/frameworks/av/media/libaudioprocessing/ |
D | AudioResamplerFirProcess.h | 79 template<int CHANNELS, typename TO> 80 class Accumulator : public Accumulator<CHANNELS-1, TO> // recursive 85 Accumulator<CHANNELS-1, TO>::clear(); in clear() 90 Accumulator<CHANNELS-1, TO>::acc(coef, data); in acc() 94 Accumulator<CHANNELS-1, TO>::volume(out, gain); in volume() 177 template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, 189 static_assert(CHANNELS > 0, "CHANNELS must be > 0"); in ProcessBase() 191 if (CHANNELS > 2) { in ProcessBase() 193 Accumulator<CHANNELS, TO> accum; in ProcessBase() 205 sP -= CHANNELS; in ProcessBase() [all …]
|
D | AudioResamplerSinc.cpp | 293 template<int CHANNELS> 298 const size_t headOffset = c.halfNumCoefs*CHANNELS; in resample() 319 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample() 322 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample() 328 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample() 337 for (size_t i=0 ; i<CHANNELS ; i++) { in resample() 338 head[i] = in[inputIndex*CHANNELS + i]; in resample() 343 filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL); in resample() 353 read<CHANNELS>(impulse, phaseFraction, in, inputIndex); in resample() 368 return outputIndex / CHANNELS; in resample() [all …]
|
D | AudioResamplerFirProcessNeon.h | 73 template <int CHANNELS, int STRIDE, bool FIXED> 86 static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2"); in ProcessNeonIntrinsic() 88 sP -= CHANNELS*((STRIDE>>1)-1); in ProcessNeonIntrinsic() 103 if (CHANNELS == 2) { in ProcessNeonIntrinsic() 127 switch (CHANNELS) { in ProcessNeonIntrinsic() 168 if (CHANNELS == 1) { in ProcessNeonIntrinsic() 171 } else if (CHANNELS == 2) { in ProcessNeonIntrinsic() 181 template <int CHANNELS, int STRIDE, bool FIXED> 194 static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2"); in ProcessNeonIntrinsic() 196 sP -= CHANNELS*((STRIDE>>1)-1); in ProcessNeonIntrinsic() [all …]
|
D | AudioResamplerDyn.cpp | 85 void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs) in resize() argument 88 size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength; in resize() 93 && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) { in resize() 107 TI* srcLo = mImpulse - halfNumCoefs*CHANNELS; in resize() 108 TI* srcHi = mImpulse + halfNumCoefs*CHANNELS; in resize() 125 mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed in resize() 126 mRingFull = state + mStateCount - halfNumCoefs*CHANNELS; in resize() 131 template<int CHANNELS> 135 TI* head = impulse + halfNumCoefs*CHANNELS; in readAgain() 136 for (size_t i=0 ; i<CHANNELS ; i++) { in readAgain() [all …]
|
D | AudioResamplerFirProcessSSE.h | 37 template <int CHANNELS, int STRIDE, bool FIXED> 50 static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2"); in ProcessSSEIntrinsic() 52 sP -= CHANNELS*(4-1); // adjust sP for a loop iteration of four in ProcessSSEIntrinsic() 61 if (CHANNELS == 2) { in ProcessSSEIntrinsic() 94 switch (CHANNELS) { in ProcessSSEIntrinsic() 157 if (CHANNELS == 1) { in ProcessSSEIntrinsic() 161 } else if (CHANNELS == 2) { in ProcessSSEIntrinsic()
|
D | AudioResamplerSinc.h | 49 template<int CHANNELS> 53 template<int CHANNELS> 57 template<int CHANNELS> 63 template<int CHANNELS>
|
D | AudioResamplerDyn.h | 118 void resize(int CHANNELS, int halfNumCoefs); 129 template<int CHANNELS> 133 template<int CHANNELS> 155 template<int CHANNELS, bool LOCKED, int STRIDE>
|
/frameworks/av/services/audioflinger/afutils/ |
D | BufLog.h | 80 #define __BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE) \ argument 81 BufLogSingleton::instance()->write(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, \ 84 #define BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE) \ argument 87 #define BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE) \ argument 88 __BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE)
|
/frameworks/av/media/libstagefright/httplive/fuzzer/corpus/ |
D | index7 | 7 …AUDIO,GROUP-ID="audio-aacl-64",LANGUAGE="en",NAME="English",DEFAULT=YES,AUTOSELECT=YES,CHANNELS="2" 8 …UDIO,GROUP-ID="audio-aacl-128",LANGUAGE="en",NAME="English",DEFAULT=YES,AUTOSELECT=YES,CHANNELS="2"
|
/frameworks/base/core/java/android/app/ |
D | NotificationChannelGroup.java | 360 channel.dumpDebug(proto, NotificationChannelGroupProto.CHANNELS); in dumpDebug()
|
/frameworks/base/media/java/android/media/ |
D | AudioRecord.java | 2574 public static final String CHANNELS = MM_PREFIX + "channels"; field in AudioRecord.MetricsConstants
|
/frameworks/av/media/libaudioclient/tests/ |
D | audio_aidl_legacy_conversion_tests.cpp | 683 testing::Values(AudioGainMode::JOINT, AudioGainMode::CHANNELS,
|
/frameworks/av/media/audioaidlconversion/ |
D | AidlConversionCppNdk.cpp | 1211 case AudioGainMode::CHANNELS: in aidl2legacy_AudioGainMode_audio_gain_mode_t() 1225 return AudioGainMode::CHANNELS; in legacy2aidl_audio_gain_mode_t_AudioGainMode()
|
/frameworks/base/services/core/java/com/android/server/notification/ |
D | PreferencesHelper.java | 2189 channel.dumpDebug(proto, RankingHelperProto.RecordProto.CHANNELS); in dumpPackagePreferencesLocked()
|
/frameworks/base/tools/aapt2/integration-tests/CommandTests/ |
D | android-33.jar | AndroidManifest.xml
META-INF/
META-INF/MANIFEST.MF
NOTICES/
NOTICES/libcore ... |
/frameworks/base/boot/hiddenapi/ |
D | hiddenapi-max-target-o.txt | 8061 Landroid/app/NotificationChannelGroupProto;->CHANNELS:J 49178 Landroid/service/notification/RankingHelperProto$RecordProto;->CHANNELS:J
|
/frameworks/base/core/api/ |
D | current.txt | 21794 field public static final String CHANNELS = "android.media.audiorecord.channels";
|