| /frameworks/base/media/libmedia/ |
| AudioTrack.cpp | 55 uint32_t sampleRate)
74 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
75 afFrameCount * minBufCount * sampleRate / afSampleRate;
88 uint32_t sampleRate,
99 mStatus = set(streamType, sampleRate, format, channelMask,
106 uint32_t sampleRate,
117 mStatus = set(streamType, sampleRate, format, channelMask,
143 uint32_t sampleRate,
177 if (sampleRate == 0) {
178 sampleRate = afSampleRate
[all...] |
| /hardware/libhardware_legacy/include/hardware_legacy/ |
| AudioHardwareInterface.h | 53 virtual uint32_t sampleRate() const = 0;
124 virtual uint32_t sampleRate() const = 0;
229 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
236 uint32_t *sampleRate=0,
244 uint32_t *sampleRate,
|
| AudioHardwareBase.h | 48 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
|
| /external/srec/srec/cfront/ |
| frontobj.c | 302 waveobj->samplerate = parameters->samplerate;
358 ASSERT(parameters->samplerate);
361 freqobj->frame_period = parameters->samplerate / freqobj->framerate;
362 freqobj->samplerate = parameters->samplerate;
378 high_cut = parameters->samplerate / 2;
380 bandwidth = parameters->samplerate / 2;
394 * ((double)parameters->samplerate / (double)11025)
398 * ((double)parameters->samplerate / (double)11025
[all...] |
| spec_anl.c | 328 ASSERT(freqobj->samplerate > 0);
330 freq_step = (freqobj->samplerate << 12) / (2 * freqobj->fft.size);
337 lo = (((freq[ii] - spread[ii]) * 2 * freqobj->fft.size) + freqobj->samplerate / 2) / freqobj->samplerate;
338 hi = (((freq[ii] + spread[ii]) * 2 * freqobj->fft.size) + freqobj->samplerate / 2) / freqobj->samplerate;
|
| /hardware/msm7k/libaudio/ |
| AudioHardware.cpp | 158 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
173 status_t lStatus = out->set(this, devices, format, channels, sampleRate);
198 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
209 status_t lStatus = in->set(this, devices, format, channels, sampleRate, acoustic_flags);
346 size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
631 uint32_t AudioHardware::getInputSampleRate(uint32_t sampleRate)
638 delta = abs(sampleRate - inputSamplingRates[i]);
677 if (lRate == 0) lRate = sampleRate();
682 (lRate != sampleRate())) {
685 if (pRate) *pRate = sampleRate();
[all...] |
| /frameworks/base/services/audioflinger/ |
| AudioFlinger.h | 80 uint32_t sampleRate,
90 virtual uint32_t sampleRate(int output) const;
118 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
197 uint32_t sampleRate,
323 uint32_t sampleRate,
360 int sampleRate() const;
417 uint32_t sampleRate() const;
583 uint32_t sampleRate,
670 uint32_t sampleRate,
725 uint32_t sampleRate,
[all...] |
| AudioResampler.cpp | 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
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),
|
| /frameworks/base/media/libeffects/lvm/lib/Bass/src/ |
| LVDBE_Init.c | 208 pInstance->Params.SampleRate = LVDBE_FS_8000;
270 LVDBE_BYPASS_MIXER_TC,pInstance->Params.SampleRate,2);
280 LVDBE_BYPASS_MIXER_TC,pInstance->Params.SampleRate,2);
|
| /frameworks/base/media/libmediaplayerservice/nuplayer/ |
| NuPlayerDecoder.cpp | 124 int32_t numChannels, sampleRate;
126 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
129 msg->setInt32("sample-rate", sampleRate);
|
| /external/bluetooth/bluez/sbc/ |
| sbctester.c | 313 (int) infosref.frames, (int) infosref.samplerate,
316 (int) infostst.frames, (int) infostst.samplerate,
326 if (infosref.samplerate != infostst.samplerate ||
|
| /external/srec/srec/include/ |
| frontpar.h | 35 int samplerate; member in struct:__anon11145
|
| front.h | 130 int samplerate; member in struct:__anon11139
149 int samplerate; member in struct:__anon11140
|
| /external/webkit/Source/WebCore/platform/audio/ |
| AudioDSPKernelProcessor.h | 53 AudioDSPKernelProcessor(double sampleRate, unsigned numberOfChannels);
|
| /external/webkit/Source/WebCore/platform/audio/mac/ |
| AudioFileReaderMac.h | 49 PassOwnPtr<AudioBus> createBus(double sampleRate, bool mixToMono);
|
| /external/webkit/Source/WebCore/webaudio/ |
| AudioBasicProcessorNode.h | 42 AudioBasicProcessorNode(AudioContext*, double sampleRate);
|
| BiquadProcessor.h | 50 BiquadProcessor(FilterType, double sampleRate, size_t numberOfChannels, bool autoInitialize = true);
|
| AudioBasicProcessorNode.cpp | 39 AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, double sampleRate)
40 : AudioNode(context, sampleRate)
|
| AudioBufferSourceNode.h | 48 static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, double sampleRate);
84 AudioBufferSourceNode(AudioContext*, double sampleRate);
|
| /frameworks/base/include/media/stagefright/ |
| AudioSource.h | 37 int inputSource, uint32_t sampleRate,
|
| /frameworks/base/media/libeffects/lvm/lib/Bass/lib/ |
| LVDBE.h | 164 /* Capabilities.SampleRate = LVDBE_CAP_32000 + LVCS_DBE_44100; */
244 LVDBE_Fs_en SampleRate;
258 LVM_UINT16 SampleRate; /* Sampling rate capabilities */
405 /* SampleRate: Changing the sample rate may cause pops and clicks. */
|
| /frameworks/base/media/libstagefright/ |
| AACWriter.cpp | 201 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) {
211 if (sampleRate == kSampleRateTable[index]) {
213 sampleRate, index);
219 LOGE("Sampling rate %d bps is not supported", sampleRate);
|
| /hardware/msm7k/libaudio-qsd8k/ |
| AudioHardware.cpp | 224 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
239 status_t lStatus = out->set(this, devices, format, channels, sampleRate);
264 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
275 status_t lStatus = in->set(this, devices, format, channels, sampleRate, acoustic_flags);
557 size_t AudioHardware::getBufferSize(uint32_t sampleRate, int channelCount)
561 if (sampleRate < 11025) {
563 } else if (sampleRate < 22050) {
565 } else if (sampleRate < 32000) {
567 } else if (sampleRate < 44100) {
577 size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount
[all...] |
| /cts/apps/CtsVerifier/jni/audioquality/ |
| GlitchTest.cpp | 25 void GlitchTest::init(float sampleRate, float stimFreq, float onsetThresh,
33 mSampleRate = sampleRate;
|
| /external/srec/srec_jni/ |
| android_speech_srec_MicrophoneInputStream.cpp | 57 (JNIEnv *env, jclass clazz, jint sampleRate, jint fifoFrames) {
60 AUDIO_SOURCE_VOICE_RECOGNITION, sampleRate,
|
