/frameworks/av/include/media/ |
H A D | AudioTrack.h | 145 uint32_t sampleRate); 170 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 225 uint32_t sampleRate, 255 uint32_t sampleRate, 285 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 299 uint32_t sampleRate, 430 status_t setSampleRate(uint32_t sampleRate); 927 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const;
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H A D | AudioRecord.h | 119 uint32_t sampleRate, 148 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 172 uint32_t sampleRate, 200 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 210 uint32_t sampleRate,
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/frameworks/av/services/audioflinger/ |
H A D | BufferProviders.cpp | 139 uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) : 147 sampleRate, sessionId); 162 mDownmixConfig.inputCfg.samplingRate = sampleRate; 163 mDownmixConfig.outputCfg.samplingRate = sampleRate; 325 audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) : 328 mSampleRate(sampleRate), 333 mSonicStream(sonicCreateStream(sampleRate, mChannelCount)), 342 this, channelCount, format, sampleRate, playbackRate.mSpeed, 136 DownmixerBufferProvider( audio_channel_mask_t inputChannelMask, audio_channel_mask_t outputChannelMask, audio_format_t format, uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) argument 324 TimestretchBufferProvider(int32_t channelCount, audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) argument
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H A D | BufferProviders.h | 95 uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount); 154 audio_format_t format, uint32_t sampleRate,
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H A D | TrackBase.h | 59 uint32_t sampleRate, 107 virtual uint32_t sampleRate() const { return mSampleRate; } function in class:TrackBase
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H A D | AudioResamplerDyn.h | 45 int32_t sampleRate, src_quality quality);
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H A D | AudioResamplerSinc.h | 37 AudioResamplerSinc(int inChannelCount, int32_t sampleRate,
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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/av/include/media/stagefright/ |
H A D | AudioSource.h | 39 uint32_t sampleRate,
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H A D | ACodec.h | 456 int32_t numChannels, int32_t sampleRate, int32_t bitRate, 461 status_t setupAC3Codec(bool encoder, int32_t numChannels, int32_t sampleRate); 463 status_t setupEAC3Codec(bool encoder, int32_t numChannels, int32_t sampleRate); 469 status_t setupG711Codec(bool encoder, int32_t sampleRate, int32_t numChannels); 472 bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel); 475 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels,
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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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/frameworks/av/media/libstagefright/ |
H A D | AMRWriter.cpp | 80 int32_t sampleRate; local 83 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 84 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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H A D | AudioPlayer.cpp | 620 uint32_t sampleRate; local 623 sampleRate = mAudioSink->getSampleRate(); 626 sampleRate = mAudioTrack->getSampleRate(); 628 if (sampleRate != 0) { 629 mSampleRate = sampleRate;
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/frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
H A D | AudioSession.cpp | 32 uint32_t sampleRate, 40 mConfig({ .format = format, .sample_rate = sampleRate, .channel_mask = channelMask}), 104 other->sampleRate() == mConfig.sample_rate && 29 AudioSession(audio_session_t session, audio_source_t inputSource, audio_format_t format, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_input_flags_t flags, uid_t uid, bool isSoundTrigger, AudioMix* policyMix, AudioPolicyClientInterface *clientInterface) argument
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/frameworks/av/media/libmedia/ |
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 | 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...] |
/frameworks/wilhelm/tests/examples/ |
H A D | slesTestRecBuffQueue.cpp | 42 uint32_t sampleRate = 48000; variable 164 info.samplerate = sampleRate; 250 pcm.sampleRate = sampleRate * 1000; // milliHz 435 sampleRate = atoi(&arg[2]);
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H A D | slesTestFeedback.cpp | 41 static SLuint32 sampleRate = 48000; // -s# variable 296 sampleRate = atoi(&arg[2]); 297 switch (sampleRate) { 310 (unsigned) sampleRate); 373 info.samplerate = sampleRate; 419 pcm.samplesPerSec = sampleRate * 1000;
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/frameworks/av/include/media/nbaio/ |
H A D | NBAIO.h | 72 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount, audio_format_t format);
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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/libstagefright/codecs/aacdec/ |
H A D | SoftAAC2.cpp | 252 aacParams->nSampleRate = mStreamInfo->sampleRate; 289 pcmParams->nSamplingRate = mStreamInfo->sampleRate; 600 if (mStreamInfo->sampleRate && mStreamInfo->numChannels) { 602 mStreamInfo->sampleRate, 691 INT prevSampleRate = mStreamInfo->sampleRate; 796 if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) { 803 } else if ((mStreamInfo->sampleRate != prevSampleRate) || 806 prevSampleRate, mStreamInfo->sampleRate, 1095 mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
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/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/base/core/java/android/speech/tts/ |
H A D | BlockingAudioTrack.java | 78 BlockingAudioTrack(AudioOutputParams audioParams, int sampleRate, argument 81 mSampleRateInHz = sampleRate;
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/frameworks/base/media/java/android/media/ |
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/base/media/java/android/media/audiopolicy/ |
H A D | AudioPolicyConfig.java | 115 int sampleRate = in.readInt(); 118 final AudioFormat format = new AudioFormat.Builder().setSampleRate(sampleRate)
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