/frameworks/av/media/libmediaplayerservice/nuplayer/ |
H A D | NuPlayer.cpp | 375 int32_t sampleRate; local 376 CHECK(codecRequest->findInt32("sample-rate", &sampleRate)); 379 sampleRate, numChannels); 403 sampleRate,
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/frameworks/av/include/media/ |
H A D | MediaProfiles.h | 242 AudioCodec(audio_encoder codec, int bitRate, int sampleRate, int channels) argument 245 mSampleRate(sampleRate),
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H A D | AudioSystem.h | 115 static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
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H A D | MediaPlayerInterface.h | 97 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
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/frameworks/av/include/media/stagefright/ |
H A D | OMXCodec.h | 248 int32_t numChannels, int32_t sampleRate, int32_t bitRate, 290 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | block_switch.c | 111 Word32 sampleRate, 138 if(sampleRate >= 16000) { 109 BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, Word16 *timeSignal, Word32 sampleRate, Word16 chIncrement) argument
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/frameworks/av/services/audioflinger/ |
H A D | AudioFlinger.cpp | 444 uint32_t sampleRate, 505 track = thread->createTrack_l(client, streamType, sampleRate, format, 547 uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const function in class:android::AudioFlinger 552 ALOGW("sampleRate() unknown thread %d", output); 555 return thread->sampleRate(); 978 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, argument 989 sample_rate: sampleRate, 1712 uint32_t sampleRate, 1752 (sampleRate == mSampleRate) && 1769 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate 441 createTrack( pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, IAudioFlinger::track_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, int *sessionId, status_t *status) argument 1709 createTrack_l( const sp<AudioFlinger::Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, status_t *status) argument 3625 unsigned sampleRate = Format_sampleRate(format); local 4140 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument 4284 int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { function in class:android::AudioFlinger::ThreadBase::TrackBase 4314 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags) argument 4854 create( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument 4872 TimedTrack( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, int sessionId) argument 5368 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, int sessionId) argument 5490 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount) argument 5878 openRecord( pid_t pid, audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, IAudioFlinger::track_flags_t flags, pid_t tid, int *sessionId, status_t *status) argument 5986 RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t id, audio_devices_t device) argument 6241 createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, status_t *status) argument [all...] |
H A D | AudioMixer.cpp | 99 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument 101 mSampleRate(sampleRate) 202 t->sampleRate = mSampleRate; 285 pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate; 286 pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate; 475 track.sampleRate = mSampleRate; 539 if (sampleRate != value) { 540 sampleRate = value; 765 t->resampler->setSampleRate(t->sampleRate); 1426 return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate); [all...] |
H A D | AudioResamplerSinc.cpp | 203 int inChannelCount, int32_t sampleRate, src_quality quality) 204 : AudioResampler(bitDepth, inChannelCount, sampleRate, quality), 202 AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) argument
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/frameworks/av/media/libstagefright/ |
H A D | OMXCodec.cpp | 507 int32_t numChannels, sampleRate, aacProfile; local 509 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 520 status_t err = setAACFormat(numChannels, sampleRate, bitRate, aacProfile, isADTS); 537 int32_t numChannels, sampleRate; local 539 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 541 setRawAudioFormat(kPortIndexInput, sampleRate, numChannels); 3273 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) { 3300 pcmParams.nSamplingRate = sampleRate; 3378 int32_t sampleRate; local 3380 CHECK(format->findInt32(kKeySampleRate, &sampleRate)); 3272 setRawAudioFormat( OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) argument 3387 setAACFormat( int32_t numChannels, int32_t sampleRate, int32_t bitRate, int32_t aacProfile, bool isADTS) argument 4355 int32_t numChannels, sampleRate; local 4414 int32_t numChannels, sampleRate, bitRate; local [all...] |
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 | ACodec.cpp | 926 int32_t numChannels, sampleRate; local 928 || !msg->findInt32("sample-rate", &sampleRate)) { 940 encoder, numChannels, sampleRate, bitRate, aacProfile, isADTS != 0); 958 int32_t numChannels, sampleRate, compressionLevel = -1; local 961 || !msg->findInt32("sample-rate", &sampleRate))) { 976 err = setupFlacCodec(encoder, numChannels, sampleRate, compressionLevel); 979 int32_t numChannels, sampleRate; local 982 || !msg->findInt32("sample-rate", &sampleRate)) { 985 err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels); 1077 bool encoder, int32_t numChannels, int32_t sampleRate, 1076 setupAACCodec( bool encoder, int32_t numChannels, int32_t sampleRate, int32_t bitRate, int32_t aacProfile, bool isADTS) argument 1259 setupFlacCodec( bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel) argument 1287 setupRawAudioFormat( OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels) argument [all...] |
H A D | MPEG4Extractor.cpp | 1849 int32_t sampleRate = 0; local 1855 sampleRate = br.getBits(24); 1867 sampleRate = kSamplingRate[freqIndex]; 1878 if (prevSampleRate != sampleRate) { 1880 "was: %d, now: %d", prevSampleRate, sampleRate); 1883 mLastTrack->meta->setInt32(kKeySampleRate, sampleRate);
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/frameworks/av/media/libmedia/ |
H A D | AudioSystem.cpp | 242 *samplingRate = af->sampleRate(output); 337 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, argument 343 if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat) 350 inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask); 353 gPrevInSamplingRate = sampleRate;
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H A D | JetPlayer.cpp | 93 pLibConfig->sampleRate,
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/frameworks/av/cmds/stagefright/ |
H A D | SimplePlayer.cpp | 581 int32_t sampleRate; local 583 CHECK(format->findInt32("sample-rate", &sampleRate)); 587 sampleRate,
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H A D | sf2.cpp | 284 int32_t numChannels, sampleRate; local 286 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 289 msg->setInt32("sample-rate", sampleRate);
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H A D | stagefright.cpp | 988 long sampleRate = strtol(filename + 5, &end, 10); local 991 sampleRate = 44100; 993 mediaSource = new SineSource(sampleRate, 1);
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/frameworks/av/libvideoeditor/lvpp/ |
H A D | VideoEditorPlayer.h | 53 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
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/frameworks/av/libvideoeditor/vss/stagefrightshells/src/ |
H A D | VideoEditorAudioDecoder.cpp | 740 int32_t sampleRate, channelCount; local 742 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 744 ALOGV("VideoEditorAudioDecoder_step: samplingFreq = %d", sampleRate); 747 (uint32_t)sampleRate;
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/frameworks/av/media/libmediaplayerservice/ |
H A D | StagefrightRecorder.h | 159 status_t setParamAudioSamplingRate(int32_t sampleRate);
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H A D | StagefrightRecorder.cpp | 316 status_t StagefrightRecorder::setParamAudioSamplingRate(int32_t sampleRate) { argument 317 ALOGV("setParamAudioSamplingRate: %d", sampleRate); 318 if (sampleRate <= 0) { 319 ALOGE("Invalid audio sampling rate: %d", sampleRate); 324 mSampleRate = sampleRate;
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | ARTPWriter.cpp | 475 int32_t sampleRate, numChannels; local 476 CHECK(mSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); 480 CHECK_EQ(sampleRate, (mMode == AMR_NB) ? 8000 : 16000); 483 sdp.append(StringPrintf("/%d/%d", sampleRate, numChannels));
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/frameworks/base/media/java/android/media/ |
H A D | AudioRecord.java | 802 int recordSource, int sampleRate, int nbChannels, int audioFormat, 801 native_setup(Object audiorecord_this, int recordSource, int sampleRate, int nbChannels, int audioFormat, int buffSizeInBytes, int[] sessionId) argument
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/frameworks/av/media/libstagefright/mpeg2ts/ |
H A D | ESQueue.cpp | 423 int32_t sampleRate; local 425 CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate)); 429 sampleRate, numChannels);
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