Searched refs:sampleRate (Results 1 - 25 of 93) sorted by relevance

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/frameworks/base/media/libstagefright/codecs/aacdec/
H A Dsbr_open.h106 void sbr_open(Int32 sampleRate,
H A Dsbr_open.cpp150 void sbr_open(Int32 sampleRate, argument
176 if (sampleRate > 24000 || bDownSampledSbr)
183 init_sbr_dec(sampleRate,
/frameworks/base/media/libeffects/testlibs/
H A DAudioPeakingFilter.cpp46 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);
H A DAudioShelvingFilter.cpp52 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
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 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
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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/base/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/base/media/libstagefright/codecs/common/include/
H A DvoAAC.h45 int sampleRate; /*! audio file sample rate */ member in struct:__anon695
/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/base/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:__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
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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/base/cmds/stagefright/
H A DSineSource.h12 SineSource(int32_t sampleRate, int32_t numChannels);
/frameworks/media/libvideoeditor/vss/common/inc/
H A DVideoEditorResampler.h26 M4OSA_Int32 sampleRate, M4OSA_Int32 quality);
/frameworks/base/services/audioflinger/
H A DAudioResampler.h45 int32_t sampleRate, int quality=DEFAULT);
70 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
H A DAudioResampler.cpp42 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
/frameworks/base/media/libmedia/
H A DAudioRecord.cpp52 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
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