sampleRate, src_quality quality) argument 264 AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality) argument [all...] |
/frameworks/av/media/libaaudio/src/core/ |
H A D | AAudioStreamParameters.h | 44 void setSampleRate(int32_t sampleRate) { argument 45 mSampleRate = sampleRate;
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