| /frameworks/av/media/libeffects/testlibs/ |
| D | AudioShelvingFilter.cpp | 50 int sampleRate) in AudioShelvingFilter() argument
52 mBiquad(nChannels, sampleRate) { in AudioShelvingFilter()
53 configure(nChannels, sampleRate); in AudioShelvingFilter()
56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) { in configure() argument
57 mNiquistFreq = sampleRate * 500; in configure()
59 mBiquad.configure(nChannels, sampleRate); in configure()
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| D | AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate) in AudioPeakingFilter() argument
45 : mBiquad(nChannels, sampleRate) { in AudioPeakingFilter()
46 configure(nChannels, sampleRate); in AudioPeakingFilter()
50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) { in configure() argument
51 mNiquistFreq = sampleRate * 500; in configure()
53 mBiquad.configure(nChannels, sampleRate); in configure()
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| D | AudioEqualizer.cpp | 39 int nChannels, int sampleRate, in CreateInstance() argument
44 pMem, nBands, nChannels, sampleRate, nPresets); in CreateInstance()
54 return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate, in CreateInstance()
58 void AudioEqualizer::configure(int nChannels, int sampleRate) { in configure() argument
60 sampleRate); in configure()
61 mpLowShelf->configure(nChannels, sampleRate); in configure()
63 mpPeakingFilters[i].configure(nChannels, sampleRate); in configure()
65 mpHighShelf->configure(nChannels, sampleRate); in configure()
288 int sampleRate, bool ownMem, in AudioEqualizer() argument
290 : mSampleRate(sampleRate) in AudioEqualizer()
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| D | AudioEqualizer.h | 81 int sampleRate,
89 void configure(int nChannels, int sampleRate);
240 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate,
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| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| D | aacenc.c | 142 config.sampleRate = 44100; in voAACEncInit()
274 pOutInfo->Format.SampleRate = hAacEnc->config.sampleRate; in voAACEncGetOutputData()
334 config.sampleRate = pAAC_param->sampleRate; in voAACEncSetParam()
345 if(config.sampleRate == sampRateTab[i]) in voAACEncSetParam()
357 if(config.sampleRate%8000 == 0) in voAACEncSetParam()
362 (config.bitRate > config.sampleRate*6*config.nChannelsOut))) in voAACEncSetParam()
364 config.bitRate = 640*config.sampleRate/tmp*config.nChannelsOut; in voAACEncSetParam()
368 else if(config.bitRate > config.sampleRate*6*config.nChannelsOut) in voAACEncSetParam()
369 config.bitRate = config.sampleRate*6*config.nChannelsOut; in voAACEncSetParam()
376 bitrate = bitrate * tmp / config.sampleRate; in voAACEncSetParam()
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| D | psy_configuration.c | 39 Word32 sampleRate; member
69 Word32 GetSRIndex(Word32 sampleRate) in GetSRIndex() argument
71 if (92017 <= sampleRate) return 0; in GetSRIndex()
72 if (75132 <= sampleRate) return 1; in GetSRIndex()
73 if (55426 <= sampleRate) return 2; in GetSRIndex()
74 if (46009 <= sampleRate) return 3; in GetSRIndex()
75 if (37566 <= sampleRate) return 4; in GetSRIndex()
76 if (27713 <= sampleRate) return 5; in GetSRIndex()
77 if (23004 <= sampleRate) return 6; in GetSRIndex()
78 if (18783 <= sampleRate) return 7; in GetSRIndex()
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| D | aacenc_core.c | 90 config.sampleRate, in AacEncOpen()
111 qcInit.averageBits = (Word16) ((config.bitRate * FRAME_LEN_LONG) / config.sampleRate); in AacEncOpen()
113 qcInit.padding.paddingRest = config.sampleRate; in AacEncOpen()
116 (config.sampleRate>>1)); in AacEncOpen()
130 hAacEnc->bseInit.sampleRate = config.sampleRate; in AacEncOpen()
172 aacEnc->config.sampleRate); in AacEncEncode()
177 aacEnc->config.sampleRate); in AacEncEncode()
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| D | qc_main.c | 61 Word32 sampleRate, in calcFrameLen() argument
69 quot = result / sampleRate; in calcFrameLen()
73 result -= quot * sampleRate; in calcFrameLen()
90 Word32 sampleRate, in framePadding() argument
99 sampleRate, in framePadding()
106 *paddingRest = *paddingRest + sampleRate; in framePadding()
546 Word32 sampleRate) /* output sampling rate */ in AdjustBitrate() argument
555 sampleRate, in AdjustBitrate()
560 sampleRate, in AdjustBitrate()
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| /frameworks/opt/net/voip/src/jni/rtp/ |
| D | AudioGroup.cpp | 100 AudioCodec *codec, int sampleRate, int sampleCount,
104 bool mix(int32_t *output, int head, int tail, int sampleRate);
166 AudioCodec *codec, int sampleRate, int sampleCount, in set() argument
178 mSampleRate = sampleRate / 1000; in set()
234 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate) in mix() argument
253 if (sampleRate == mSampleRate) { in mix()
476 bool set(int sampleRate, int sampleCount);
572 bool AudioGroup::set(int sampleRate, int sampleCount) in set() argument
580 mSampleRate = sampleRate; in set()
594 sampleRate, sampleCount, -1, -1)) { in set()
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| D | G711Codec.cpp | 37 int set(int sampleRate, const char *fmtp) { in set() argument
38 mSampleCount = sampleRate / 50; in set()
88 int set(int sampleRate, const char *fmtp) { in set() argument
89 mSampleCount = sampleRate / 50; in set()
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| D | AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp);
67 int AmrCodec::set(int sampleRate, const char *fmtp) in set() argument
97 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1; in set()
211 int set(int sampleRate, const char *fmtp) { in set() argument
212 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1; in set()
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| /frameworks/av/media/libstagefright/rtsp/ |
| D | APacketSource.cpp | 473 int32_t sampleRate, numChannels; in APacketSource() local
475 desc.c_str(), &sampleRate, &numChannels); in APacketSource()
477 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource()
489 int32_t sampleRate, numChannels; in APacketSource() local
491 desc.c_str(), &sampleRate, &numChannels); in APacketSource()
493 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource()
496 if (sampleRate != 8000 || numChannels != 1) { in APacketSource()
502 int32_t sampleRate, numChannels; in APacketSource() local
504 desc.c_str(), &sampleRate, &numChannels); in APacketSource()
506 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource()
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| D | ARawAudioAssembler.cpp | 134 int32_t sampleRate, numChannels; in MakeFormat() local
136 desc, &sampleRate, &numChannels); in MakeFormat()
138 format->setInt32(kKeySampleRate, sampleRate); in MakeFormat()
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| /frameworks/av/services/audioflinger/ |
| D | AudioResampler.cpp | 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : in AudioResamplerOrder1() argument
43 AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { in AudioResamplerOrder1()
136 int32_t sampleRate, src_quality quality) { in create() argument
189 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); in create()
193 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); in create()
197 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); in create()
201 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality); in create()
211 int32_t sampleRate, src_quality quality) : in AudioResampler() argument
213 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), in AudioResampler()
222 if (sampleRate <= 0) { in AudioResampler()
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| D | FastMixer.cpp | 72 unsigned sampleRate = 0; in threadLoop() local
219 sampleRate = 0; in threadLoop()
222 sampleRate = Format_sampleRate(format); in threadLoop()
225 dumpState->mSampleRate = sampleRate; in threadLoop()
234 if (frameCount > 0 && sampleRate > 0) { in threadLoop()
238 mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks); in threadLoop()
240 periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00 in threadLoop()
241 underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75 in threadLoop()
242 overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50 in threadLoop()
243 forceNs = (frameCount * 950000000LL) / sampleRate; // 0.95 in threadLoop()
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| /frameworks/av/media/libmedia/ |
| D | AudioRecord.cpp | 40 uint32_t sampleRate, in getMinFrameCount() argument
50 status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); in getMinFrameCount()
58 sampleRate, format, channelMask); in getMinFrameCount()
85 uint32_t sampleRate, in AudioRecord() argument
98 mStatus = set(inputSource, sampleRate, format, channelMask, in AudioRecord()
123 uint32_t sampleRate, in set() argument
140 ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, in set()
153 if (sampleRate == 0) { in set()
154 sampleRate = DEFAULT_SAMPLE_RATE; in set()
156 mSampleRate = sampleRate; in set()
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| D | SoundPool.cpp | 497 uint32_t sampleRate; in doLoad() local
503 p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format); in doLoad()
505 p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format); in doLoad()
515 p->pointer(), p->size(), sampleRate, numChannels); in doLoad()
517 if (sampleRate > kMaxSampleRate) { in doLoad()
518 ALOGE("Sample rate (%u) out of range", sampleRate); in doLoad()
533 mSampleRate = sampleRate; in doLoad()
581 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5); in play() local
582 uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate; in play()
610 newTrack = new AudioTrack(streamType, sampleRate, sample->format(), in play()
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| D | AudioTrack.cpp | 55 uint32_t sampleRate) in getMinFrameCount() argument
87 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : in getMinFrameCount()
88 afFrameCount * minBufCount * sampleRate / afSampleRate; in getMinFrameCount()
107 uint32_t sampleRate, in AudioTrack() argument
122 mStatus = set(streamType, sampleRate, format, channelMask, in AudioTrack()
129 uint32_t sampleRate, in AudioTrack() argument
149 mStatus = set(streamType, sampleRate, format, channelMask, in AudioTrack()
177 uint32_t sampleRate, in set() argument
212 if (sampleRate == 0) { in set()
217 sampleRate = afSampleRate; in set()
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| /frameworks/base/core/java/android/speech/srec/ |
| D | MicrophoneInputStream.java | 46 public MicrophoneInputStream(int sampleRate, int fifoDepth) throws IOException { in MicrophoneInputStream() argument
47 mAudioRecord = AudioRecordNew(sampleRate, fifoDepth); in MicrophoneInputStream()
105 private static native int AudioRecordNew(int sampleRate, int fifoDepth); in AudioRecordNew() argument
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| D | WaveHeader.java | 73 …public WaveHeader(short format, short numChannels, int sampleRate, short bitsPerSample, int numByt… in WaveHeader() argument
75 mSampleRate = sampleRate; in WaveHeader()
132 public WaveHeader setSampleRate(int sampleRate) { in setSampleRate() argument
133 mSampleRate = sampleRate; in setSampleRate()
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| /frameworks/av/include/media/ |
| D | AudioTrack.h | 119 uint32_t sampleRate = 0);
158 uint32_t sampleRate = 0,
181 uint32_t sampleRate = 0,
206 uint32_t sampleRate = 0,
299 status_t setSampleRate(uint32_t sampleRate);
521 uint32_t sampleRate,
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| D | AudioRecord.h | 100 uint32_t sampleRate,
135 uint32_t sampleRate = 0,
160 uint32_t sampleRate = 0,
353 status_t openRecord_l(uint32_t sampleRate,
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| /frameworks/av/media/libstagefright/ |
| D | VBRISeeker.cpp | 49 int sampleRate; in CreateFromSource() local
50 if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) { in CreateFromSource()
70 numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate; in CreateFromSource()
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| /frameworks/av/media/libstagefright/codecs/aacdec/ |
| D | SoftAAC2.cpp | 188 aacParams->nSampleRate = mStreamInfo->sampleRate; in internalGetParameter()
221 pcmParams->nSamplingRate = mStreamInfo->sampleRate; in internalGetParameter()
340 if (mStreamInfo->sampleRate && mStreamInfo->numChannels) { in onQueueFilled()
343 mStreamInfo->sampleRate, in onQueueFilled()
470 int prevSampleRate = mStreamInfo->sampleRate; in onQueueFilled()
535 if (mStreamInfo->sampleRate != prevSampleRate || in onQueueFilled()
539 prevSampleRate, mStreamInfo->sampleRate, in onQueueFilled()
546 } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) { in onQueueFilled()
562 + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate; in onQueueFilled()
606 mStreamInfo->sampleRate = 0; in onReset()
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| /frameworks/av/media/libstagefright/codecs/aacenc/SampleCode/ |
| D | AAC_E_SAMPLES.c | 57 param->sampleRate = 44100; in parsecmdline()
84 param->sampleRate = atoi(*argv); in parsecmdline()
116 if(param->sampleRate%8000 == 0) in parsecmdline()
118 param->bitRate = 640*param->nChannels*param->sampleRate/scale; in parsecmdline()
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