/frameworks/base/media/libstagefright/codecs/aacdec/ |
H A D | sbr_open.h | 106 void sbr_open(Int32 sampleRate,
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H A D | sbr_open.cpp | 150 void sbr_open(Int32 sampleRate, argument 176 if (sampleRate > 24000 || bDownSampledSbr) 183 init_sbr_dec(sampleRate,
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/frameworks/base/media/libeffects/testlibs/ |
H A D | AudioPeakingFilter.cpp | 46 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate) argument 47 : mBiquad(nChannels, sampleRate) { 48 configure(nChannels, sampleRate); 52 void AudioPeakingFilter::configure(int nChannels, int sampleRate) { argument 53 mNiquistFreq = sampleRate * 500; 55 mBiquad.configure(nChannels, sampleRate);
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H A D | AudioShelvingFilter.cpp | 52 int sampleRate) 54 mBiquad(nChannels, sampleRate) { 55 configure(nChannels, sampleRate); 58 void AudioShelvingFilter::configure(int nChannels, int sampleRate) { argument 59 mNiquistFreq = sampleRate * 500; 61 mBiquad.configure(nChannels, sampleRate); 51 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 LOGV("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/base/media/libstagefright/codecs/aacenc/inc/ |
H A D | bitenc.h | 35 Word32 sampleRate; member in struct:BITSTREAMENCODER_INIT
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H A D | psy_main.h | 50 Word32 sampleRate, 67 Word32 sampleRate);
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/frameworks/base/media/libstagefright/codecs/common/include/ |
H A D | voAAC.h | 45 int sampleRate; /*! audio file sample rate */ member in struct:__anon695
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/frameworks/base/voip/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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H A D | AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp); 67 int AmrCodec::set(int sampleRate, const char *fmtp) argument 97 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1; 211 int set(int sampleRate, const char *fmtp) { argument 212 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
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/frameworks/base/media/libstagefright/codecs/aacenc/src/ |
H A D | aacenc_core.c | 90 config.sampleRate, 111 qcInit.averageBits = (Word16) ((config.bitRate * FRAME_LEN_LONG) / config.sampleRate); 113 qcInit.padding.paddingRest = config.sampleRate; 116 (config.sampleRate>>1)); 130 hAacEnc->bseInit.sampleRate = config.sampleRate; 172 aacEnc->config.sampleRate); 177 aacEnc->config.sampleRate);
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H A D | aacenc.c | 142 config.sampleRate = 44100; 274 pOutInfo->Format.SampleRate = hAacEnc->config.sampleRate; 334 config.sampleRate = pAAC_param->sampleRate; 345 if(config.sampleRate == sampRateTab[i]) 357 if(config.sampleRate%8000 == 0) 362 (config.bitRate > config.sampleRate*6*config.nChannelsOut)) 364 config.bitRate = 640*config.sampleRate/tmp*config.nChannelsOut; 368 else if(config.bitRate > config.sampleRate*6*config.nChannelsOut) 369 config.bitRate = config.sampleRate* [all...] |
H A D | psy_configuration.c | 39 Word32 sampleRate; member in struct:__anon626 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/base/core/java/android/speech/srec/ |
H A D | MicrophoneInputStream.java | 42 * @param sampleRate sample rate of the microphone, typically 11025 or 8000. 43 * @param fifoDepth depth of the real time fifo, measured in sampleRate clock ticks. 46 public MicrophoneInputStream(int sampleRate, int fifoDepth) throws IOException { argument 47 mAudioRecord = AudioRecordNew(sampleRate, fifoDepth); 105 private static native int AudioRecordNew(int sampleRate, int fifoDepth); argument
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/frameworks/base/cmds/stagefright/ |
H A D | SineSource.h | 12 SineSource(int32_t sampleRate, int32_t numChannels);
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/frameworks/media/libvideoeditor/vss/common/inc/ |
H A D | VideoEditorResampler.h | 26 M4OSA_Int32 sampleRate, M4OSA_Int32 quality);
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/frameworks/base/services/audioflinger/ |
H A D | AudioResampler.h | 45 int32_t sampleRate, int quality=DEFAULT); 70 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
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H A D | AudioResampler.cpp | 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 43 AudioResampler(bitDepth, inChannelCount, sampleRate), mX0L(0), mX0R(0) { 82 int32_t sampleRate, int quality) { 100 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); 104 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); 108 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); 118 int32_t sampleRate) : 120 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), 81 create(int bitDepth, int inChannelCount, int32_t sampleRate, int quality) argument 117 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate) argument
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/frameworks/base/media/libmedia/ |
H A D | AudioRecord.cpp | 52 uint32_t sampleRate, 57 if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size) 64 LOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d", 65 sampleRate, format, channelCount); 89 uint32_t sampleRate, 100 mStatus = set(inputSource, sampleRate, format, channelMask, 123 uint32_t sampleRate, 135 LOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount); 147 if (sampleRate 50 getMinFrameCount( int* frameCount, uint32_t sampleRate, int format, int channelCount) argument 87 AudioRecord( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 121 set( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) argument 449 openRecord_l( uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, audio_io_handle_t input) argument [all...] |