/frameworks/av/media/libaudioprocessing/tests/ |
D | test_utils.h | 194 size_t channels, double sampleRate, double freq) 196 double tscale = 1. / sampleRate; 218 size_t channels, double sampleRate, double minfreq, double maxfreq) 220 double tscale = 1. / sampleRate; 256 void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time) 258 createBufferByFrames<T>(channels, sampleRate, sampleRate*time); 264 double freq, double sampleRate, double time) 266 createBufferByFrames<T>(channels, sampleRate, sampleRate*time); 290 void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) 297 mSampleRate = sampleRate;
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/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() [all …]
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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/base/tests/Camera2Tests/SmartCamera/SimpleCamera/src/androidx/media/filterfw/decoder/ |
D | AudioSample.java | 21 public final int sampleRate; field in AudioSample 25 public AudioSample(int sampleRate, int channelCount, byte[] bytes) { in AudioSample() argument 26 this.sampleRate = sampleRate; in AudioSample()
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/frameworks/av/media/libaudioprocessing/ |
D | AudioResampler.cpp | 45 AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) : in AudioResamplerOrder1() argument 46 AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { in AudioResamplerOrder1() 151 int32_t sampleRate, src_quality quality) { in create() argument 222 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate); in create() 227 resampler = new AudioResamplerCubic(inChannelCount, sampleRate); in create() 232 resampler = new AudioResamplerSinc(inChannelCount, sampleRate); in create() 237 resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality); in create() 245 sampleRate, quality); in create() 250 sampleRate, quality); in create() 253 sampleRate, quality); in create() [all …]
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/frameworks/av/media/libmedia/include/media/ |
D | JAudioFormat.h | 29 uint32_t sampleRate, in createAudioFormatObj() argument 37 if (sampleRate == 0) { in createAudioFormatObj() 41 sampleRate = env->GetStaticIntField(jAudioFormatCls, jSampleRateUnspecified); in createAudioFormatObj() 51 jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetSampleRate, sampleRate); in createAudioFormatObj()
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/frameworks/base/core/java/android/bluetooth/ |
D | BluetoothAudioConfig.java | 35 public BluetoothAudioConfig(int sampleRate, int channelConfig, int audioFormat) { in BluetoothAudioConfig() argument 36 mSampleRate = sampleRate; in BluetoothAudioConfig() 70 int sampleRate = in.readInt(); 73 return new BluetoothAudioConfig(sampleRate, channelConfig, audioFormat);
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/frameworks/av/media/libaaudio/tests/ |
D | test_open_params.cpp | 44 int32_t sampleRate, in testOpenOptions() argument 61 direction, channelCount, sampleRate, format); in testOpenOptions() 68 AAudioStreamBuilder_setSampleRate(aaudioBuilder, sampleRate); in testOpenOptions() 97 if (sampleRate != AAUDIO_UNSPECIFIED) { in testOpenOptions() 98 EXPECT_EQ(sampleRate, actualSampleRate); in testOpenOptions()
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/frameworks/opt/net/voip/src/jni/rtp/ |
D | AudioGroup.cpp | 102 AudioCodec *codec, int sampleRate, int sampleCount, 106 bool mix(int32_t *output, int head, int tail, int sampleRate); 168 AudioCodec *codec, int sampleRate, int sampleCount, in set() argument 180 mSampleRate = sampleRate / 1000; in set() 236 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate) in mix() argument 255 if (sampleRate == mSampleRate) { in mix() 478 bool set(int sampleRate, int sampleCount); 577 bool AudioGroup::set(int sampleRate, int sampleCount) in set() argument 585 mSampleRate = sampleRate; in set() 599 sampleRate, sampleCount, -1, -1)) { in set() [all …]
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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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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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/frameworks/av/media/libstagefright/rtsp/ |
D | APacketSource.cpp | 470 int32_t sampleRate, numChannels; in APacketSource() local 472 desc.c_str(), &sampleRate, &numChannels); in APacketSource() 474 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource() 486 int32_t sampleRate, numChannels; in APacketSource() local 488 desc.c_str(), &sampleRate, &numChannels); in APacketSource() 490 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource() 493 if (sampleRate != 8000 || numChannels != 1) { in APacketSource() 499 int32_t sampleRate, numChannels; in APacketSource() local 501 desc.c_str(), &sampleRate, &numChannels); in APacketSource() 503 mFormat->setInt32(kKeySampleRate, sampleRate); in APacketSource() [all …]
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D | ARawAudioAssembler.cpp | 136 int32_t sampleRate, numChannels; in MakeFormat() local 138 desc, &sampleRate, &numChannels); in MakeFormat() 140 format->setInt32(kKeySampleRate, sampleRate); in MakeFormat()
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/frameworks/av/cmds/stagefright/ |
D | audioloop.cpp | 102 int32_t sampleRate = !name.empty() ? 44100 : outputWBAMR ? 16000 : 8000; in main() local 103 int32_t bitRate = sampleRate; in main() 113 sampleRate, in main() 117 source = new SineSource(sampleRate, channels); in main() 132 meta->setInt32("sample-rate", sampleRate); in main()
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/frameworks/av/media/libaudioclient/tests/ |
D | test_create_audiotrack.cpp | 61 uint32_t sampleRate; in testTrack() local 81 &sampleRate, &format, &channelMask, in testTrack() 99 offloadInfo.sample_rate = sampleRate; in testTrack() 115 sampleRate, in testTrack()
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D | test_create_audiorecord.cpp | 63 uint32_t sampleRate; in testRecord() local 77 &sampleRate, &format, &channelMask, in testRecord() 96 sampleRate, in testRecord()
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/frameworks/av/media/libaudiohal/4.0/ |
D | StreamPowerLog.h | 43 void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) { in init() argument 51 (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000; in init() 53 sampleRate, in init()
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/frameworks/av/media/libaudiohal/2.0/ |
D | StreamPowerLog.h | 42 void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) { in init() argument 50 (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000; in init() 52 sampleRate, in init()
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/frameworks/av/media/extractors/mp3/ |
D | VBRISeeker.cpp | 51 int sampleRate; in CreateFromSource() local 52 if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) { in CreateFromSource() 72 numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate; in CreateFromSource()
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/frameworks/av/media/libstagefright/ |
D | MetaDataUtils.cpp | 56 int32_t sampleRate; in MakeAACCodecSpecificData() local 62 sampleRate = kSamplingFreq[sampling_freq_index]; in MakeAACCodecSpecificData() 97 meta.setInt32(kKeySampleRate, sampleRate); in MakeAACCodecSpecificData()
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/frameworks/av/media/libnbaio/ |
D | AudioStreamInSource.cpp | 50 uint32_t sampleRate; in negotiate() local 52 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat); in negotiate() 54 mFormat = Format_from_SR_C(sampleRate, in negotiate()
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D | AudioStreamOutSink.cpp | 48 uint32_t sampleRate; in negotiate() local 50 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat); in negotiate() 52 mFormat = Format_from_SR_C(sampleRate, in negotiate()
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/frameworks/base/media/java/android/media/ |
D | AudioFormat.java | 666 private AudioFormat(int encoding, int sampleRate, int channelMask, int channelIndexMask) { in AudioFormat() argument 668 mSampleRate = sampleRate; in AudioFormat() 973 public Builder setSampleRate(int sampleRate) throws IllegalArgumentException { in setSampleRate() argument 977 if (((sampleRate < SAMPLE_RATE_HZ_MIN) || (sampleRate > SAMPLE_RATE_HZ_MAX)) && in setSampleRate() 978 sampleRate != SAMPLE_RATE_UNSPECIFIED) { in setSampleRate() 979 throw new IllegalArgumentException("Invalid sample rate " + sampleRate); in setSampleRate() 981 mSampleRate = sampleRate; in setSampleRate()
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