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* Revert "libstagefright: Fix incorrectness nPortIndex value for QueryCodec"Ricardo Cerqueira2013-06-201-1/+1
| | | | | | | | This reverts commit 9a814ad626233ff02dd2d393929f32225bc94b68. This is wrong. kPortIndexInput is defined as 0, the original value was correct. Additionally, it breaks android.media.cts.MediaCodecListTest Change-Id: Ib273cde69a4c622daf239bab5d12c5e7d568af2f
* Squashed commit of A/V changes from CodeAuroraKrishnankutty Kolathappilly2013-06-1843-213/+1859
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * Brings us current with AU_LINUX_ANDROID_JB_2.5.04.02.02.040.367 Camera: Fix deadlock due to mLock in pcb and takepicture In non-zsl case of takepicture, we do streamoff for preview stream which is waiting on preview callback thread to exit. By that time the lock has already been acquired by takePicture. So preivew callback will not exit until it acquires lock and takePicture cannot continue until PCB call back is returned. Fix: Avoid the mLock at services when both Preview cb & Compressed cb are enabled. Change-Id: I6c264928bf1540c7b51f1add65f9c3e968506e15 CRs-fixed: 479419 audioflinger: Fix the LPA-AudioEffects crash issue - Issue:crash is observed during LPA playback on enabling effects followed by plug-out->plug-in of wired headset - Rootcause: while deleteing the effectchain in deleteEffect EffctChain is being unlocked after clearing the chain which leads to accessing the lock which might already deleted. - Fix: first unlock the effectChain and then call clear CRs-Fixed: 491774 Change-Id: I518ff086c5ad71486cd29142563145137ebc15b6 libstagefright: Fix for crash in sound recorder during device switch -Crash seen in sound recorder during frequent insertion and removal of wired headset -During device switch some time Codec's input buffers are too small to accomodate buffer read from source. Omx codec doesn't read the fix size buffer from source, during device switch scenario sometime buffer read from source exceeds input buffer size so it goes in error state which leads to crash. -Increasing the input buffer size fix this issue Change-Id: Id15378670880d0c3c0bd4408841b28be963549a0 CRs-Fixed: 488449 libstagefright: Fix for FPS drop issue during A-V playback. Issues: -The AAC decoder was not updating the timestamp when EOS is reached. -Logic to smoothen the real time update in AudioPlayer uses system time. This introduces corrupt timestamp during EOS. Fix: -Update the timestamp in AAC decoder when EOS is reached. -Extrapolate realtime using system time in AudioPlayer when EOS is reached. Cap the value to realtime if extrapolated time becomes greater than realtime. CRs-Fixed: 384183 Change-Id: Ice54501436431d2527fcd3d710d65d9732fcffdd libstagefright: Reset buffer size value with SurfaceTexture - OMXCodec explicitly sets the decoder output buffer size using the native window perform API. (to accomodate extra-data) - This size is reset only when the SurfaceTexture is destroyed. - Unless reset, this size will be assumed for all output buffers if the SurfaceTexture is re-used. CRs-Fixed: 337660, 432309 Change-Id: I28aed12ad02adeac61caffbb00e3082640a5f6d4 audio: Add support for tunnel mode recording - Add support for tunnel mode recording. Change-Id: I95cdfff729affd784141487521c9f2f714221d11 audio: Add support for non-pcm VOIP vocoders - Add support for non-pcm VOIP vocoders - non-pcm vocoders use AUDIO_SOURCE_VOICE_COMMUNICATION as inputSource. Add check to verify inputSource and then configure framecount accordingly Change-Id: Ia38da4f6ba0ee40c794d3c97325327cdb7dcb32a CRs-Fixed: 467850 frameworks/av: Add metadata mode changes to LPAPlayer -Seek to EOS was causing playback to hang for 3 seconds before switching to the next clip. -This is because the lpa driver works on period size. Partial buffers are not handled. -Add support for metadata mode changes to LPAPlayer to support partial frames. CRs-Fixed: 458904 Change-Id: I8673756b54ae7bca18855d326c85ae1064652514 libstagefright: Add support for WMA in ACodec - WMA support is not there in ACodec - In the case of wma format, since not getting the complete information of wma version so instead of allocating the component in onAllocateComponent function it will create in onConfigureCompoenent function. bitspersample is find as "bsps" from AMessage while configuring the WMA10PRO and WMALOSSLESS format CRs-Fixed: 453951 Change-Id: I98baa701dbf8a5c012f4be5e83831c0be2111dcc libstagefright: Flush the pending buffers when EOS is received For the use case where the first frame in the buffer is EOS, decode the aac config frame buffer to update the sample rate and channel mode and flush out the buffer. Change-Id: I0354802cdbf61ac1ba0fecbbdf616705806b0f4a CRs-Fixed: 459334 audio: Fix The Linux Foundation copyright - Fix copyright format based on The Linux Foundation copyright template Change-Id: I100a5c86302d1a1a3d79543d95e242734daae746 media, audioflinger: check for divide by zero possibilities and err When output stream is not available to audioflinger due to any reason , sampleRate and frameCount have zero values when trying to create new Audiotrack. This might result in divide by 0 situation. Change-Id: Ic13cb51facb8497e68ab596abb027b44f496b907 CRs-Fixed: 478480 framewroks/av:Fix ANR at the end of video recording - While doing video recording, when the recording ends ANR observed while doing stress test for many hours - When the recording is stopped, audio HAL receives error from driver and audio HAL propagates this error to AudioFlinger. But AudioFlinger is not sending error status to audio source to stop recording. Because of this audiorecord thread keeps on waiting for buffers which is resulting in ANR. - To avoid indefinite wait, a timeout of 1 sec is set for buffer in audioSource and after timeout, -ETIMEDOUT is returned to recorder thread. CRs-Fixed: 479968 Change-Id: I91aba6922086e711992d9d991dea9c35d33eaee9 audioflinger: Integrate SRS TruMedia Change-Id: If61ae91556120ddd5f5ebcc6dbbfe6583c7df67d audioflinger: Fix apply SRS effects if tones diabled in tunnel mode For the use case of SRS post processing in Tunnel mode, the API's of SRS are called only from write. With the huge buffering for tunnel mode, once EOS is received there would not be further write. With system tone enabled, the SRS API's are called during the check for Parameters change through normal mixer thread. With system tones disabled, SRS will not be applied after EOS as no write and mixer thread would not be active. Fix the issue by adding the Effects Thread for SRS in Tunnel mode. Fix the compilation issue with ALOGV messages enabled Change-Id: Ic7e62894840f786119dfe8ae471c5d24812917d7 audioflinger: Enhance LPA-effect logic to handle rapid config. -Issue:Rapid Config events cause pops/glitches, raw data playback. -Rootcause1:Raw data leakage to DSP: applyEffectsOn() applies effects chunk by chunk in a loop, if effects change during this time the loop exits and this results in creation of a buffer in which part of it is effects processed and rest raw, this causes raw data to leak to DSP. -RootCause2:Effectsthread directly works on the DSP buffers, while DSP is rendering from there, so that effect application is instantaneous and for this it gives the DSP buffers as output to effects chain, this means that all the effects in the chain update the DSP buffers one after the other, this can create unpredictable rendering patterns. RootCause1 and 2 combined seem to fragment memory with parts of it with effects and parts with raw data etc. -Fix1:Dont update DSP mem unless the effects are applied completely on a buffer. -Fix2:Effectschain will work on a temp scrath buffer instead of DSP mem and when effects are applied completely on this scrath buffer, memcpy this to DSP mem with this DSP mem is updated in one shot. -Remove repetetive logs which clutter the logcat if msgs are enabled in audioflinger. Change-Id: I9051e7b8531aa5c8cb3dcfafe0be3136a2cf0f9d CRs-Fixed: 463880 frameworks/av: Update framecount and buffersize values -framecount should be calculated based on mMaxBufferSize returned from HAL -update the buffersize with the value returned from HAL CRs-Fixed: 482744 Change-Id: I90dd9c3ebbbc8a9f1f2f92c5347ae9cb01719e13 audioflinger: Fix the LPA-AudioEffects dead lock issue. - Issue:Deadlock occurs when the LPA clips are subjected to rapid next from BT device and simultaneously on/off the audio effects. - Rootcause:some times flinger thread processing LPAPlayer/directtrack next deadlocks with the thread working on effect configuration as both of them contend for the audioflinger::mlock and effectmodule::mlock. - Fix1:AudioFlinger::deleteEffectSession() not to acquire audioflinger:mLock instead take the mLPAEffectChain.mlock. - Fix2:ThreadBase::effectConfigChanged() not to acquire audioflinger::mlock. Change-Id: I056c8297802f81644fa1371836db42bdbd3825fd CRs-Fixed: 477511 libstagefright: Add support for High Frame Rate Encoding - Based on kkeyhfr key value from meta data, add support in OMXCodec and MPEG4Writer for HFR mode - Assume normal mode recording if kKeyHfr is absent - Increase bit rate for high frame rate (HFR) recording feature to reflect the corresponding increase in frame rate Change-Id: I0a69f8d9322a768677781d08dd910dc5772c5292 libstagefright: Support some userdefine properties - support property to disable audio - support property to change recorder profile mode - support b frame encoding Change-Id: I175decec83f6027cbd7988caf680f7fec2836f83 CRs-Fixed: 443327 libstagefright: Add support for H/W AAC decoder - Currently, only software AAC decoding is supported. - Add support for H/W AAC decoding by including it in the list of available decoders and use it for decoding only if the property 'media.aaccodectype' is set to 0. Change-Id: I4bb9df1bd10bd8ee91e63dadd6c473fc4e29813a CRs-Fixed: 449145 libstagefright: Move checks for creating new extractor to ExtendedExtractor - Move all the checks and creation of the extended extractor into ExtendedExtractor. - Restrict creation of new extractor to the following conditions o default extractor is NULL o default extractor says the content is video only or has an unrecognized audio stream o the audio stream is a amr-wb (plus). - This change is being added to avoid unnecessary creation of two extractors thereby improving the startup latency. CRs-Fixed: 462087 Change-Id: Ia87eca73c4f81d37697fa85fd4f7c8cc8d406104 [StageFright] Enable 4 channel support This patches enables 4 channel WAV audio support and fixes invalid data size in WAV header field if it exceeds the actual source size. This patch is needed to support WebAudio in WebKit as some of the chrome demos use 4 channel WAV audio and bogus header information. Change-Id: I307026107ab4e4342b1c0d7bb64761a416fb2c65 audioflinger: Fix crash on LPA shutdown * Decrement the refcount after unlocking the mutex Change-Id: Ic3210700e0aaf5e8df78f85f501621a455058e24 libstagefright: Accept vendor specific NV12 colorformat from component - Accept OMX_QCOM_COLOR_FormatYUV420PackedSemiPlanar32m color format which is NV12 + 32 aligned stride and slice. - This is different from vanilla NV12 which is 16 aligned. Change-Id: I6de2ec3a78215dbcc28a6006b746e3e0afe69c3c libstagefright: various fixes for avc_utils - skip seq_scaling_matrix_present_flag assertion if checking for interlaced property. - correct interlace check to outside of if-block Change-Id: Ia5854110feb1c56ddc86b312d2ba2dbb73d37804 CRs-Fixed: 445527, 445692 libstagefright: print stats at end of playback - prints statistics before reset at the end of playback onto logcat - print statistics after each pause and seek Change-Id: I68edcc3153a04209e7382e4d3fba0bf734f3e33f CRs-Fixed: 457926, 447109 frameworks/base : Fix to play a specific Mp4 clip due to SYNCH_LOST_ERROR. -Unable to play a Specific Mp4 clip. -Mp3 playback is stopped if the Decoder errors out with SYNCH_LOST_ERROR. -Ignore the frame with SYNCH_LOST_ERROR and play silence instead. Change-Id: I6b94a83cf89e8bc6792d8ee3804042d629aa505b Add checks before removing an active buffer in OMXNodeInstance With this change, OMXNodeInstance will remove a buffer from it's active list only if OMX_FreeBuffer returns successfully. Change-Id: I685b39ac7ba762a2fc1b64d7f6c1efd391513598 libstagefright: Add interlaced video support - Adds call to set output buffer size on the native window Change-Id: If4a67b3f877bef557c46bb67b29d1e7051553335 audio: fix for AMRWB param overwritten issue - Overwrite AMRWB params with default value only when setParameters is not invoked CRs-Fixed: 456459 Change-Id: I3fa6b56101ca408ed5b5b82707c6dc75a9d9f17b audio: fix encoder parameters for AMRWB format - AMRWB encoder only accepts SampleRate 16k and channel count 1. Always overwrite AMRWB SampleRate and channel count to default values. - AMRWB encoder accepts BitRate from 6.6k to 23.85k, only overwrite AMRWB BitRate to default(23.85k)if setParameters() is not invoked Change-Id: I75a96b54ef04bc59dab9074ec112071e62fd51aa CRs-Fixed: 460931 stagefright: Add QCOM_BSP ifdefs for interlaced video handling Change-Id: I856ae4a97f1bf13ab18d386b3486e742a4804b2a Camera : Changes to support camcorder profiles. Change-Id: I9c4bf14f273839fd36d5f52db0f215873e8291a0 av: Ifdef all the things! Change-Id: If9dd6c6442e9d2ac9e55e48369f2da85f5f951f7 Camera: Add profiles for camcorder. Change-Id: Icdaf1fae0018de1fb04f41125cfbe34a91b5eda7 libvideoeditor: use vWidth and vHeight for buffer allocation - video editor detects crop information from decoder, crop width and height will override metadata width and height. - decoder is capable of sending crop information where crop width and height are smaller than actual resolution. - use actual metadata width and height for calculating buffer size. Change-Id: Id1d77c316e3892e6d51a00418052f256629f495f CRs-Fixed: 452511 Add ifdefs around enhanced media types Change-Id: I64b8853660ac4fe90ddb218b237f63b635cdb47b
* Merge "libstagefright: support for disabling buffer metadata" into cm-10.1Ricardo Cerqueira2013-06-111-1/+10
|\ | | | | | | Change-Id: I6a7a91d930f7789ca78370f0c0e0e306dad87028
| * libstagefright: support for disabling buffer metadataHaynes Mathew George2013-06-111-1/+9
| | | | | | | | | | | | | | - Metada mode video recording is enabled by default. - use setprop debug.camcorder.disablemeta 1 to disable metadata mode recording. Change-Id: I422c49c0ace0c3a3e1f4459c7e4bf29e70af763a
* | Noise will be heard if audio sample rate not matched with audio trackMing Zhou2013-06-101-1/+1
|/ | | | | | | | | | | | When audio sample rate which set to audio track is not the same with the actual pcm data, noise will be heard. Fix the bug when write 8 bit pcm samples. AOSP commit: https://android-review.googlesource.com/#/c/59837/ Change-Id: Idcb0d7b0e9aaa250dd22b758c8337e23d1706049 Signed-off-by: Ming Zhou <b42586@freescale.com> Signed-off-by: guoyin.chen <guoyin.chen@freescale.com>
* Revert "frameworks/av: Add metadata mode changes to LPAPlayer"Giulio Cervera2013-06-011-2/+3
| | | | This reverts commit ae57fbc021cfc8b018cfb23b90112b1b17173d1b.
* frameworks/av: Add metadata mode changes to LPAPlayerKrishnankutty Kolathappilly2013-05-221-3/+2
| | | | | | | | | | | | -Seek to EOS was causing playback to hang for 3 seconds before switching to the next clip. -This is because the lpa driver works on period size. Partial buffers are not handled. -Add support for metadata mode changes to LPAPlayer to support partial frames. CRs-Fixed: 458904 Change-Id: I8673756b54ae7bca18855d326c85ae1064652514
* camera: fixup htc paramsDaniel Hillenbrand2013-05-052-5/+15
| | | | Change-Id: Iee0086e8a7319ada09f5191092c314c099e0d66a
* [WIP]: camera: add parameters for htc camerasDaniel Hillenbrand2013-05-042-0/+71
| | | | Change-Id: I1456e5af22d1cb68fd19626a136fef68c5573074
* camera: Don't segfault if we get a NULL parameterSteve Kondik2013-05-011-0/+3
| | | | | | * Values end up NULL on some drivers, don't crash. Change-Id: Ic897dbd4629cf3af98c85f93be202c382dde806b
* camera: Add a few missing legacy camera params.Matt Filetto2013-04-281-0/+4
| | | | | | Fixes camera for some HTC msm7x30 devices Change-Id: If6975e3a5470fdeb603bf8ad9c164c9a000309ef
* Add camera parameters for htc evo 3D. (1/2)Kevin Bruckert2013-04-287-2/+63
| | | | | | Use BOARD_HTC_3D_SUPPORT to enable. Change-Id: I28fa3f1586071bcc78b8e887bbbf699d338a0ceb
* libmedia: Add ICS AudioStreamType constructorArne Coucheron2013-04-281-0/+7
| | | | | | Needed for Samsung legacy camera libs. Change-Id: If03d8525b55181ea20dc934dbcbfef85402c42c7
* OMXCodec: Re-implement requires-flush-before-shutdown quirkPawit Pornkitprasan2013-04-241-0/+4
| | | | | | | Support is already there, but is not in the codec quirk reading list. Re-implement it as required by Broadcom's OMX Change-Id: I1beac06af8118dcf0c248b631bc8e6dbbab2c1d5
* Revert "audioflinger: apply volume on direct track when track is active"detule2013-04-211-7/+5
| | | | | | | | | | | | | | | | Causes issues with MortPlayer (LPA). Track starts/resumes playing, but volume is forced to zero, until a track-change occurs. 4-17 21:41:51.492 D/ALSADevice( 256): setHardwareParams: reqBuffSize 262144 channels 2 sampleRate 44100 04-17 21:41:51.492 D/ALSADevice( 256): setHardwareParams: buffer_size 1048576, period_size 262144, period_cnt 4 04-17 21:41:51.492 W/AudioFlinger( 256): There was no effectChain created for the sessionId(283) 04-17 21:41:51.512 E/AudioFlinger( 256): setting observer mOutputDesc track 0x40c598a0, obv 0x40c598b0 04-17 21:41:51.512 D/AudioSessionOutALSA( 256): setLpaVolume(0) 04-17 21:41:51.512 D/AudioSessionOutALSA( 256): Setting LPA volume to 0 (available range is 0 to 100) 04-17 21:41:51.512 D/ALSADevice( 256): setMixerControl:: name LPA RX Volume value 0 index 0 04-17 21:41:51.512 D/AudioSessionOutALSA( 256): setLpaVolume(0) 04-17 21:41:51.512 D/AudioSessionOutALSA( 256): Setting LPA volume to 0 (available range is 0 to 100) Change-Id: I6e0ee8cd7c2f577ca5b4cb834c7a83703db4b167
* Camera parameters for Samsung's qcom based legacy devicesArne Coucheron2013-04-212-0/+52
| | | | Change-Id: I8f84e06771338d992521e69704d21cd113201b4b
* Camera: QCOM legacy definitions,HTC cam switch,NO_UPDATE_PREVIEWtbalden2013-04-146-0/+131
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | Updating Camera parameters, and service with several compatibility patches: 1., QCOM legacy CameraParameters and functions: Adding these will make ICS qcomm camera blobs work on JB 4.2. Use BOARD_USES_QCOM_LEGACY_CAM_PARAMS := true in BoardConfig to use this. Also fixes the problem of legacy qcom camera blobs with recent libcameraservice QCOM_HARDWARE changes, where the orientation is messed up on first preview: the fix is leaving out setPreviewWindow(window) for legacy qcomm cam blobs. 2., Also adding NO_UPDATE_PREVIEW flag for legacy purposes. 3., Adding adding the set orientation patch: Always set buffers orientation when setting the qcom camera preview window Fixes the rotated preview on TouchPad https://github.com/CyanogenMod/android_frameworks_base/commit/1a8e41a3c7434737db89b604352575f8b3109e7a 4., camera: Full support for HTC camera switch 5., Adding more parameters for HTC Explorer Change-Id: I2bfc72b8ef027665356788f1db7f96b31d037dbe
* libstagefright: Squashed audio fixes from CodeAuroraHaynes Mathew George2013-04-0811-92/+241
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | libstagefright: Return seek position until seek has been processed If it so happens that the client to TunnelPlayer (e.g. AwesomePlayer) queries for the current time before data from the new seek position is given to the compressed driver, we need to return the seek position Change-Id: If709e61f67cc8e81d34c14d19145dc61ecd82c2b CRs-Fixed: 454825 libstagefright: Use 64 bit offsets only when needed. For enabling >2GB recording, 64 bit offsets are needed for file writing. So, this feature was turned on by default. This in turn increased the file size. With this change, by default this feature will be off and turned on only when required. - Use 64 bit offsets for resolutions >= 720p. - Limit maximum file size for recording to 4GB. - Set max file size only if no value is set from the client. - Fix MPEG4Extractor to use 64 bit offsets CRs-Fixed: 273144, 285785, 288319 (cherry picked from commit 04476a3fb89dfbb025f7852dd4d62cae72385f1a) Change-Id: I00af2c7cddbbf86c566fe4bb989fe728ca06dd19 libstagefright: TunnelPlayer sync fix - Allow close on the AudioSink to be called from the extractor thread and the application thread. - This fixes a race condition where an onPauseTimeout event scheduled from the main thread closes the audio sink while the extractor thread was about to issue write() on audio HAL. (note: on HAL, not audio sink) Change-Id: I22a5c655dfcb40f3cbda3765dc23ad8e6f99c9bb CRs-Fixed: 443205 Frameworks/av: Fix to prevent deadlock in AudioEffects -Write is blocked waiting for effect chain lock and this causes decoder thread to wait indefintely. -Sometimes it is observed that effectschain is locked before mLPAEffectChain is initialized and but unlocking is skipped if mLPAEffectChain is initialized in between.Due to this LPA silence and framework reboot issues are observed as applyEffectsOn() cannot acquire lock to progress further. -Use flag to check if all effects have been locked and unlock accordingly to prevent the deadlock scenario. (cherry picked from commit 011db22abf565dfbe7f9d0a5c7af7564587b3b48) Change-Id: I82cfdab045ecf077f0ba0185fc693fc623fa10db CRs-Fixed: 435661, 435664, 435680, 430309 audio: Use tunnel player only for music stream - Check stream type before creating tunnel player to use tunnel player only for STREAM_MUSIC Change-Id: I6e4b58524e61441ad2e09499bd9187c6dd56cd3d framework/av: Fix for audio recording test through CTS - Issue: Failure in stop is observed with the audio recording test through CTS. TestScenario: When the audio record test is initiated in the CTS console, the recording session is force closed with a notification File Size limit exceeded. Further, the stop fails with the same message(notification of the File size exceeded error). - Cause: The calculation of nTotalBytesEstimate for the recording session exceeds the limit 95 percent of mMaxFileSizeLimitBytes. As a result of size deficit, the recording is stopped at the beginning of the recording session notifying MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED. - Fix: The factor size used in the calculation of nTotalBytesEstimate has been updated properly for 64bit file offset setting. The setParam64BitFileOffset in StagefrightRecorder::prepare() is executed based on two additional validations so that the factor size is updated appropriately. Change-Id: I4749ce8f9735ccc9e1d9e49718c36470837ab27f CRs-Fixed: 396057 audioflinger: apply volume on direct track when track is active During back to back tunnel playback, we encounter a race condition where setVolume can be called when the track is not updated to active state. Fix to apply the volume on direct track only when the track is in active state. Change-Id: I70c289fbf8a9266bae0bd01b04be9f43ad32c70d CRs-Fixed: 464148 LPAPlayer: Update condition to ignore seek - Reject seek if the new seek time is greater than the current position and within an empirical limit (default 60ms). - This limit must be measured for each target. Change-Id: I86b44679fb5fe442bb5adb510c62514f6be3d304 CRs-Fixed: 453067 audioflinger: for DirectAudioTrak, call startOutput before stream is active For LPA and Tunnel playback, when resume is done in paused state, before starting actual playback, volume should be set through AudioPolicy command thread. Change-Id: I7ee1098058c01a35a3e7181d3b291304abf3cac1 CRs-Fixed: 464348
* Silence error when DASH player isn't presentSteve Kondik2013-04-081-3/+1
| | | | Change-Id: I9e5c22af1a5cf916b0efaec7ca1c5f48f6d0c82a
* MediaScanner: Ignore dirs with .noscanandnomtpDavid Daynard2013-03-241-0/+12
| | | | | | | | | | | | | | Completely ignore directories with .noscanandnomtp files in them. Placing a .nomedia file will still scan a directory but exclude it from the media database. This is so the file may still be presented for MTP purposes. Placing .noscanandnomtp completely prevents the scan, which saves considerable processing power and battery life on systems with numerous media files, but prevents them from being seen over MTP. Change-Id: Ibff2a9f2525255a2ac34132eeee36734962fbdd7
* qcom-fm: audio: add support for FM featureMingming Yin2013-03-0613-12/+618
| | | | | | | | | | Change-Id: Idd5c7a0364710d54809ef5d4c7b2404b22dc4cf6 Conflicts: include/media/IAudioFlinger.h media/libmediaplayerservice/StagefrightRecorder.cpp media/libstagefright/Android.mk
* audio: Subsystem Restart changesRavishankar Sarawadi2013-03-047-5/+88
| | | | | | | | | | | | | | | | - Handle new ADSP status parameter - media/libmedia: Add new ADSP status audio parameter - framework/av: Add handling of new key-pair value in Audio Flinger - Handle Tunnel mode SubSys Restart - framework/av: Post SSR event to Audio Flinger - media/libmedia: Post SSR event to AudioTrack - media/libmediaplayerservice: Post SSR event to MediaPlayerService - media/libstagefright: Post SSR event to TunnelPlayer Change-Id: I8c8385af45be91caf7d7160ab2e0236d6591b159
* libstagefright: Stop the logspam from LPAPlayerSteve Kondik2013-02-271-2/+2
| | | | Change-Id: Id6319774806e152333d468b9ff62d148e476555a
* libstagefright: Fix incorrectness nPortIndex value for QueryCodecYuanQY2013-02-271-1/+1
| | | | | | | The query index is wrong, it will make a death loop in my ME722 when get resource thumbnail for MPEG4 video. Change-Id: I64532156e762b847a8eae59560fb828549c29519
* libstagefright: Squashed commit of LPA/tunnel updates from CAFHaynes Mathew George2013-02-275-98/+289
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | libstagefright: Exceptions in using Tunnel mode decode - Accumulate all known exceptions to a separate function Change-Id: I61bbc288c9a087559db210e76141b8c57e67fff0 CRs-Fixed: 432080 libstagefright : Stability fixes for Tunnel Player (part 2) - Synchronize b/w reset() and onPauseTimeout - Synchronize b/w seekTo() and onPauseTimeout Change-Id: Ia5cfc6b4dcc326ead440fba35d809d4f3f1b5a81 CRs-Fixed: 449122 Revert "Revert "libstagefright: Convert mono to stereo for LPA clips"" This reverts commit 0db8a19fb3216a8a83d5d6cbd5f1ccbf997a20d8. libstagefright: Port Tunnel mode fixes to LPA - Miscellaneous fixes for seek, pause/resume, EOS handling - Miscellaneous fixes for synchronization between the decoder thread, TimedEventQueue and the player thread. - This change is a port of a similar set of changes made for TunnelPlayer Change-Id: I82c2904f7aedfb9c4f03200419fcba8b038e3d54 libstagefright: Avoid use of extra bytes to signal seek processed - A few bytes were reserved in the buffer sent by Tunnel/LPA player to audio HAL to indicate a seek has been processed and there is no need to skip it. - We won't need this method anymore as this can be fixed instead by synchronizing seekTo() and the extractor/decoder threads. Change-Id: Ic02ae1699bb59e2f6b8d9fb599d0fa43fd3f19e3 libstagefright: LPAPlayer synchronization fixes - synchronize b/w seekTo() and onPauseTimeout() - synchronize b/w reset() and onPauseTimeout() Change-Id: I29a4ccf02e28fe7b7c00e35a679ff2b5271ffb6f libstagefright: TunnelPlayer performance tweaks Some tweaks when TunnelPlayer is used for audio/video playback - Keep the extractor thread at ANDROID_PRIORITY_NORMAL - sched_yield() after reading a frame to give the video thread(s) (CallbackDispatcher and/or TimedEventQueue) to be scheduled Change-Id: If0d86d629fd0e15aff917af8589472578cd28bf4 CRs-Fixed: 444041
* libstagefright: Check for duration > 0 to avoid DivideByZero crashvivek mehta2013-02-191-1/+1
| | | | | | | | | - duration = 0 can cause divide by zero and for this clip duration is indiacted as 0. - check for duration > 0 rather than duration >= 0 Change-Id: I58ccacbf7ede892dff9626715162ea7b1f2ddbc6 CRs-Fixed: 451855
* audioflinger: Fix to set correct volume in Tunnel playbackAmal Paul2013-02-192-1/+6
| | | | | | | | | | | | - After a pause and resume, tunnel playback volume is always set to maximum irrespective of the volume value before pause. - The cause for this is, the stream volume is not used to set the volume in directaudiotrack. - Fix is to use the stream volume to set volume during tunnel playback. Change-Id: I59cda146ed88bd5c4186aeb9ae5d165f4a27493f CRs-fixed: 452285
* libstagefright: Add support for frame-by-frame modeShalaj Jain2013-02-196-5/+40
| | | | | | | - Set decoder in frame-by-frame mode always, except for interlaced content, for which arbitary mode should be set Change-Id: I8195a40549898b43a0e03d65663c7148f458c448
* ACodec: Support for dynamic port reconfigApurupaPattapu2013-02-192-2/+171
| | | | | | | | | | | - On port settings changed first flush output port - Move ACodec to new state called FlushingOutputState - Flush all output buffers, wait till all decoded buffers are rendered - Then disable output port, and allocate output buffers with new resolution, and reset native window Change-Id: Iafa266371ed2a87b909fbcb4eeae6b64208df617
* libmediaplayerservice: Add new player for DASHAjit Khare2013-02-193-0/+27
| | | | | | | | | - Add new player factory to support dash playback. - DASH urls end with .mpd. When media player receives an url with .mpd, it will use new factory to instantiate the player to be used for supporting DASH playback. Change-Id: I69e5a08fb2baf89d97b1e0711dbe52a8b1c39c29
* audio: apply volume on DirectOutput streamsMingming Yin2013-02-161-0/+1
| | | | | | | | - Call stream->set_volume to apply volume on particular stream of DirectOutputs Change-Id: I98e3b856ee508d407f893afad66caade5eda3e4a CRs-Fixed: 445953
* libstagefright: Adjustment for TunnelPlayer buffer sizeHaynes Mathew George2013-02-161-1/+1
| | | | | | | | | | - Make necessary adjustments to align the buffer size for TunnelPlayer to satisfy the following conditions - Buffer size is a multiple of LCM(1,2,4,6,8) - Buffer size is aligned to 4k. CRs-Fixed: 447274, 442365 Change-Id: I1b7f9ac3cf8ff86f972a8b6798bfcff3a4ba7c64
* Merge "fmradio: use caf naming" into cm-10.1Giulio Cervera2013-02-161-1/+1
|\
| * fmradio: use caf namingGiulio Cervera2013-02-141-1/+1
| | | | | | | | Change-Id: I9e0fefa3f2dabe991c6be63ab13a18ca38c37f71
* | Merge tag 'android-4.2.2_r1' of ↵Steve Kondik2013-02-1268-5216/+2718
|\ \ | |/ |/| | | | | | | https://android.googlesource.com/platform/frameworks/av into 1.1 Android 4.2.2 release 1
| * Logging to investigate a crashGlenn Kasten2012-12-072-1/+17
| | | | | | | | | | Bug: 6490974 Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
| * Bug fix for the MediaPlayer::prepare() api.Dylan Powers2012-11-293-5/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For an MP3 source, within the prepare command, ID3 tags are checked in search of gapless playback info. This causes problems for streamed sources. If ID3v2 tags aren't present, then a check is done for ID3v1 tags. This results in a read command that asks the cache to jump to the end of the file, and subsequently make an extra http call to request those bytes. For a streamed source, this causes the file to not be downloaded automatically when MediaPlayer::prepare() is called, and causes stuttering and extra buffering time to be needed when start() is finally called. The solution is to ignore the ID3v1 tags as the gapless info would never exist there, and only check for ID3v2 tags. Cherrypicked from external contribution ffd6ffc5429c45577fd8e9f8fa90e79bb91b8a84 b/7638165 Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
| * Fix log spamGlenn Kasten2012-11-291-1/+1
| | | | | | | | Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
| * Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into ↵Andreas Huber2012-11-2910-30/+267
| |\ | | | | | | | | | jb-mr1.1-dev
| | * [wfd] Support a low(er) power state by triggering PAUSE/RESUME.Andreas Huber2012-11-2910-30/+267
| | | | | | | | | | | | | | | Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727 related-to-bug: 7638150
| * | Enable retransmission of UDP packets in case we want to use itAndreas Huber2012-11-291-1/+1
| |/ | | | | | | | | | | | | in our upcoming wfd _sink_ implementation. Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a related-to-bug: 7638155
| * Merge "Reduce the frequency of IDR frames and add intra-fresh mode support ↵James Dong2012-11-281-1/+18
| |\ | | | | | | | | | for WiFi display" into jb-mr1.1-dev
| | * Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi ↵James Dong2012-11-281-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | display The time interval between periodic neighboring IDR frames is increased from 1 second to 15 seconds. o related-to-bug: 7524791 Change-Id: Ic32f37448f952f329549eda5e73637ee3b02f046
| * | Added optional intra macroblock refresh support for encodingJames Dong2012-11-282-1/+50
| |/ | | | | | | | | | | o related-to-bug: 7524791 Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
| * Add support for HLS playlists of type 'event'.Andreas Huber2012-11-2723-46/+276
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | related-to-bug: 6870049 Squashed commit of the following: commit eee2f3ba6bb7335f4e285632726db85645669929 Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 15:02:01 2012 -0800 Make everything a lot less verbose by default. Change-Id: I884d7a7901aa1e7d4ff590f065ca57a79d2af8b3 commit 6bbdb837ed5bd88008e45efb8faf595e4051ba26 Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 14:34:46 2012 -0800 HLS now properly signals media time changes at discontinuities including the start of playback (which may not necessarily be at time 0 if the playlist is of type 'event' and hasn't completed yet). Change-Id: I5ab747d024f9b8d0df72a4e06a12ebb29f62802e commit 1555589832b1878a144a976a643e1af4d61f877c Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 14:32:28 2012 -0800 As part of a time discontinuity, clients of IStreamListener can now signal the corresponding media time after the discontinuity, i.e. the first PTS timestamp following the discontinuity will be considered equivalent to the specified media time and media buffers timestamped accordingly. Change-Id: Id7db7679b7faa6efd6270620ff52e34e884f3e92 commit 5c24c605c073a11c426d025b1e7478fc1ad8365a Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 13:00:56 2012 -0800 NuPlayer sources now expose flags() and can announce that duration may change (increase) dynamically, in which case duration will be polled at 1 second intervals and communicated to the upper layers. Change-Id: I45102909b7a19eed0dda576747e3814d742a0eea commit ecb71de8e281e61971a2cd73e7161a97540bc357 Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 12:57:47 2012 -0800 Stop caching duration in MediaPlayer, duration could increase dynamically. Change-Id: I7bb2f16c0abe49debdf45c776d2266aa069d7791 commit 544aec5823e6d7a3e97e15b6b23546616bcd343e Author: Andreas Huber <andih@google.com> Date: Tue Nov 27 08:46:28 2012 -0800 An attempt to add support for "event" style HLS playlists. Change-Id: I3dfb2e801ecaff8f5d8bdb3a4fca1b18aeeb2c60 Change-Id: I48cf7f65a654d33f2f49ded74f8be22aed9e3b98
| * Merge "New VHQ resampler" into jb-mr1.1-devGlenn Kasten2012-11-2711-4818/+1427
| |\
| | * New VHQ resamplerGlenn Kasten2012-11-2711-4818/+1427
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Squashed commit of the following: commit 12b6952da9f25e94d06dd7185bce255924e7e791 Author: Mathias Agopian <mathias@google.com> Date: Mon Nov 19 15:27:26 2012 -0800 fix a typo in SINC resampler that prevented tracks to be mixed we were always erasing the current mix instead of mixing into it. Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2 commit 0019ce082df430278f14ab922e900ce33b64897d Author: Dave Bort <dbort@google.com> Date: Tue Nov 13 01:30:32 2007 -0800 Rename "TARGET" to "MODULE" in the build system. Part one of the grand renaming. API_CHANGE: Third parties may need to update their makefiles. Any variables with "LOCAL" and "TARGET" in their names should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE, LOCAL_MODULE_TAGS. PRESUBMIT=passed OCL=39840 Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe Signed-off-by: Glenn Kasten <gkasten@google.com> commit f01adc0cef0e39e75c76d9195ac26a94cac0a100 Author: Glenn Kasten <gkasten@google.com> Date: Wed Nov 14 08:32:08 2012 -0800 Fix build warnings Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b commit 9bb031a565c753a03d9c9397edea318947d80528 Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 10 04:44:30 2012 -0800 more optimizations... calculate the offsets from the phase differently, this happens to reduce the register pressure in the main loop, which in turns allows the compiler to generate much better code (doesn't need to spill a lot of stuff on the stack). this gives another 15% performance increase Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96 commit 5a951598f31217b8cd2babd0720c9608ee17291a Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 10 03:26:39 2012 -0800 refactor code to improve neon code we want to make sure we don't transfer data from the neon unit to the arm register file, as this can be quite slow. instead we do all the calculation on the neon side and write the result directly to main memory. Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187 commit b381ee9e83bc9fd18986e79c7809841514ed590e Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 15:16:13 2012 -0800 NEON optimized SINC resampler this currently gives us a 60% to 80% boost depending on the quality level selected. Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b commit bea077354210242ea193a50b0dbab0fedab25df3 Author: Mathias Agopian <mathias@google.com> Date: Mon Nov 5 01:51:37 2012 -0800 minor cleanups Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f commit 8f4ed7decbe161a5ff38200b218f5216d80aba46 Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 18:49:14 2012 -0800 improve resample test - handle stereo input - input file can now be ommited, in this case a linear chirp will be used automatically - better usage information Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22 commit 5fcd634ea6cb4df27c495abe20f5f9b8ff55d128 Author: Mathias Agopian <mathias@google.com> Date: Sun Nov 4 02:03:49 2012 -0800 change how we store the FIR coefficients The coefficient table is now transposed and shows much better its polyphase nature: we now have a FIR per line, each line corresponding to a phase. This doesn't change at all the results produced by the filter, but allows us to make slightly better use of the data cache and improves performance a bit (although not as much as I thought it would). The main benefit is that it is the first step before we can make much larger optimizations (like using NEON). Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43 commit d652231abf4c7e2ea1fc89caae730cec1f7259a1 Author: Mathias Agopian <mathias@google.com> Date: Sat Nov 3 23:37:53 2012 -0700 improve SINC resampler performance The improvement is about 60% by just tweaking a few things to help the compiler generate better code. It turns out that inlining too much stuff manually was hurting us. Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304 commit 9dc68ef5b94c700c4ee68790e8cbb334c90a538d Author: Mathias Agopian <mathias@google.com> Date: Thu Nov 1 21:03:46 2012 -0700 new coefficients for the vhq resampler previous coefficients were provided by a 3rd party and didn't have a way to re-generate them. we're now using the 'fir' utility. the performance of the filter is virtually identical, except for the down-sampling case which seems slightly better now: It looks like both the previous and new coefficients are generating some sort of clipping for full-scale signals in the down-sampling case (although the new ones seem better), the reason for that is unknown (see bug: 7453062) Also updated the HQ coefficients for the down-samplers, previous ones were a little bit too conservative -- the new ones push the cut-off frequency up by about 1 KHz. Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647 commit 38e0b8560a6fc1b7124e22e0e09a84a285182f8e Author: Mathias Agopian <mathias@google.com> Date: Tue Oct 30 13:51:44 2012 -0700 fix SINC resampler on non ARM architectures make sure the C version of the code generates the same output than the ARM assemply version. Change-Id: Ide218785c35d02598b2d7278e646b1b178148698 commit a1878128b182696ba508569b4d211d0dfae92463 Author: Mathias Agopian <mathias@google.com> Date: Tue Oct 30 12:49:07 2012 -0700 fix another issue with generating FIR coefficients the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were missing the 2*f scale factor. This explains why we were seeing clipping and had to manually scale the filter down. Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186 commit 1a0fb993430acc9f601e6c538305bc407c20ac5d Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 17:13:20 2012 -0700 fir a typo that caused up-sampling coefficiens to be wrong up-sample coefficient were generated with a cut-off frequency of 24KHz intead of ~20KHz, which caused more aliasing in the audible band. also increased the attenuation to 1.3 dB on both up and down sampling coefficient to avoid clipping. Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e commit 9520ad6862bd682ad075a9d9e3e94ada9f6e58b6 Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 17:13:16 2012 -0700 test-resample: clip instead of overflowing Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878 commit ba36656300f250f7f1fdeb75149749344260e6cb Author: Mathias Agopian <mathias@google.com> Date: Sun Oct 21 01:01:38 2012 -0700 a test app for the resamplers Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607 commit 056a08b9bfd33cf27228c992adc8293a56b01be8 Author: Mathias Agopian <mathias@google.com> Date: Fri Oct 26 14:11:01 2012 -0700 reenable the cubic resampler cubic resampler was disabled because it hadn't been qualified, however after I did some tests, it does improve significantly the sound quality over the order-1 resampler, even if it is still quite bad. also HIGH_QUALITY resampler was partially disabled, it's now fully enabled. It's a big improvement over the cubic resampler in terms of aliasing noise (it's not as good in the pass-band). Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b commit 8c0241d3ff50ae85167f69b3bd369244894cfa44 Author: Mathias Agopian <mathias@google.com> Date: Fri Oct 26 13:48:42 2012 -0700 improve SINC resampler coefficients - we increase the interpolation precision from 4 to 7 bits this doesn't increase CPU power required, it only increases the size of the filter table but significantly reduces the noise introduced by the quantization of the impulse response. - the parameters of the filter are set such that aliasing is rejected at 80 dB below 20 KHz. Because we don't use a lot of coefficient (to save compute power), there are quite a bit of attenuation in the pass-band: starting at 9KHz for the down-sampler (48 to 44.1), and starting at 13 KHz for the up-sampler (44.1 to 48) -- the transition band is about 15 KHz. Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838 commit 69e7dab2192adc1f780464146810629ebd01b145 Author: Pixelflinger <mathias.agopian@gmail.com> Date: Thu Oct 25 19:43:49 2012 -0700 improve fir tool: cleanup, better default, bug fixes - all parameters can be changed on the command-line - added float output - added debug option - added an option to generate a polyphase filter coefficients - added an attenuation option in dBFS - added a lot of comments and references - fixed kaiser window parameter also the default should generate a filter with 80 dB rejection (of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition band around ~20 KHz (for 48 KHz sampling rate). It's not very good but corresponds to the current code. commit 8347499d105a50257c18e9dac652e750b06428b1 Author: Glenn Kasten <gkasten@google.com> Date: Mon Oct 22 17:09:27 2012 -0700 Increase allowed number of VHQ resamplers to 3 Bug: 7378660 Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6 commit f91cf3cad7f5c4d52614271c89ab468741c5d24c Author: Mathias Agopian <mathias@google.com> Date: Sun Oct 21 03:04:05 2012 -0700 Fix a typo that caused the high quality resampler to produce garbage the problem is that if libaudio_resampler is present, it is those coefficients that will always be selected, but the correct meta-data. Bug: 7385994 Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621 commit e158a9e4262a174c59469a205658bc3ca4078234 Author: Dan Bornstein <danfuzz@google.com> Date: Fri Oct 3 10:34:57 2008 -0700 Manually merge change #111620 from tc3 to mainline, to keep the automerger from choking on it. p4 sync p4 integrate -r -b android_to_tc3 //...@111620,111620 p4 resolve -a p4 resolve # resolve a couple merge travesties PRESUBMIT=passed BUG=1399648 TBR=edheyl OCL=111902 Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2 Signed-off-by: Glenn Kasten <gkasten@google.com> commit b9f3c26032be7a6ea01a10d93d94826f449e68ab Author: Dave Bort <dbort@google.com> Date: Fri Jan 18 14:51:05 2008 -0800 Rename "Makefile" to "Android.mk" throughout the tree. For <http://b/issue?id=960416>. I've tested this as much as I can, but 1500 open files = easy to mess things up. Please let me know if there's a problem rather than rolling back this change. PRESUBMIT=passed BUG=960416 TBR=joeo OCL=46537 Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb Signed-off-by: Glenn Kasten <gkasten@google.com> commit 0c22a9a44c4103483fba1d944acf1354c5eb1617 Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 23:44:25 2007 -0700 Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now. Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507 Signed-off-by: Glenn Kasten <gkasten@google.com> commit b85e41487983ad085b859acf8251e7e54480308a Author: Mathias Agopian <mathias@google.com> Date: Mon Oct 29 04:34:36 2007 -0700 A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time. Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec Signed-off-by: Glenn Kasten <gkasten@google.com> commit ba3949ef17cac2ba71cc3096c413782a49c922e5 Author: Mathias Agopian <mathias@google.com> Date: Thu Aug 23 21:01:28 2007 -0700 fix a few small typos in the FIR computation Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40 Signed-off-by: Glenn Kasten <gkasten@google.com> commit 7474bfa7de2604021963794dddfe44985648db6a Author: Mathias Agopian <mathias@google.com> Date: Thu Aug 23 03:16:02 2007 -0700 This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler. Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067 Signed-off-by: Glenn Kasten <gkasten@google.com> Bug: 7577965 Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
| * | Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !nullIgor Murashkin2012-11-263-9/+14
| |/ | | | | | | | | Bug: 7564718 Change-Id: Ie7821cdee57966d88af048759578439a3e6ecb2e
| * Static AudioTrack plays twice initiallyGlenn Kasten2012-11-161-1/+1
| | | | | | | | | | Bug: 7528721 Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
| * Merge "Properly signal an error if codec configuration goes wrong." into ↵Andreas Huber2012-11-141-1/+6
| |\ | | | | | | | | | jb-mr1.1-dev
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