/frameworks/av/media/libstagefright/ |
H A D | AudioSource.cpp | 51 audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) 54 mSampleRate(sampleRate), 58 ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount); 63 sampleRate, 78 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 50 AudioSource( audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) argument
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H A D | Utils.cpp | 89 int32_t numChannels, sampleRate; local 91 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 94 msg->setInt32("sample-rate", sampleRate); 380 int32_t sampleRate; local 381 if (msg->findInt32("sample-rate", &sampleRate)) { 382 meta->setInt32(kKeySampleRate, sampleRate);
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H A D | AACWriter.cpp | 210 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) { argument 220 if (sampleRate == kSampleRateTable[index]) { 222 sampleRate, index); 228 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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H A D | AMRWriter.cpp | 91 int32_t sampleRate; local 94 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 95 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | AMPEG4ElementaryAssembler.cpp | 89 static bool GetSampleRateIndex(int32_t sampleRate, size_t *tableIndex) { argument 99 if (sampleRate == kSampleRateTable[index]) { 189 int32_t sampleRate, numChannels; local 191 desc.c_str(), &sampleRate, &numChannels); 194 CHECK(GetSampleRateIndex(sampleRate, &mSampleRateIndex));
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/frameworks/av/media/libmedia/ |
H A D | SoundPool.cpp | 497 uint32_t sampleRate; local 503 p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format); 505 p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format); 514 ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d", 515 p->pointer(), p->size(), sampleRate, numChannels); 517 if (sampleRate > kMaxSampleRate) { 518 ALOGE("Sample rate (%u) out of range", sampleRate); 533 mSampleRate = sampleRate; 581 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rat local 845 uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5); local [all...] |
H A D | IAudioFlinger.cpp | 89 uint32_t sampleRate, 105 data.writeInt32(sampleRate); 138 uint32_t sampleRate, 152 data.writeInt32(sampleRate); 180 virtual uint32_t sampleRate(audio_io_handle_t output) const function in class:android::BpAudioFlinger 356 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, argument 361 data.writeInt32(sampleRate); 723 uint32_t sampleRate = data.readInt32(); local 734 (audio_stream_type_t) streamType, sampleRate, format, 745 uint32_t sampleRate local 86 createTrack( pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, track_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, int *sessionId, status_t *status) argument 135 openRecord( pid_t pid, audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, track_flags_t flags, pid_t tid, int *sessionId, status_t *status) argument 874 uint32_t sampleRate = data.readInt32(); local [all...] |
/frameworks/av/libvideoeditor/lvpp/ |
H A D | VideoEditorPlayer.cpp | 393 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 407 ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount); 422 frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate; 441 sampleRate, 451 sampleRate, 466 mMsecsPerFrame = 1.e3 / (float) sampleRate; 392 open( uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, audio_format_t format, int bufferCount, AudioCallback cb, void *cookie, audio_output_flags_t flags) argument
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/frameworks/av/services/audioflinger/ |
H A D | FastMixer.cpp | 64 unsigned sampleRate = 0; local 202 sampleRate = 0; 205 sampleRate = Format_sampleRate(format); 208 dumpState->mSampleRate = sampleRate; 217 if (frameCount > 0 && sampleRate > 0) { 221 mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks); 223 periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00 224 underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75 225 overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50 226 forceNs = (frameCount * 950000000LL) / sampleRate; // 0.9 [all...] |
H A D | AudioMixer.h | 39 AudioMixer(size_t frameCount, uint32_t sampleRate, 185 uint32_t sampleRate; member in struct:android::AudioMixer::track_t 199 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
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H A D | AudioFlinger.h | 92 uint32_t sampleRate, 106 uint32_t sampleRate, 115 virtual uint32_t sampleRate(audio_io_handle_t output) const; 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 388 uint32_t sampleRate, 424 int sampleRate() const; // FIXME inline after cblk sr moved 556 uint32_t sampleRate() const { return mSampleRate; } function in class:android::AudioFlinger::ThreadBase 780 uint32_t sampleRate, 904 uint32_t sampleRate, 947 uint32_t sampleRate, [all...] |
/frameworks/av/include/media/stagefright/ |
H A D | AudioSource.h | 38 uint32_t sampleRate,
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H A D | ACodec.h | 223 int32_t numChannels, int32_t sampleRate, int32_t bitRate, 233 bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel); 236 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
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/frameworks/av/include/private/media/ |
H A D | AudioTrackShared.h | 92 uint32_t sampleRate; member in struct:android::audio_track_cblk_t
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/frameworks/av/media/libstagefright/codecs/aacenc/ |
H A D | AACEncoder.cpp | 84 params.sampleRate = mSampleRate; 96 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) { argument 103 if (sampleRate == kSampleRateTable[i]) { 109 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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H A D | SoftAACEncoder.cpp | 324 params.sampleRate = mSampleRate; 337 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) { argument 346 if (sampleRate == kSampleRateTable[i]) {
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/frameworks/av/include/media/ |
H A D | SoundPool.h | 58 int sampleRate() { return mSampleRate; } function in class:android::Sample 68 void init(int numChannels, int sampleRate, audio_format_t format, size_t size, sp<IMemory> data ) { argument 69 mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; mData = data; }
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/frameworks/wilhelm/tests/examples/ |
H A D | slesTestFeedback.cpp | 38 static SLuint32 sampleRate = 44100; // -s# variable 214 sampleRate = atoi(&arg[2]); 215 switch (sampleRate) { 228 (unsigned) sampleRate); 303 pcm.samplesPerSec = sampleRate * 1000;
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/frameworks/av/include/media/nbaio/ |
H A D | NBAIO.h | 68 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
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/frameworks/av/media/libmediaplayerservice/ |
H A D | MediaPlayerService.cpp | 1212 *pSampleRate = cache->sampleRate(); 1215 ALOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat); 1262 *pSampleRate = cache->sampleRate(); 1265 ALOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat); 1374 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 1388 ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask, 1401 frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate; 1417 sampleRate, 1429 sampleRate, 1457 } else if ((mRecycledTrack->getSampleRate() != sampleRate) || 1373 open( uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, audio_format_t format, int bufferCount, AudioCallback cb, void *cookie, audio_output_flags_t flags) argument 1774 open( uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, audio_format_t format, int bufferCount, AudioCallback cb, void *cookie, audio_output_flags_t flags) argument [all...] |
H A D | MediaPlayerService.h | 94 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 195 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 209 uint32_t sampleRate() const { return mSampleRate; } function in class:android::MediaPlayerService::AudioCache
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | tns.c | 134 Word32 sampleRate, /*!< Sampling frequency */ 162 tC->tnsStartBand = FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 166 sampleRate, 171 sampleRate, 201 Word32 sampleRate, /*!< Sampling frequency */ 228 tC->tnsStartBand=FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 232 sampleRate, 237 sampleRate, 133 InitTnsConfigurationLong(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_LONG *pC, Word16 active) argument 200 InitTnsConfigurationShort(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_SHORT *pC, Word16 active) argument
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H A D | psy_main.c | 187 Word32 sampleRate, 197 sampleRate, 203 err = InitTnsConfigurationLong(bitRate, sampleRate, channels, 209 sampleRate, 213 err = InitTnsConfigurationShort(bitRate, sampleRate, channels, 252 Word32 sampleRate) 270 sampleRate, 186 psyMainInit(PSY_KERNEL *hPsy, Word32 sampleRate, Word32 bitRate, Word16 channels, Word16 tnsMask, Word16 bandwidth) argument 242 psyMain(Word16 nChannels, ELEMENT_INFO *elemInfo, Word16 *timeSignal, PSY_DATA psyData[MAX_CHANNELS], TNS_DATA tnsData[MAX_CHANNELS], PSY_CONFIGURATION_LONG *hPsyConfLong, PSY_CONFIGURATION_SHORT *hPsyConfShort, PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS], PSY_OUT_ELEMENT *psyOutElement, Word32 *pScratchTns, Word32 sampleRate) argument
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/frameworks/base/core/java/android/speech/tts/ |
H A D | BlockingAudioTrack.java | 76 BlockingAudioTrack(int streamType, int sampleRate, argument 80 mSampleRateInHz = sampleRate;
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/frameworks/av/media/libstagefright/wifi-display/source/ |
H A D | TSPacketizer.cpp | 295 int32_t sampleRate; local 296 CHECK(mFormat->findInt32("sample-rate", &sampleRate)); 297 CHECK(sampleRate == 44100 || sampleRate == 48000); 304 unsigned sampling_frequency = (sampleRate == 44100) ? 1 : 2;
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