Searched refs:sampleRate (Results 1 - 25 of 109) sorted by relevance

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/frameworks/av/media/libeffects/testlibs/
H A DAudioPeakingFilter.cpp44 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);
H A DAudioShelvingFilter.cpp50 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
H A DAudioShelvingFilter.h50 // 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);
H A DAudioEqualizer.h70 // 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,
H A DAudioEqualizer.cpp39 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
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H A DAudioBiquadFilter.h44 // sampleRate Sample rate, in Hz.
45 AudioBiquadFilter(int nChannels, int sampleRate);
49 // sampleRate Sample rate, in Hz.
50 void configure(int nChannels, int sampleRate);
H A DAudioPeakingFilter.h43 // 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);
/frameworks/av/media/libstagefright/codecs/aacenc/inc/
H A Dbitenc.h35 Word32 sampleRate; member in struct:BITSTREAMENCODER_INIT
H A Dpsy_main.h50 Word32 sampleRate,
67 Word32 sampleRate);
/frameworks/av/media/libstagefright/codecs/common/include/
H A DvoAAC.h45 int sampleRate; /*! audio file sample rate */ member in struct:__anon634
/frameworks/base/voip/jni/rtp/
H A DAudioCodec.h29 virtual int set(int sampleRate, const char *fmtp) = 0;
H A DGsmCodec.cpp42 int set(int sampleRate, const char *fmtp) { argument
43 return (sampleRate == 8000 && mEncode && mDecode) ? 160 : -1;
H A DG711Codec.cpp37 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;
H A DAmrCodec.cpp53 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;
/frameworks/av/media/libstagefright/codecs/aacenc/src/
H A Daacenc_core.c90 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);
H A Daacenc.c142 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*
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H A Dpsy_configuration.c39 Word32 sampleRate; member in struct:__anon561
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
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/frameworks/base/core/java/android/speech/srec/
H A DMicrophoneInputStream.java42 * @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
/frameworks/av/cmds/stagefright/
H A DSineSource.h12 SineSource(int32_t sampleRate, int32_t numChannels);
/frameworks/av/libvideoeditor/vss/common/inc/
H A DVideoEditorResampler.h26 M4OSA_Int32 sampleRate, M4OSA_Int32 quality);
/frameworks/av/media/libnbaio/
H A DAudioStreamInSource.cpp47 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local
50 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
H A DAudioStreamOutSink.cpp44 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local
47 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
H A DNBAIO.cpp85 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount) argument
87 if (sampleRate == 44100 && channelCount == 2) return Format_SR44_1_C2_I16;
88 if (sampleRate == 48000 && channelCount == 2) return Format_SR48_C2_I16;
89 if (sampleRate == 44100 && channelCount == 1) return Format_SR44_1_C1_I16;
90 if (sampleRate == 48000 && channelCount == 1) return Format_SR48_C1_I16;
/frameworks/av/media/libmedia/
H A DAudioRecord.cpp40 uint32_t sampleRate,
50 if (AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size)
57 ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
58 sampleRate, format, channelMask);
84 uint32_t sampleRate,
95 mStatus = set(inputSource, sampleRate, format, channelMask,
119 uint32_t sampleRate,
130 ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount);
142 if (sampleRate
38 getMinFrameCount( int* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument
82 AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, callback_t cbf, void* user, int notificationFrames, int sessionId) argument
117 set( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) argument
425 openRecord_l( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_io_handle_t input) argument
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/frameworks/av/include/media/
H A DAudioTrack.h119 uint32_t sampleRate = 0);
135 * sampleRate: Track sampling rate in Hz.
155 uint32_t sampleRate = 0,
167 uint32_t sampleRate = 0,
187 uint32_t sampleRate = 0,
207 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
211 uint32_t sampleRate = 0,
295 status_t setSampleRate(int sampleRate);
486 uint32_t sampleRate,

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