| /hardware/qcom/msm8960/original-kernel-headers/linux/ |
| msm_audio_wmapro.h | 14 unsigned short samplingrate; member in struct:msm_audio_wmapro_config
|
| msm_audio_wma.h | 25 uint32_t samplingrate; member in struct:msm_audio_wma_config_v2
|
| /hardware/qcom/msm8994/original-kernel-headers/linux/ |
| msm_audio_wmapro.h | 14 unsigned short samplingrate; member in struct:msm_audio_wmapro_config
|
| msm_audio_wma.h | 25 uint32_t samplingrate; member in struct:msm_audio_wma_config_v2
|
| /hardware/qcom/msm8996/original-kernel-headers/linux/ |
| msm_audio_wmapro.h | 14 uint32_t samplingrate; member in struct:msm_audio_wmapro_config
|
| msm_audio_wma.h | 25 uint32_t samplingrate; member in struct:msm_audio_wma_config_v2
|
| /hardware/qcom/msm8x84/original-kernel-headers/linux/ |
| msm_audio_wmapro.h | 14 unsigned short samplingrate; member in struct:msm_audio_wmapro_config
|
| msm_audio_wma.h | 25 uint32_t samplingrate; member in struct:msm_audio_wma_config_v2
|
| /cts/suite/audio_quality/test_description/processing/ |
| calc_thd.py | 23 def calc_thd(data, signalFrequency, samplingRate, frequencyMargin):
28 baseI = fftLen * signalFrequency * 2 / samplingRate
49 samplingRate = 44100
52 samples = float(samplingRate) * float(durationInSec)
54 time = index / samplingRate
55 multiplier = 2.0 * np.pi * signalFrequency / float(samplingRate)
57 thd = calc_thd(data, signalFrequency, samplingRate, 0.02)
|
| calc_delay.py | 62 samplingRate = 44100
67 samples = float(samplingRate) * float(durationInSec)
69 time = index / samplingRate
70 multiplier = 2.0 * np.pi * signalFrequency / float(samplingRate)
72 DELAY = durationInSec / 2.0 * samplingRate
|
| check_spectrum.py | 39 def do_check_spectrum(hostData, DUTData, samplingRate, fLow, fHigh, margainLow, margainHigh):
42 iLow = N * fLow / samplingRate + 1 # 1 for DC
45 iHigh = N * fHigh / samplingRate + 1 # 1 for DC
48 print fLow, iLow, fHigh, iHigh, samplingRate
50 Phh, freqs = plt.psd(hostData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\
53 Pdd, freqs = plt.psd(DUTData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\
113 samplingRate = inputData[2]
133 samplingRate, fLow, fHigh, margainLow, margainHigh)
155 samplingRate = 44100
158 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHigh,
[all...] |
| check_spectrum_playback.py | 38 def do_check_spectrum_playback(hostData, samplingRate, fLow, fHigh, margainLow, margainHigh):
41 iLow = N * fLow / samplingRate + 1 # 1 for DC
44 iHigh = N * fHigh / samplingRate + 1 # 1 for DC
47 print fLow, iLow, fHigh, iHigh, samplingRate
49 Phh, freqs = plt.psd(hostData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\
93 samplingRate = inputData[1]
99 samplingRate, fLow, fHigh, margainLow, margainHigh)
121 samplingRate = 44100
124 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHigh,\
127 (passFail, minVal, maxVal, amp) = do_check_spectrum_playback(data, samplingRate, fLow,
[all...] |
| gen_random.py | 30 def do_gen_random(peakAmpl, durationInMSec, samplingRate, fHigh, stereo=True):
31 samples = durationInMSec * samplingRate / 1000
36 iHigh = freqSamples * fHigh * 2 / samplingRate + 1
48 #freq = np.linspace(0.0, samplingRate, num=len(fftData), endpoint=False)
94 samplingRate = 44100
98 result = do_gen_random(peakAmplitude, durationInMSec, samplingRate, fHigh)
|
| playback_sample.py | 46 samplingRate = 44100
48 freq = calc_freq(hostRecording, samplingRate)
61 def calc_freq(recording, samplingRate):
|
| /cts/suite/audio_quality/lib/include/audio/ |
| AudioLocal.h | 36 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int gain,
46 virtual bool doPrepare(AudioHardware::SamplingRate, int samplesInOneGo) = 0;
71 AudioHardware::SamplingRate mSamplingRate;
|
| AudioRemote.h | 29 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int volume,
38 AudioHardware::SamplingRate mSamplingRate;
|
| AudioSignalFactory.h | 31 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples,
|
| /frameworks/base/media/java/android/media/ |
| AudioDevicePortConfig.java | 29 AudioDevicePortConfig(AudioDevicePort devicePort, int samplingRate, int channelMask,
31 super((AudioPort)devicePort, samplingRate, channelMask, format, gain);
35 this(config.port(), config.samplingRate(), config.channelMask(), config.format(),
|
| /frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
| AudioProfile.cpp | 31 status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
36 supportsRate(samplingRate)) {
63 status_t AudioProfile::checkCompatibleSamplingRate(uint32_t samplingRate,
66 ALOG_ASSERT(samplingRate > 0);
69 updatedSamplingRate = samplingRate;
76 size_t orderOfDesiredRate = mSamplingRates.orderOf(samplingRate);
81 if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
89 if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
228 status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
238 if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR)
[all...] |
| /cts/suite/audio_quality/lib/src/audio/ |
| AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode)
27 mSamplingRate = samplingRate;
|
| /cts/suite/audio_quality/test/ |
| AudioLocalTest.cpp | 30 AudioHardware::SamplingRate mSamplingRate;
48 virtual bool doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) {
|
| /external/libgdx/backends/gdx-backend-android/src/com/badlogic/gdx/backends/android/ |
| AndroidAudioRecorder.java | 32 public AndroidAudioRecorder (int samplingRate, boolean isMono) {
34 int minBufferSize = AudioRecord.getMinBufferSize(samplingRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
35 recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, samplingRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT,
|
| /external/libgdx/backends/gdx-backend-lwjgl/src/com/badlogic/gdx/backends/lwjgl/audio/ |
| JavaSoundAudioRecorder.java | 32 public JavaSoundAudioRecorder (int samplingRate, boolean isMono) {
34 AudioFormat format = new AudioFormat(Encoding.PCM_SIGNED, samplingRate, 16, isMono ? 1 : 2, isMono ? 2 : 4,
35 samplingRate, false);
|
| /external/libgdx/backends/gdx-backend-lwjgl3/src/com/badlogic/gdx/backends/lwjgl3/audio/ |
| JavaSoundAudioRecorder.java | 32 public JavaSoundAudioRecorder (int samplingRate, boolean isMono) {
34 AudioFormat format = new AudioFormat(Encoding.PCM_SIGNED, samplingRate, 16, isMono ? 1 : 2, isMono ? 2 : 4,
35 samplingRate, false);
|
| /external/libgdx/backends/gdx-backend-headless/src/com/badlogic/gdx/backends/headless/mock/audio/ |
| MockAudio.java | 32 public AudioDevice newAudioDevice(int samplingRate, boolean isMono) {
37 public AudioRecorder newAudioRecorder(int samplingRate, boolean isMono) {
|
