/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 | aacenc_core.h | 40 Word32 sampleRate; /* audio file sample rate */ member in struct:__anon314
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/frameworks/base/tests/Camera2Tests/SmartCamera/SimpleCamera/src/androidx/media/filterfw/decoder/ |
H A D | AudioSample.java | 21 public final int sampleRate; field in class:AudioSample 25 public AudioSample(int sampleRate, int channelCount, byte[] bytes) { argument 26 this.sampleRate = 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:__anon436
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/frameworks/av/cmds/stagefright/ |
H A D | SineSource.cpp | 12 SineSource::SineSource(int32_t sampleRate, int32_t numChannels) argument 14 mSampleRate(sampleRate),
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H A D | SimplePlayer.cpp | 582 int32_t sampleRate; local 584 CHECK(format->findInt32("sample-rate", &sampleRate)); 588 sampleRate,
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/frameworks/opt/net/voip/src/jni/rtp/ |
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/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 | AudioBiquadFilter.cpp | 28 AudioBiquadFilter::AudioBiquadFilter(int nChannels, int sampleRate) { argument 29 configure(nChannels, sampleRate); 33 void AudioBiquadFilter::configure(int nChannels, int sampleRate) { argument 35 assert(sampleRate > 0); 39 / 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...] |
/frameworks/av/media/libnbaio/ |
H A D | AudioStreamInSource.cpp | 46 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 49 mFormat = Format_from_SR_C(sampleRate,
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H A D | AudioStreamOutSink.cpp | 43 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 46 mFormat = Format_from_SR_C(sampleRate,
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H A D | NBAIO.cpp | 48 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount, argument 51 if (sampleRate == 0 || channelCount == 0 || !audio_is_valid_format(format)) { 55 ret.mSampleRate = sampleRate;
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | ARawAudioAssembler.cpp | 136 int32_t sampleRate, numChannels; local 138 desc, &sampleRate, &numChannels); 140 format->setInt32(kKeySampleRate, sampleRate);
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/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerCubic.h | 31 AudioResamplerCubic(int inChannelCount, int32_t sampleRate) : argument 32 AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
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/frameworks/base/core/java/android/bluetooth/ |
H A D | BluetoothAudioConfig.java | 35 public BluetoothAudioConfig(int sampleRate, int channelConfig, int audioFormat) { argument 36 mSampleRate = sampleRate; 70 int sampleRate = in.readInt(); 73 return new BluetoothAudioConfig(sampleRate, channelConfig, audioFormat);
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/frameworks/av/include/media/ |
H A D | AudioResamplerPublic.h | 169 static inline bool isMusicRate(uint32_t sampleRate) { argument 170 return sampleRate >= AUDIO_PROCESSING_MUSIC_RATE;
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/frameworks/av/media/libstagefright/ |
H A D | AMRWriter.cpp | 80 int32_t sampleRate; local 83 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 84 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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H A D | VBRISeeker.cpp | 52 int sampleRate; local 53 if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) { 73 numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate;
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H A D | AACWriter.cpp | 195 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) { argument 205 if (sampleRate == kSampleRateTable[index]) { 207 sampleRate, index); 213 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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H A D | AudioSource.cpp | 54 uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate) 56 mSampleRate(sampleRate), 57 mOutSampleRate(outSampleRate > 0 ? outSampleRate : sampleRate), 62 ALOGV("sampleRate: %u, outSampleRate: %u, channelCount: %u", 63 sampleRate, outSampleRate, channelCount); 65 CHECK(sampleRate > 0); 69 sampleRate, 84 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 52 AudioSource( audio_source_t inputSource, const String16 &opPackageName, uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate) argument
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/frameworks/base/core/java/android/speech/tts/ |
H A D | SynthesisPlaybackQueueItem.java | 66 SynthesisPlaybackQueueItem(AudioOutputParams audioParams, int sampleRate, argument 77 mAudioTrack = new BlockingAudioTrack(audioParams, sampleRate, audioFormat, channelCount);
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