| /device/google/dragon/ |
| media_profiles.xml | 37 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
92 sampleRate="8000"
105 sampleRate="48000"
118 sampleRate="8000"
131 sampleRate="8000"
144 sampleRate="48000"
157 sampleRate="48000"
170 sampleRate="48000"
187 sampleRate="8000"
204 sampleRate="48000
[all...] |
| /hardware/libhardware_legacy/audio/ |
| AudioDumpInterface.h | 42 uint32_t sampleRate);
46 virtual uint32_t sampleRate() const;
84 uint32_t sampleRate);
87 virtual uint32_t sampleRate() const;
125 uint32_t *sampleRate=0,
149 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
152 uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
|
| /frameworks/av/media/libeffects/lvm/lib/Bass/src/ |
| LVDBE_Control.c | 114 LVM_UINT16 Offset = (LVM_UINT16)((LVM_UINT16)pParams->SampleRate + \
120 LVM_UINT16 Offset = (LVM_UINT16)((LVM_UINT16)pParams->SampleRate + \
196 pInstance->pData->AGCInstance.AGC_Attack = LVDBE_AGC_ATTACK_Table[(LVM_UINT16)pParams->SampleRate]; /* Attack multiplier */
197 pInstance->pData->AGCInstance.AGC_Decay = LVDBE_AGC_DECAY_Table[(LVM_UINT16)pParams->SampleRate]; /* Decay multipler */
291 pInstance->pData->AGCInstance.VolumeTC = LVDBE_VolumeTCTable[(LVM_UINT16)pParams->SampleRate]; /* Volume update time constant */
322 (LVM_Fs_en)pInstance->Params.SampleRate,
327 (LVM_Fs_en)pInstance->Params.SampleRate,
346 /* SampleRate: Changing the sample rate may cause pops and clicks. */
385 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
[all...] |
| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/audiolib/ |
| StreamRecorder.java | 57 public static int calcNumBufferBytes(int numChannels, int sampleRate, int encoding) {
62 numBytes = AudioRecord.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_IN_STEREO,
66 numBytes = AudioRecord.getMinBufferSize(sampleRate,
73 public static int calcNumBufferFrames(int numChannels, int sampleRate, int encoding) {
74 return calcNumBufferBytes(numChannels, sampleRate, encoding) /
117 private boolean open_internal(int numChans, int sampleRate) {
118 Log.i(TAG, "StreamRecorder.open_internal(chans:" + numChans + ", rate:" + sampleRate);
121 mSampleRate = sampleRate;
144 public boolean open(int numChans, int sampleRate, int numBurstFrames) {
145 boolean sucess = open_internal(numChans, sampleRate);
[all...] |
| StreamPlayer.java | 90 private static int calcNumBufferBytes(int sampleRate, int numChannels, int encoding) {
91 return AudioTrack.getMinBufferSize(sampleRate,
96 private static int calcNumBufferFrames(int sampleRate, int numChannels, int encoding) {
97 return calcNumBufferBytes(sampleRate, numChannels, encoding)
106 public boolean open(int numChans, int sampleRate, int numBurstFrames, AudioFiller filler) {
107 // Log.i(TAG, "StreamPlayer.open(chans:" + numChans + ", rate:" + sampleRate +
111 mSampleRate = sampleRate;
115 calcNumBufferFrames(sampleRate, numChans, AudioFormat.ENCODING_PCM_FLOAT);
|
| /cts/tests/tests/media/src/android/media/cts/ |
| AudioRecordNative.java | 43 public boolean open(int numChannels, int sampleRate, boolean useFloat, int numBuffers) {
44 return open(numChannels, 0, sampleRate, useFloat,numBuffers);
47 public boolean open(int numChannels, int channelMask, int sampleRate,
50 sampleRate, useFloat, numBuffers) == STATUS_OK) {
111 public static boolean test(int numChannels, int sampleRate, boolean useFloat,
113 return test(numChannels, 0, sampleRate, useFloat, msecPerBuffer, numBuffers);
116 public static boolean test(int numChannels, int channelMask, int sampleRate, boolean useFloat,
118 return nativeTest(numChannels, channelMask, sampleRate, useFloat, msecPerBuffer, numBuffers)
147 int sampleRate, boolean useFloat, int numBuffers);
164 int numChannels, int channelMask, int sampleRate,
[all...] |
| AudioTrackNative.java | 46 public boolean open(int numChannels, int sampleRate, boolean useFloat, int numBuffers) {
47 return open(numChannels, 0, sampleRate, useFloat, numBuffers);
50 public boolean open(int numChannels, int channelMask, int sampleRate,
53 sampleRate, useFloat, numBuffers) == STATUS_OK) {
118 public static boolean test(int numChannels, int sampleRate, boolean useFloat,
120 return test(numChannels, 0, sampleRate, useFloat, msecPerBuffer, numBuffers);
123 public static boolean test(int numChannels, int channelMask, int sampleRate, boolean useFloat,
125 return nativeTest(numChannels, channelMask, sampleRate, useFloat, msecPerBuffer, numBuffers)
154 int sampleRate, boolean useFloat, int numBuffers);
171 int numChannels, int channelMask, int sampleRate,
[all...] |
| /frameworks/av/media/libaudioprocessing/tests/ |
| 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)
[all...] |
| /external/sonic/ |
| main.c | 26 int sampleRate,
29 sonicStream stream = sonicCreateStream(sampleRate, numChannels);
83 int sampleRate, numChannels;
125 inFile = openInputWaveFile(inFileName, &sampleRate, &numChannels);
129 outFile = openOutputWaveFile(outFileName, sampleRate, numChannels);
135 sampleRate, numChannels);
|
| Main.java | 29 int sampleRate,
32 Sonic sonic = new Sonic(sampleRate, numChannels);
72 int sampleRate = (int)format.getSampleRate();
80 sampleRate, numChannels);
|
| /frameworks/av/media/libeffects/testlibs/ |
| AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate)
45 : mBiquad(nChannels, sampleRate) {
46 configure(nChannels, sampleRate);
50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) {
51 mNiquistFreq = sampleRate * 500;
53 mBiquad.configure(nChannels, sampleRate);
|
| AudioShelvingFilter.cpp | 50 int sampleRate)
52 mBiquad(nChannels, sampleRate) {
53 configure(nChannels, sampleRate);
56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) {
57 mNiquistFreq = sampleRate * 500;
59 mBiquad.configure(nChannels, sampleRate);
|
| 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) {
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);
[all...] |
| /hardware/qcom/media/msm8998/conf_files/msm8937/ |
| media_profiles_8937.xml | 40 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
107 sampleRate="8000"
120 sampleRate="48000"
133 sampleRate="48000"
146 sampleRate="8000"
159 sampleRate="48000"
172 sampleRate="48000"
185 sampleRate="48000"
198 sampleRate="8000"
211 sampleRate="48000
[all...] |
| /hardware/qcom/media/sdm845/conf_files/msm8937/ |
| media_profiles_8937.xml | 40 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
107 sampleRate="8000"
120 sampleRate="48000"
133 sampleRate="48000"
146 sampleRate="8000"
159 sampleRate="48000"
172 sampleRate="48000"
185 sampleRate="48000"
198 sampleRate="8000"
211 sampleRate="48000
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Reverb/src/ |
| LVREV_ApplyNewSettings.c | 77 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
83 Omega = LVM_GetOmega(pPrivate->NewParams.HPF, pPrivate->NewParams.SampleRate);
96 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
106 if(pPrivate->NewParams.LPF <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
108 Omega = LVM_GetOmega(pPrivate->NewParams.LPF, pPrivate->NewParams.SampleRate);
143 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
149 LVM_INT32 Fs = LVM_GetFsFromTable(pPrivate->NewParams.SampleRate);
[all...] |
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| psy_configuration.h | 91 Word32 GetSRIndex(Word32 sampleRate);
95 Word32 samplerate,
100 Word32 samplerate,
|
| psy_main.h | 50 Word32 sampleRate,
67 Word32 sampleRate);
|
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Control.c | 87 if (pParams->SampleRate != pInstance->Params.SampleRate)
89 pInstance->TimerParams.SamplingRate = LVCS_SampleRateTable[pParams->SampleRate];
103 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
142 LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
144 LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
158 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
160 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
|
| /frameworks/base/core/java/android/bluetooth/ |
| BluetoothAudioConfig.java | 35 public BluetoothAudioConfig(int sampleRate, int channelConfig, int audioFormat) {
36 mSampleRate = sampleRate;
70 int sampleRate = in.readInt();
73 return new BluetoothAudioConfig(sampleRate, channelConfig, audioFormat);
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| aacenc.c | 148 config.sampleRate = 44100;
280 pOutInfo->Format.SampleRate = hAacEnc->config.sampleRate;
340 config.sampleRate = pAAC_param->sampleRate;
347 /* check the samplerate */
351 if(config.sampleRate == sampRateTab[i])
363 if(config.sampleRate%8000 == 0)
368 (config.bitRate > config.sampleRate*6*config.nChannelsOut)))
370 config.bitRate = 640*config.sampleRate/tmp*config.nChannelsOut
[all...] |
| /device/generic/goldfish/camera/ |
| media_profiles.xml | 38 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
90 sampleRate="8000"
103 sampleRate="8000"
124 sampleRate="8000"
137 sampleRate="8000"
158 sampleRate="8000"
171 sampleRate="8000"
192 sampleRate="8000"
205 sampleRate="8000"
226 sampleRate="8000
[all...] |
| /hardware/qcom/audio/legacy/alsa_sound/ |
| AudioHardwareALSA.cpp | 724 uint32_t *sampleRate,
728 ALOGV("openOutputStream: devices 0x%x channels %d sampleRate %d",
729 devices, *channels, *sampleRate);
747 ((*sampleRate == VOIP_SAMPLING_RATE_8K) || (*sampleRate == VOIP_SAMPLING_RATE_16K))) {
762 if(*sampleRate == VOIP_SAMPLING_RATE_8K) {
765 else if(*sampleRate == VOIP_SAMPLING_RATE_16K) {
769 ALOGE("unsupported samplerate %d for voip",*sampleRate);
783 alsa_handle.sampleRate = *sampleRate
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Common/lib/ |
| Filter.h | 54 LVM_Fs_en SampleRate);
75 LVM_Fs_en SampleRate);
|
| /frameworks/av/media/libstagefright/codecs/aacenc/SampleCode/ |
| AAC_E_SAMPLES.c | 38 "voAACEncTest -if <inputfile.pcm> -of <outputfile.aac> -sr <samplerate> -ch <channel> -br <bitrate> -adts <adts> \n"
41 "-sr input pcm samplerate, default 44100 \n"
43 "-br encoded aac bitrate, default 64000 * (samplerate/100)*channel/441(480)\n"
53 // bitRate/nChannels < sampleRate*6
57 param->sampleRate = 44100;
84 param->sampleRate = atoi(*argv);
116 if(param->sampleRate%8000 == 0)
118 param->bitRate = 640*param->nChannels*param->sampleRate/scale;
|
