/device/samsung/crespo/libaudio/ |
AudioHardware.h | 36 struct pcm; 59 // Kernel pcm out buffer size in frames at 44.1kHz 71 // Kernel pcm in buffer size in frames at 44.1kHz (before resampling) 136 struct pcm *openPcmOut_l(); 166 struct pcm* mPcm; 261 struct pcm *mPcm; 344 struct pcm *mPcm;
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/frameworks/base/media/java/android/media/ |
ResampleInputStream.java | 37 // pcm input stream 44 // input pcm data 56 * @param inputStream InputStream containing 16 bit PCM.
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/frameworks/media/libvideoeditor/vss/common/inc/ |
M4PCMR_CoreReader.h | 80 * - M4WAVR_kPCMblockSize: sets the size of the PCM block to read 107 M4OSA_UInt32 m_offset; /**< Offset of the PCM read (i.e m_offset of the 110 M4OSA_FilePosition m_dataStartOffset;/**< offset of the pcm data beginning into
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/cts/apps/CtsVerifier/jni/audioquality/ |
LinearityTest.cpp | 54 sampleRate in pcm: the RMS of the highest amplitude 30ms segment 59 static int peakLevels(short* pcm, int numSamples, float sampleRate, 77 double s = pcm[i]; 95 if (pcm[i] >= 0) { 96 bubbleUp(maxVal, topM, pcm[i]); 98 bubbleUp(maxVal, topM, -pcm[i]);
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/frameworks/base/core/java/android/speech/srec/ |
WaveHeader.java | 27 * <li> format - usually PCM, ALAW or ULAW. 30 * <li> bitsPerSample - usually 16 for PCM, 8 for ALAW, or 8 for ULAW. 45 /** Indicates PCM format. */ 70 * @param bitsPerSample usually 16 for PCM, 8 for ULAW or 8 for ALAW. 140 * usually 16 for PCM, 8 for ULAW or 8 for ALAW. 149 * usually 16 for PCM, 8 for ULAW or 8 for ALAW.
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/device/samsung/tuna/audio/ |
audio_hw.c | 488 struct pcm *pcm_modem_dl; 489 struct pcm *pcm_modem_ul; 518 struct pcm *pcm[PCM_TOTAL]; member in struct:tuna_stream_out 535 struct pcm *pcm; member in struct:tuna_stream_in 623 /* Open modem PCM channels */ 627 LOGE("cannot open PCM modem DL stream: %s", pcm_get_error(adev->pcm_modem_dl)); 635 LOGE("cannot open PCM modem UL stream: %s", pcm_get_error(adev->pcm_modem_ul)); [all...] |
/cts/apps/CtsVerifier/src/com/android/cts/verifier/audioquality/experiments/ |
SoundLevelExperiment.java | 62 short[] pcm = Utils.byteToShortArray(record); local 63 float[] results = mNative.measureRms(pcm, AudioQualityVerifierActivity.SAMPLE_RATE, ONSET_THRESH);
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SpectrumShapeExperiment.java | 51 short[] pcm = Utils.byteToShortArray(record); local 53 float[] ret = mNative.compareSpectra(pcm, refPcm, AudioQualityVerifierActivity.SAMPLE_RATE);
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/prebuilt/linux-x86/toolchain/i686-linux-glibc2.7-4.4.3/sysroot/usr/include/alsa/ |
pcm_rate.h | 11 * ALSA external PCM rate-converter plugin SDK (draft version) 108 * Define the object entry for external PCM rate-converter plugins
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/system/media/wilhelm/src/android/include/ |
AacBqToPcmCbRenderer.h | 28 // the PCM samples through a registered callback (just like its superclass, AudioToCbRenderer). 30 // of the data source creation, but all other behavior remains the same (e.g. PCM format metadata)
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/system/media/wilhelm/tests/examples/ |
slesTestDecodeToBuffQueue.cpp | 29 Signed 16-bit PCM 69 /* metadata key index for the PCM format information we want to retrieve */ 72 /* size of the struct to retrieve the PCM format metadata values: the values we're interested in 79 /* we only want to query / display the PCM format once */ 242 /* Example: query of the decoded PCM format */ 251 // but the call was successful for the PCM format keys, so those conditions are implied 296 SLDataFormat_PCM pcm; local 311 /* allocate memory to receive the PCM format metadata */ 346 pcm.formatType = SL_DATAFORMAT_PCM; 348 pcm.numChannels = 1 [all...] |
slesTestDecodeAac.cpp | 26 Enqueueing initial empty buffers to receive decoded PCM data 0 1 38 Signed 16-bit PCM 40 1 Channel (Mono) / 2 Channels (Stereo) based on the PCM information obtained when decoding 41 Sample rate based on the PCM information obtained when decoding 74 /* Size of the decoded PCM buffer queue */ 76 /* Size of each PCM buffer in the queue */ 79 /* Local storage for decoded PCM audio data */ 86 /* metadata key index for the PCM format information we want to retrieve */ 93 /* size of the struct to retrieve the PCM format metadata values: the values we're interested in 100 /* we only want to query / display the PCM format once * [all...] |
slesTestFeedback.cpp | 294 SLDataFormat_PCM pcm; local 301 pcm.formatType = SL_DATAFORMAT_PCM; 302 pcm.numChannels = channels; 303 pcm.samplesPerSec = sampleRate * 1000; 304 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 305 pcm.containerSize = 16; 306 pcm.channelMask = channels == 1 ? SL_SPEAKER_FRONT_CENTER : 308 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 310 audiosrc.pFormat = &pcm; 350 audiosnk.pFormat = &pcm; [all...] |
slesTestRecBuffQueue.cpp | 29 Signed 16-bit PCM 167 SLDataFormat_PCM pcm; local 204 pcm.formatType = SL_DATAFORMAT_PCM; 205 pcm.numChannels = 1; 206 pcm.samplesPerSec = SL_SAMPLINGRATE_22_05; 207 pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 208 pcm.containerSize = 16; 209 pcm.channelMask = SL_SPEAKER_FRONT_LEFT; 210 pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 213 recDest.pFormat = (void * ) &pcm; [all...] |
/external/clang/test/Modules/ |
macros.c | 1 // RUN: %clang_cc1 -emit-module -o %t/macros.pcm -DMODULE %s
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objc-categories.m | 2 // RUN: %clang_cc1 -emit-module -o %t/diamond_top.pcm %s -D MODULE_TOP 3 // RUN: %clang_cc1 -fmodule-cache-path %t -fdisable-module-hash -emit-module -o %t/diamond_left.pcm %s -D MODULE_LEFT 4 // RUN: %clang_cc1 -fmodule-cache-path %t -fdisable-module-hash -emit-module -o %t/diamond_right.pcm %s -D MODULE_RIGHT 5 // RUN: %clang_cc1 -fmodule-cache-path %t -fdisable-module-hash -emit-module -o %t/diamond_bottom.pcm %s -D MODULE_BOTTOM
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/external/kernel-headers/original/linux/ |
a1026.h | 94 /* Get/Set PCM Device Parameter ID List */ 95 /* PCM-0 */ 103 /* PCM-1 */ 111 /* Possible setting values for PCM I/F */
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/frameworks/base/include/media/ |
Visualizer.h | 59 CAPTURE_WAVEFORM = 0x00000001, // capture callback returns a PCM wave form 84 // callback used to return periodic PCM or FFT captures to the application. Either one or both 85 // types of data are returned (PCM and FFT) according to flags indicated when installing the 111 // return a capture in PCM 8 bit unsigned format. The size of the capture is equal to
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/frameworks/base/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" 42 "-ch input pcm channel, default 2 channel \n" 46 "./voAACEncTest -if raw.pcm -of raw.aac -sr 44100 -ch 2 -br 128000\n";
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/system/media/wilhelm/src/android/ |
android_defs.h | 26 // audio player, playing PCM buffers in a buffer queue data source 30 // audio player, decoding from a URI or FD data source to a buffer queue data sink in PCM format 37 // PCM buffer queue data sink 40 // to a buffer queue data sink in PCM format
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/system/media/wilhelm/src/desktop/ |
OutputMixExt.h | 35 /** \brief Track describes each PCM input source to OutputMix */
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/external/webrtc/src/modules/audio_processing/aecm/main/matlab/matlab/ |
mainProgram.m | 8 % fid=fopen('./htcTouchHd/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid); 9 % fid=fopen('./htcTouchHd/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid); 10 % fid=fopen('./samsungBlackjack/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid); 11 % fid=fopen('./samsungBlackjack/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid); 12 % fid=fopen('aecFarPoor.pcm'); xFar=fread(fid,'short'); fclose(fid); 13 % fid=fopen('aecNearPoor.pcm'); yNear=fread(fid,'short'); fclose(fid); 14 % fid=fopen('out_aes.pcm'); outAES=fread(fid,'short'); fclose(fid); 15 fid=fopen('aecFar4.pcm'); xFar=fread(fid,'short'); fclose(fid); 16 fid=fopen('aecNear4.pcm'); yNear=fread(fid,'short'); fclose(fid); 271 fid=fopen('aecOutMatlabC.pcm','w');fwrite(fid,int16(emicrophone),'short');fclose(fid) [all...] |
/prebuilt/linux-x86/toolchain/i686-linux-glibc2.7-4.4.3/sysroot/usr/lib/ |
libasound.so | |
libasound.so.2 | |
libasound.so.2.0 | |