/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerDyn.cpp | 167 int inChannelCount, int32_t sampleRate, src_quality quality) 168 : AudioResampler(inChannelCount, sampleRate, quality), 166 AudioResamplerDyn( int inChannelCount, int32_t sampleRate, src_quality quality) argument
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H A D | AudioResamplerSinc.cpp | 212 int inChannelCount, int32_t sampleRate, src_quality quality) 213 : AudioResampler(inChannelCount, sampleRate, quality), 211 AudioResamplerSinc( int inChannelCount, int32_t sampleRate, src_quality quality) argument
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H A D | AudioMixer.cpp | 101 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument 103 mSampleRate(sampleRate) 214 t->sampleRate = mSampleRate; 293 const uint32_t resetToSampleRate = track.sampleRate; 296 track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate. 297 // recreate the resampler with updated format, channels, saved sampleRate. 339 sampleRate, sessionId, kCopyBufferFrameCount); 690 track.sampleRate = mSampleRate; 760 if (sampleRate != trackSampleRate) { 761 sampleRate [all...] |
H A D | Tracks.cpp | 69 uint32_t sampleRate, 86 mSampleRate(sampleRate), 340 uint32_t sampleRate, 350 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 385 mFrameSize, !isExternalTrack(), sampleRate); 553 uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { function in class:android::AudioFlinger::PlaybackThread::Track 1129 uint32_t sampleRate, 1135 sampleRate, format, channelMask, frameCount, 1154 mClientProxy->setSampleRate(sampleRate); 1327 uint32_t sampleRate, 66 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int clientUid, IAudioFlinger::track_flags_t flags, bool isOut, alloc_type alloc, track_type type) argument 336 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, const sp<IMemory>& sharedBuffer, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 1126 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int uid) argument 1325 PatchTrack(PlaybackThread *playbackThread, audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) argument 1461 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 1661 PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) argument [all...] |
/frameworks/base/media/jni/soundpool/ |
H A D | SoundPool.h | 59 int sampleRate() { return mSampleRate; } function in class:android::Sample
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H A D | SoundPool.cpp | 654 uint32_t sampleRate; local 661 status = decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format, 670 ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d", 671 mHeap->getBase(), mSize, sampleRate, numChannels); 673 if (sampleRate > kMaxSampleRate) { 674 ALOGE("Sample rate (%u) out of range", sampleRate); 686 mSampleRate = sampleRate; 739 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5); local 750 uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRat 1024 uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5); local [all...] |
/frameworks/wilhelm/include/SLES/ |
H A D | OpenSLES_Android.h | 50 SLuint32 sampleRate; member in struct:SLAndroidDataFormat_PCM_EX_
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/frameworks/av/cmds/stagefright/ |
H A D | stagefright.cpp | 1002 long sampleRate = strtol(filename + 5, &end, 10); local 1005 sampleRate = 44100; 1007 mediaSource = new SineSource(sampleRate, 1);
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/frameworks/av/include/media/ |
H A D | MediaProfiles.h | 209 AudioCodec(audio_encoder codec, int bitRate, int sampleRate, int channels) argument 212 mSampleRate(sampleRate),
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/frameworks/av/include/private/media/ |
H A D | AudioTrackShared.h | 378 void setSampleRate(uint32_t sampleRate) { argument 379 mCblk->mSampleRate = sampleRate; 545 size_t frameSize, bool clientInServer = false, uint32_t sampleRate = 0) 549 mCblk->mSampleRate = sampleRate;
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/frameworks/av/media/libmedia/ |
H A D | AudioRecord.cpp | 38 uint32_t sampleRate, 47 status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size); 49 ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, " 50 "channelMask %#x; status %d", sampleRate, format, channelMask, status); 58 ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x", 59 sampleRate, format, channelMask); 77 uint32_t sampleRate, 100 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 135 uint32_t sampleRate, 150 ALOGV("set(): inputSource %d, sampleRate 36 getMinFrameCount( size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument 75 AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& opPackageName, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument 133 set( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, bool threadCanCallJava, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument [all...] |
H A D | AudioSystem.cpp | 276 *samplingRate = af->sampleRate(ioHandle); 360 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, argument 367 return afc->getInputBufferSize(sampleRate, format, channelMask, buffSize); 569 uint32_t sampleRate, audio_format_t format, 578 if ((mInBuffSize == 0) || (sampleRate != mInSamplingRate) || (format != mInFormat) 580 size_t inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask); 582 ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x", 583 sampleRate, format, channelMask); 588 mInSamplingRate = sampleRate; 568 getInputBufferSize( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t* buffSize) argument
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H A D | AudioTrack.cpp | 53 static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed) argument 55 return ((double)frames * 1000000000) / ((double)sampleRate * speed); 74 static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch) argument 76 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate; 93 uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/) 108 "sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/, 109 afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount 112 sampleRate, afFrameCount, afSampleRate, speed); 119 uint32_t sampleRate) 91 calculateMinFrameCount( uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, uint32_t sampleRate, float speed ) argument 116 getMinFrameCount( size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) argument 187 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, int32_t notificationFrames, audio_session_t sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect, float maxRequiredSpeed) argument 218 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const sp<IMemory>& sharedBuffer, audio_output_flags_t flags, callback_t cbf, void* user, int32_t notificationFrames, audio_session_t sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect, float maxRequiredSpeed) argument 277 set( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, int32_t notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, audio_session_t sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect, float maxRequiredSpeed) argument 838 uint32_t sampleRate = 0; local 1862 uint32_t sampleRate = mSampleRate; local 2260 isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const argument [all...] |
H A D | IAudioFlinger.cpp | 100 uint32_t sampleRate, 117 data.writeInt32(sampleRate); 178 uint32_t sampleRate, 197 data.writeInt32(sampleRate); 273 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const function in class:android::BpAudioFlinger 440 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, argument 445 data.writeInt32(sampleRate); 946 uint32_t sampleRate = data.readInt32(); local 969 (audio_stream_type_t) streamType, sampleRate, format, 984 uint32_t sampleRate local 98 createTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t pid, pid_t tid, audio_session_t *sessionId, int clientUid, status_t *status) argument 176 openRecord( audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& opPackageName, size_t *pFrameCount, track_flags_t *flags, pid_t pid, pid_t tid, int clientUid, audio_session_t *sessionId, size_t *notificationFrames, sp<IMemory>& cblk, sp<IMemory>& buffers, status_t *status) argument 1124 uint32_t sampleRate = data.readInt32(); local [all...] |
/frameworks/av/media/libstagefright/ |
H A D | AVIExtractor.cpp | 339 int sampleRate; local 342 header, &frameSize, &sampleRate, NULL, NULL, &numSamples)) { 353 int64_t timeUs = mBaseTimeUs + (mNumSamplesRead * 1000000ll) / sampleRate; 713 uint32_t sampleRate = U32LE_AT(&data[4]); local 716 track->mMeta->setInt32(kKeySampleRate, sampleRate);
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H A D | Utils.cpp | 685 int32_t numChannels, sampleRate; local 687 || !meta->findInt32(kKeySampleRate, &sampleRate)) { 692 msg->setInt32("sample-rate", sampleRate); 1324 int32_t sampleRate; local 1325 if (msg->findInt32("sample-rate", &sampleRate)) { 1326 meta->setInt32(kKeySampleRate, sampleRate); 1448 int32_t sampleRate = 0; local 1456 if (meta->findInt32(kKeySampleRate, &sampleRate)) { 1457 param.addInt(String8(AUDIO_OFFLOAD_CODEC_SAMPLE_RATE), sampleRate); local 1472 ALOGV("sendMetaDataToHal: bitRate %d, sampleRate [all...] |
/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | tns.c | 136 Word32 sampleRate, /*!< Sampling frequency */ 164 tC->tnsStartBand = FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 168 sampleRate, 173 sampleRate, 203 Word32 sampleRate, /*!< Sampling frequency */ 230 tC->tnsStartBand=FreqToBandWithRounding(tC->tnsStartFreq, sampleRate, 234 sampleRate, 239 sampleRate, 135 InitTnsConfigurationLong(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_LONG *pC, Word16 active) argument 202 InitTnsConfigurationShort(Word32 bitRate, Word32 sampleRate, Word16 channels, TNS_CONFIG *tC, PSY_CONFIGURATION_SHORT *pC, Word16 active) argument
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/frameworks/av/media/libstagefright/mpeg2ts/ |
H A D | ESQueue.cpp | 694 int32_t sampleRate; local 696 if (!mFormat->findInt32(kKeySampleRate, &sampleRate)) { 706 sampleRate, numChannels);
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/frameworks/base/media/java/android/media/ |
H A D | AudioFormat.java | 44 * <li><a href="#sampleRate">sample rate</a> 52 * <h4 id="sampleRate">Sample rate</h4> 591 private AudioFormat(int encoding, int sampleRate, int channelMask, int channelIndexMask) { argument 593 mSampleRate = sampleRate; 887 * @param sampleRate the sample rate expressed in Hz 891 public Builder setSampleRate(int sampleRate) throws IllegalArgumentException { argument 895 if (((sampleRate < SAMPLE_RATE_HZ_MIN) || (sampleRate > SAMPLE_RATE_HZ_MAX)) && 896 sampleRate != SAMPLE_RATE_UNSPECIFIED) { 897 throw new IllegalArgumentException("Invalid sample rate " + sampleRate); [all...] |
H A D | MediaFormat.java | 890 * @param sampleRate The sampling rate of the content. 895 int sampleRate, 899 format.setInteger(KEY_SAMPLE_RATE, sampleRate); 893 createAudioFormat( String mime, int sampleRate, int channelCount) argument
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/frameworks/opt/net/voip/src/jni/rtp/ |
H A D | AudioGroup.cpp | 102 AudioCodec *codec, int sampleRate, int sampleCount, 106 bool mix(int32_t *output, int head, int tail, int sampleRate); 168 AudioCodec *codec, int sampleRate, int sampleCount, 180 mSampleRate = sampleRate / 1000; 236 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate) argument 255 if (sampleRate == mSampleRate) { 478 bool set(int sampleRate, int sampleCount); 577 bool AudioGroup::set(int sampleRate, int sampleCount) argument 585 mSampleRate = sampleRate; 599 sampleRate, sampleCoun 167 set(int mode, int socket, sockaddr_storage *remote, AudioCodec *codec, int sampleRate, int sampleCount, int codecType, int dtmfType) argument 788 int sampleRate = mGroup->mSampleRate; local 978 int sampleRate = -1; local [all...] |
/frameworks/wilhelm/src/android/ |
H A D | AudioPlayer_to_android.cpp | 1471 uint32_t sampleRate = sles_to_android_sampleRate(df_pcm->samplesPerSec); local 1506 sampleRate, // sampleRate
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/frameworks/av/media/libmediaplayerservice/ |
H A D | StagefrightRecorder.cpp | 352 status_t StagefrightRecorder::setParamAudioSamplingRate(int32_t sampleRate) { argument 353 ALOGV("setParamAudioSamplingRate: %d", sampleRate); 354 if (sampleRate <= 0) { 355 ALOGE("Invalid audio sampling rate: %d", sampleRate); 360 mSampleRate = sampleRate;
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/frameworks/av/media/libmediaplayerservice/nuplayer/ |
H A D | NuPlayerRenderer.cpp | 1030 int32_t sampleRate = offloadingAudio() ? local 1032 if (sampleRate == 0) { 1033 ALOGE("sampleRate is 0 in %s mode", offloadingAudio() ? "offload" : "non-offload"); 1037 return (int64_t)((int32_t)numFrames * 1000000LL / sampleRate); 1764 int32_t sampleRate; local 1765 CHECK(format->findInt32("sample-rate", &sampleRate)); 1797 offloadInfo.sample_rate = sampleRate; 1820 sampleRate, 1872 sampleRate 1893 (unsigned long long)sampleRate * getAudioSinkPcmMsSettin [all...] |
/frameworks/av/media/libstagefright/httplive/ |
H A D | PlaylistFetcher.cpp | 1991 int32_t sampleRate; local 1992 CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); 2036 int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate;
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