/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/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/av/media/libstagefright/codecs/common/include/ |
H A D | voAAC.h | 45 int sampleRate; /*! audio file sample rate */ member in struct:__anon649
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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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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/av/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:__anon576 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/av/cmds/stagefright/ |
H A D | SineSource.h | 12 SineSource(int32_t sampleRate, int32_t numChannels);
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/frameworks/av/libvideoeditor/vss/common/inc/ |
H A D | VideoEditorResampler.h | 26 M4OSA_Int32 sampleRate, M4OSA_Int32 quality);
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/frameworks/av/media/libnbaio/ |
H A D | AudioStreamInSource.cpp | 47 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 50 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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H A D | AudioStreamOutSink.cpp | 44 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 47 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | ARawAudioAssembler.cpp | 134 int32_t sampleRate, numChannels; local 136 desc, &sampleRate, &numChannels); 138 format->setInt32(kKeySampleRate, sampleRate);
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/frameworks/av/services/audioflinger/ |
H A D | AudioResampler.cpp | 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 43 AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { 136 int32_t sampleRate, src_quality quality) { 189 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); 193 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); 197 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); 201 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality); 211 int32_t sampleRate, src_quality quality) : 213 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputInde 135 create(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) argument 210 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) argument [all...] |
/frameworks/av/include/media/ |
H A D | AudioRecord.h | 100 uint32_t sampleRate, 117 * sampleRate: Track sampling rate in Hz. 135 uint32_t sampleRate = 0, 155 * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) 160 uint32_t sampleRate = 0, 353 status_t openRecord_l(uint32_t sampleRate,
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