summaryrefslogtreecommitdiffstats
path: root/media/libmediaplayerservice/MediaPlayerService.h
Commit message (Collapse)AuthorAgeFilesLines
* Squashed commit of A/V changes from CodeAuroraKrishnankutty Kolathappilly2013-06-181-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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
* libstagefright: Squashed audio fixes from CodeAuroraHaynes Mathew George2013-04-081-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* Squashed commit of audio changes from CodeAuroravivek mehta2012-12-231-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | libstagefright: Add QC specific media format - Add QC specific media extensions - Add QC specific media definitions Change-Id: I7dca90be3b977701d9537f5e017117790a030f1f audio: Compile AudioParameter as shared library - AudioParameter as shared lib is needed by BT support in WFD source. Change-Id: I464b428ace0cbb57ce6bf7bf3b57d51a7d56f032 libstagefright: Send flush on both i/p and o/p ports together - ANR occurs in music due to race condition in OMX component if flush is issued separately for i/p and o/p ports as DSP only handles simultaneous flush on i/p and o/p ports. Change-Id: I5b16cd5a9b57c857dc8bed489d2663b8f54769e3 libstagefright: Enable extended A\V format - Add new files to support extended A\V format Change-Id: I1e61d78d35b868d55fd8e99f95de8cab9c465db4 libstagefright: Framework to plug-in propritory parser - Extend the current framework to plug-in propritory parser Change-Id: Ia586a3048420ddf1515261f20035589447263b7b audio: add support for QCOM audio formats - Add support for EVRC, QCELP, and WMA formats. Change-Id: Iaf80f982fc8b08617132dbd7d524a1748866745c frameworks/av: Support Tunnel Playback - Implement DirectTrack and DirectTrackClient - DirectTrack exposes API to client so it can create a direct output. - DirectTrackClient allows notifications to be sent to the client from DirectTrack - DirectTrack is being used for Tunnel Audio Change-Id: I2fbb18a781d8e44b8d65da9a357f6e39375f063a frameworks/av: Support LPA Playback Add support to enable Playback in LPA mode Change-Id: I1b8ac4904f4735017d62f3757ede7bbb56e62fd3 audio: Send correct channel mask in voice call recording. -Using popCount function to get channel count gives incorrect value on voice call recording. -Only STEREO and MONO bits to be considered to count channels on input Change-Id: I04c2c802422e868bdba0538ff8623dbf9eb659fe libstagefright: Thumbnail mode initial commit - use sync frame decoding mode when kClientNeedsFrameBuffer is set for hardware decoders - hardware decoder will only expect I frames, OMXCodec will set EOS on first ETB to stop more frames from being pulled - skip EOS check on FTB so that the first frame will be handled Change-Id: I0e8974e088fdcc468e27764861c128cfe291499f audio: Add support for QCOM's VOIP solution Change-Id: I1150f536fa204b535ca4019fdaa84f33f4695d93 audio: define QCOM audio parameters - Define QCOM audio paramters for FM, VOIP, fluence, SSR, and A2DP Change-Id: I29d02e37685846f6d4f00dee02e2726b015eaae7 Add ifdefs for QCOM enhanced features Change-Id: Ic8e5fe6ecc058466ced71030883b1af6c2bc055c
* Remove legacy code and APIs in wifi display related code.Andreas Huber2012-09-211-4/+0
| | | | Change-Id: Ia010e7a00534f9356b3247369d0ffd65591d91aa
* HDCP module binderizedAndreas Huber2012-09-191-0/+1
| | | | Change-Id: I866768b1e3f3b232f1934a35b65f66befc12f3f6
* Implement remote display service stubs.Jeff Brown2012-09-051-0/+5
| | | | Change-Id: I582ed000026bba6d116db8304e15a3c52f8a9a01
* Only run the wifi display RTSP server on demand, and only on the wifi directAndreas Huber2012-08-301-1/+1
| | | | | | interface. Change-Id: I7d3c44cb79cd40e73499f2d7ccf35c69b628e6d7
* Initial checkin of support for acting as a wifi display sourceAndreas Huber2012-08-291-0/+3
| | | | Change-Id: I08f17efa0c7d007e17408feb7d4fbef0a19f531a
* Add the ability to dynamicaly register MediaPlayer types.John Grossman2012-08-131-5/+1
| | | | | | | | | | | | Add the ability to dynamically register low level MediaPlayer factories which will be probed at setDataSource time to determine the proper MediaPlayerBase to instantiate. This change is in preparation for moving libaah_rtp out of frameworks/base and into phantasm platform directory. Change-Id: Icf8904db3ab9e3c85df6e780d5546d9988cb9076 Signed-off-by: John Grossman <johngro@google.com>
* Add support for deep audio buffersEric Laurent2012-04-201-2/+5
| | | | | | | | | | Allow AudioSink to use deep audio buffering when the source is audio only and its duration is more than a certain threshold. This helps improve battery life but implies higher audio latency. Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
* Add gapless playback support for NuPlayerMarco Nelissen2012-04-021-4/+7
| | | | | | | | | This makes NuPlayer use a SkipCutBuffer when needed, and adds a new AudioSink method to retrieve the number of frames written so far, so NuPlayerRenderer can calculate how much data it can write without blocking. Also make some more methods const. Change-Id: Id7d253ad8a7b85e9a84ca2baafbe32817b16c744
* Provisional support for secure decryption of media streams.Andreas Huber2012-03-261-0/+2
| | | | Change-Id: Ib3982a9c960bfdb0cb7e1b174440b141b194cfbe
* Gapless playback, step 1.Marco Nelissen2012-03-131-4/+47
| | | | | | | Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly by reusing the initial AudioTrack for subsequent players. Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
* Merge "MediaPlayerService: fix AudioSink latency"Eric Laurent2012-03-061-1/+0
|\
| * MediaPlayerService: fix AudioSink latencyEric Laurent2012-03-051-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | The AudioSink latency is currently cached when the associated AudioTrack is created. However, the AudioTrack latency can change if the AudioTrack is moved from one output stream to another. The AudioPlayer must also periodically update its view of the latency as it is needed to compensate the real audio time used for A/V sync. This fixes an A/V sync problem seen when switching A2DP on and off while playing a video. Change-Id: I28b24049ca114e1af3e24791dcc900f463536ba4
* | Add channel mask in AudioSinkJean-Michel Trivi2012-03-021-3/+3
|/ | | | | | | | | | | | | | | | Add support for specifying a channel mask when opening an AudioSink. This parameter does not replace the channel count parameter in order to not have to duplicate the logic to derive a mask from the channel count everywhere an AudioSink is used without a known mask. A mask of 0 (CHANNEL_MASK_USE_CHANNEL_ORDER) means a mask will be automatically derived from the number of channels. Update existing AudioSink implementations to use the channel mask, and users of AudioSink to specify the mask if available, and CHANNEL_MASK_USE_CHANNEL_ORDER otherwise. Change-Id: Ifa9bd259874816dbc25ead2b03ea52e873cff474
* Switch the way we configure for MediaPlayer retransmission.John Grossman2012-03-011-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | This is a cherry-pick of I6ab07d89b2eeb0650e634b8c3b7a0b36aba4e7dd with merge conflicts addressed by hand and additional changes made in response to code review feedback. Move in the direction of a more publishable API for configuring a media player for retransmission. It used to be that we used a custom invoke and a modified URL (prefixed with aahTX://). There are many issues with this technique and it was never meant to stand the test of time. This CL gets rid of all that. A new (but currently hidden) method was introduced to the java level MediaPlayer API, called setRetransmitTarget(InetSocketAddress), which allows an app writer to set the retransmit target. For now, this method needs to be called before a call to setDataSource (which is pretty unusual for the MediaPlayer API) because this mid level code uses this as a cue to instantiate an aahTX player instead of relying on the data source to select a player. When retranmit functionality becomes part of the existing android player implemenation, this set-retrans-before-set-data-source behavior can go away, along with the aahTX player itself. Change-Id: I3b46c5227bbf69acb2f3cc4f93cfccad9777be98 Signed-off-by: John Grossman <johngro@google.com>
* Playback rate on MediaPlayerJean-Michel Trivi2012-02-131-0/+4
| | | | | | | | | Add support for modifying the playback rate of a MediaPlayer by altering the sample rate of its AudioTrack. The playback rate is expressed in permille, where 1000 is the playback at normal speed. Change-Id: I981d060ab32f7bae7a767e82c60c88ae635dceed
* Fix incorrect includes of AudioTrack.hGlenn Kasten2012-01-181-0/+1
| | | | | | | Remove unnecessary includes of AudioTrack.h. Use forward declaration of class names in preference to #include when possible. Change-Id: I12982811fa75c2c7695d8bbfa595a7aaec047dc0
* Use audio_stream_type_t consistentlyGlenn Kasten2012-01-131-4/+4
| | | | | | | | | At native level it was a mixture of audio_stream_type_t, int, uint32_t, and uint8_t. Java is still int. Also fixed a couple of hard-coded -1 instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0 instead of AUDIO_STREAM_VOICE_CALL. Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
* Use audio_format_t consistentlyGlenn Kasten2012-01-051-6/+6
| | | | | | | | Was int, uint32_t, uint16_t, and uint8_t with 2-bit bitfield. Also replace 0 by AUDIO_FORMAT_DEFAULT and replace 1 by AUDIO_FORMAT_PCM_16_BIT. Change-Id: Ia8804f53f1725669e368857d5bb2044917e17975
* Remove surface legacy APIs and code.Andreas Huber2011-10-281-1/+0
| | | | | | All surfaces are now supported through surface textures. Change-Id: I95dd823e7099c0c32a48a1121624149dcc29d9c6
* Stagefright: ANW::connect in MediaPlayerServiceJamie Gennis2011-10-261-0/+5
| | | | | | | | This change moves the ANativeWindow connect and disconnect logic from MediaPlayer to MediaPlayerService::Client. Bug: 5502654 Change-Id: Ifc43b98b01ad8f35d62d7ece43110724ec7fda3d
* Require INTERNET permission for network-based content.Dave Burke2011-09-021-13/+4
| | | | | | Bug #1870981 Change-Id: Ia3ad166390c4d60cea19c3783895b078a2c4c15f
* Charge network traffic to the uid of the process using the MediaPlayer.Andreas Huber2011-07-011-1/+3
| | | | | Change-Id: I2bcb54b8232afd3fc7ee16289f37c7a7b3f23067 related-to-bug: 4517282
* Revert "Use AudioTrack::event_type not int in callback"Glenn Kasten2011-06-011-1/+1
| | | This reverts commit 2225e4b7049fa3fb9d39a068b8268b63c952d7c1
* Use AudioTrack::event_type not int in callbackGlenn Kasten2011-06-011-1/+1
| | | | Change-Id: Ifefc708d46874e04fd0d01cb6e2d43b987ee796c
* update for new audio.h header locationDima Zavin2011-05-121-1/+1
| | | | | Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876 Signed-off-by: Dima Zavin <dima@android.com>
* Add setParameter/getParameter to MediaPlayer API.Gloria Wang2011-04-291-0/+2
| | | | | | for bug 1982947 Change-Id: If3f40e4f18cbba155af29944af38bdc627f8cd53
* audio/media: convert to using the audio HAL and new audio defsDima Zavin2011-04-271-1/+3
| | | | | Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5 Signed-off-by: Dima Zavin <dima@android.com>
* - Add another parameter in notify() to be able to send timed text sampleGloria Wang2011-04-131-2/+4
| | | | | | | | through listener during video playback. - Add OnTimedTextListener in the MediaPlayer For feature request 800939. Change-Id: I65072c27acb4c0037109a72be38c73e9f667420f
* - To track the usage of all audio output devicesGloria Wang2011-03-021-0/+19
| | | | | | | | - To track the currently used audio device - The devices are separated as speaker and other audio devices - Provide the collected data to battery application through pullBatteryData() Change-Id: I374c755266b5ac6b1c6c630400f4daf901ea8acc
* Bug 3438258 Add SurfaceTexture as MediaPlayer sinkGlenn Kasten2011-02-231-0/+2
| | | | | | | | | | | | | | This change enables the use of a SurfaceTexture in place of a Surface as the video sink for an android.media.MediaPlayer. The new API MediaPlayer.setTexture is currently hidden. This includes: - New Java and C++ interfaces - C++ plumbing and implementation (JNI, Binder) - Stagefright AwesomePlayer and NuPlayer use ANativeWindow (either Surface or SurfaceTextureClient) Change-Id: I2b568bee143d9eaf3dfc6cc4533c1bebbd5afc51
* - Add method in MediaPlayerService to collect and pullGloria Wang2011-02-221-0/+24
| | | | | | | codec usage (duration) for the battery app - Collect MediaPlayer decoding usage data Change-Id: I0ef4e32b6a041ba1fe73c19f9c67185c61d03965
* Support streaming data across binder boundaries.Andreas Huber2010-11-231-0/+8
| | | | Change-Id: Ifbac61406dcb81343765f99ccba08bd90f9274cc
* Remove all traces of legacy renderer support in stagefright.Andreas Huber2010-11-161-1/+0
| | | | Change-Id: I17b8e0dbf53fca37c96830c41131b4bc0c24ca6d
* Squashed commit of the following:Andreas Huber2010-10-291-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 0d5694ba2d399dd0869532a4d6256448185a1be0 Author: Andreas Huber <andih@google.com> Date: Fri Oct 29 11:59:23 2010 -0700 suspend() and resume() methods on VideoView are back but don't do anything. They need to be back because they were public before. Change-Id: Iddfd1021ffcf341f26e8d55ba761fd33701e2425 commit 16192891ed7d349ee97e47d1729d20a2d0d247b8 Author: Andreas Huber <andih@google.com> Date: Fri Oct 29 11:47:05 2010 -0700 Revert "New API on VideoView widget to suspend/resume a session. Do not release the MediaPlayer client for video suspending/resuming." This reverts commit 2e1818a4d16c3309660f27286c77d8d1eee95a25. Conflicts: api/current.xml Change-Id: I68dd1d05871044faf3f832d0838aa40bc7f890e5 commit 8f934dc1a3ae4e60f0790fcf97671e063fa20fad Author: Andreas Huber <andih@google.com> Date: Fri Oct 29 11:44:16 2010 -0700 Revert "Release mediaplayer if the current state is not suspending. Fix for bug 2480093." This reverts commit efb882cf75eef39ecaf9f8920ed302a019fa629f. commit f2ed03550887986f39d36b5dabcd9e919949c7cf Author: Andreas Huber <andih@google.com> Date: Fri Oct 29 11:44:08 2010 -0700 Revert "Release MediaPlayer if suspend() returns false." This reverts commit 047212fd4ea360675e94d3ce83c7f5544f65b268. commit 441ecce678bd24e9660a72c8627b5bd94433ff8b Author: Andreas Huber <andih@google.com> Date: Fri Oct 29 11:40:46 2010 -0700 manually. Change-Id: I4fdd43c9f7c8b3eedddb31a196da4984e1c58e87 Change-Id: I60d4b10e7a9e4ed8d9a796f1711618f557eb6e89
* am 4f21e517: am b37fcbfd: Merge "Added getter for session Id to AudioSink" ↵Eric Laurent2010-10-101-0/+2
|\ | | | | | | | | | | | | | | | | into gingerbread Merge commit '4f21e517d09b9d793d20d64547df330fba705b3c' * commit '4f21e517d09b9d793d20d64547df330fba705b3c': Added getter for session Id to AudioSink
| * Added getter for session Id to AudioSinkEric Laurent2010-10-071-0/+2
| | | | | | | | | | | | | | | | | | | | | | Added a method to expose the audio session id at AudioSink interface so that the AudioPlayer in stagefright can retrieve it. Also: - Fixed audio effect send level not being initialized in mediaplayer. - Fixed compilation error when LOGV is enabled in mediaplayer JNI Change-Id: I4bb55454fd63d646e0e677692d737c4843fb05fb
* | Squashed commit of the following:Andreas Huber2010-08-231-1/+2
|/ | | | | | | | | | | | | | | | | | | | | | commit 35cc68814a9537c31fde146e171e7b0bbdfe211e Author: Andreas Huber <andih@google.com> Date: Mon Aug 16 08:48:42 2010 -0700 Only enable support for yuv to yuv conversion on passion, where it's available, use the slower yuv->rgb565 path everywhere else. commit d8ac5a8814103e60d11d2acf61997fc31a1dc58d Author: Andreas Huber <andih@google.com> Date: Fri Aug 13 13:56:44 2010 -0700 The software renderer takes over all rendering, converting from yuv to yuv if possible and rgb565 otherwise. commit 684972074b74318bdcb826ed9b5b0864d2d2e273 Author: Andreas Huber <andih@google.com> Date: Fri Aug 13 09:34:35 2010 -0700 A first shot at supporting the new rendering APIs. Change-Id: Iea9b32856da46950501f1a700f616b5feac710fd
* Added support for auxiliary audio effects to AudioTrack and MediaPlayer.Eric Laurent2010-07-211-1/+6
| | | | | | | | Added methods to AudioTrack and MediaPlayer java classes to enable use of auxiliary audio effects. The effect can be attached and detached by specifying its ID and the send level controlled. Change-Id: Ie74ff54a453096a742688476f612ce355543b6f3
* Added Visualizer effect.Eric Laurent2010-07-071-4/+0
| | | | | | | | | | | | | | The visualizer enables application to retrieve part of the currently playing audio for visualization purpose. It is not an audio recording interface and only returns partial and low quality audio content as a waveform or a frequency representation (FFT). Removed temporary hack made in MediaPlayer for animated wall papers based on audio visualization (snoop() method. This commit also includes a change in AudioEffect class: - the enable()/disable() methods have been replaced bya more standard setEnabled() method. - some fixes in javadoc Change-Id: Id092a1340e9e38dae68646ade7be054e3a36980e
* Added support for audio sessions in MediaPlayer and AudioTrack.Eric Laurent2010-06-221-4/+9
| | | | | | Audio sessions are used to associate audio effects to particular instances (or groups) of MediaPlayers or AudioTracks. Change-Id: Ib94eec43241cfcb416590f435ddce7ab39a07640
* New API on java's MediaPlayer to suspend/resume a session.Andreas Huber2010-02-121-0/+2
| | | | related-to-bug: 2231576
* split libsurfaceflinger_client and libcamera_client out of libuiMathias Agopian2010-02-111-1/+0
|
* Implementation of stagefright-enabled MediaPlayerService::decode functionalityAndreas Huber2010-02-101-2/+4
| | | | related-to-bug: 2359268
* API Extension: Support for optionally specifying a map of extra request ↵Andreas Huber2010-01-281-2/+10
| | | | | | | | headers when specifying the uri of media data to be played. related-to-bug: 2393577 Original change by Andrei Popescu <andreip@google.com>
* Fix issue 2285561: New AudioFlinger and audio driver API needed for A/V syncEric Laurent2010-01-261-0/+2
| | | | | | | | | | | | Added getRenderPosition() API to IAudioFlinger to retreive number of audio frames written by AudioFlinger to audio HAL and by DSP to DAC. Added getRenderPosition() API to AudioHardwareInterface to retreive number of audio frames written by DSP to DAC. Exposed AudioTrack::getPosition() to AudioSink() to make it available to media player. Removed excessive log in AudioHardwareGeneric.
* b/2353646Marco Nelissen2010-01-121-0/+1
| | | | | | | | | | | | | Fix music visualizations to also work with audiotrack callbacks, which stagefright uses. This slightly changes the way the data is stored, since before we were relying on the buffers being written always being at least 4K, whereas the callbacks are generally for smaller amounts of data. Now we append all the data to a big circular buffer, then return chunks of that buffer for visualization. When there are multiple things playing at the same time, this will give the wrong result, but (1) that was the case before as well, and (2) will be fixed once we start visualizing the mixer output instead of the mixer inputs.
* Remove old hack A/V sync hack that is no longer needed.Dave Sparks2009-12-031-2/+0
|
generated by cgit v1.1 at 2024-09-22 10:50:06 +0000