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* Merge "audio policy: use audio_devices_t when appropriate"Eric Laurent2012-03-082-2/+2
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| * audio policy: use audio_devices_t when appropriateEric Laurent2012-03-082-2/+2
| | | | | | | | Change-Id: I1b3a5879e81c789fb53d356af3d3a1ee2dca955f
* | remove files that moved to frameworks/nativeMathias Agopian2012-03-077-584/+0
| | | | | | | | Change-Id: I140d291e520097b1148930f736823650e08488f7
* | IAudioFlingerClient::ioConfigChanged param2 constGlenn Kasten2012-03-062-3/+3
|/ | | | | | | | | The 3rd parameter (param2) to AudioFlingerClient::ioConfigChanged is used as an input. So changed it from void * to const void *. It is then cast to const OutputDescriptor * or const audio_stream_type_t * depending on the event. Change-Id: Ieec0d284f139b74b3389b5ef69c7935a8e5650ee
* Merge "AudioTrack: relax check on minimum buffer size"Eric Laurent2012-03-061-2/+4
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| * AudioTrack: relax check on minimum buffer sizeEric Laurent2012-03-051-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | Current AudioTrack implementation enforces that the requested audio buffer size is at least corresponding the audio latency. This requirement is too strong and leads to problems with current stagefright and AudioSink implementations when playing over output streams with long latency. Ultimately, the AudioSink design should be changed to specify a minimum buffer size in time or frames units but not in buffer count units. Change-Id: I8ba603956f92ac49143a8249572665aa548f2f0f
* | Merge "Remove unused code - MediaSourceSplitter"James Dong2012-03-061-193/+0
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| * | Remove unused code - MediaSourceSplitterJames Dong2012-03-051-193/+0
| |/ | | | | | | Change-Id: Icf37832a1c234501ee3ff91ff13eece2242425ae
* | Merge "Add channel mask in AudioSink"Jean-Michel Trivi2012-03-052-1/+4
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| * Add channel mask in AudioSinkJean-Michel Trivi2012-03-022-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | Merge "Fix typos and line length in AudioRecord comments"Glenn Kasten2012-03-021-5/+6
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| * Fix typos and line length in AudioRecord commentsGlenn Kasten2012-02-241-5/+6
| | | | | | | | Change-Id: I85cfb9a2b9b3ade098161aa7687b4d4f7eb226ea
* | Switch the way we configure for MediaPlayer retransmission.John Grossman2012-03-013-0/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | Merge "Instead of hardcoding OMX component names in our code, support"Andreas Huber2012-03-013-21/+121
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| * | Instead of hardcoding OMX component names in our code, supportAndreas Huber2012-03-013-21/+121
| | | | | | | | | | | | | | | | | | a config file instead. Change-Id: I5835903ab9f1c4a22ccc605ca99ed966767adf57
* | | Merge "renamed audio policy output flag."Eric Laurent2012-03-012-2/+2
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| * | renamed audio policy output flag.Eric Laurent2012-03-012-2/+2
| | | | | | | | | | | | | | | | | | | | | Renamed AUDIO_POLICY_OUTPUT_FLAG_INDIRECT to AUDIO_POLICY_OUTPUT_FLAG_NONE which is more appropriate. Change-Id: Ia14d60397df0f2dcd9bea0186400a09da35bc104
* | | Merge "Added permission check for all sensitive drm API calls"James Dong2012-02-291-0/+1
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| * | Added permission check for all sensitive drm API callsJames Dong2012-02-281-0/+1
| | | | | | | | | | | | | | | | | | o all drm calls using a DecryptHandle now check the permission before proceeding Change-Id: Ic992a991357876c6e4bfbfab4f5eec1fa0634a13
* | | Merge "AudioRecord const methods"Glenn Kasten2012-02-291-8/+8
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| * | | AudioRecord const methodsGlenn Kasten2012-02-281-8/+8
| |/ / | | | | | | | | | Change-Id: Ifae4fd7820b650aaca2b13c8658c292db1c46c0f
* | | Merge "Separate the notion of "stop" from that of "release", i.e."Andreas Huber2012-02-292-1/+18
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| * | | Separate the notion of "stop" from that of "release", i.e.Andreas Huber2012-02-282-1/+18
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | stop - means transition back to LOADED state and keeping the component instance allocated. release - means we get rid of the component completely. Change-Id: I40ad01ce70821faaad43f57999249904f9144924
* | | Merge "Moves TimedTextDriver.h into include/media/stagefright/timedtext."James Dong2012-02-281-0/+81
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| * | Moves TimedTextDriver.h into include/media/stagefright/timedtext.Insun Kang2012-02-271-0/+81
| | | | | | | | | | | | | | | | | | So that other players can include TimedTextDriver.h properly. Change-Id: I15e72bf655de8555eef6244a6c853c27a0828a1d
* | | fix libgui header locationMathias Agopian2012-02-273-4/+3
| | | | | | | | | | | | Change-Id: Iec71706cdd4f29c6904993648ce873e83ef9cafe
* | | libagl shouldn't export anythingMathias Agopian2012-02-271-640/+0
|/ / | | | | | | Change-Id: Ia823dbc56aab2a0b8a6063df4348fe6baac124c6
* | remove dependency on android_native{s_priv|buffer}.hMathias Agopian2012-02-242-18/+1
| | | | | | | | Change-Id: Ie4b95f7061c240f37c504414259f92d72c4ffc89
* | Add new APIs AMessage::(set|find)Buffer to make it safer to passAndreas Huber2012-02-221-0/+7
| | | | | | | | | | | | ABuffer objects through messages. Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
* | Implementation of a java media codec interface and associated tools.Andreas Huber2012-02-226-30/+316
|/ | | | Change-Id: I13e54062d4de584355c5d82bb027a68aeaf2923b
* Merge "Limit the amount of audio record data in each buffer"James Dong2012-02-211-0/+1
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| * Limit the amount of audio record data in each bufferJames Dong2012-02-151-0/+1
| | | | | | | | | | | | | | o The size of each input buffer should be less than or equal to kMaxBufferSize o related-to-bug: 5977032 Change-Id: I04343169aac3df56694aad4ba7967ec45337ad7e
* | Upintegreate AAH TX and RX players from ICS_AAHJohn Grossman2012-02-161-0/+3
| | | | | | | | | | | | | | | | Upintegrate the android at home TX and RX players developed in the ICS_AAH branch. Change-Id: I8247d3702e30d8b0e215b31a92675d8ab28dccbb Signed-off-by: John Grossman <johngro@google.com>
* | Upintegrate Audio Flinger changes from ICS_AAHJohn Grossman2012-02-163-2/+43
| | | | | | | | | | | | | | | | Bring in changes to audio flinger made to support timed audio tracks and HW master volume control. Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae Signed-off-by: John Grossman <johngro@google.com>
* | Upintegrate the common_time service from ics-aah.Mike J. Chen2012-02-164-0/+300
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Move the common_time service developed in the ics-aah branch back into master. The common_time service is a small service build to synchronize an arbitrary timeline amongst peers on a local sub-net. While running and configured, the service will elect a master from the set of available devices within the subnet, define a relationship between the common_time timeline the local time timeline (provided by the local time HAL), and then attempt to maintain synchronization between common and local time by controlling the frequency of the local time clock via the HAL, or by disciplining local time in the digital domain if the local time HAL implementation does not support HW slewing. On its own, the native common time service will do nothing until it is configured. The CommonTimeManagementService (running out of the system server process) is responsible for implementing policy regarding configuration and operation of the common_time service and will be added in a subsequent CL. Change-Id: I71292f9b9b1797665865689c4572c9d3a0552f64 Signed-off-by: John Grossman <johngro@google.com>
* | Merge "Update comments"Glenn Kasten2012-02-146-30/+65
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| * | Update commentsGlenn Kasten2012-02-146-30/+65
| | | | | | | | | | | | | | | | | | We no longer put the filename at start of file. Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
* | | Merge "Remove dead code AudioTrack::getLoop"Glenn Kasten2012-02-141-1/+0
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| * | Remove dead code AudioTrack::getLoopGlenn Kasten2012-02-081-1/+0
| | | | | | | | | | | | Change-Id: I868329c52f31bc20125f068500d8f892b4ec9796
* | | AudioRecord and AudioTrack client tidGlenn Kasten2012-02-143-3/+7
| |/ |/| | | | | | | | | Inform AudioFlinger of the tid of the callback thread. Change-Id: I670df92dd06749b057238b48ed1094b13aab720b
* | Playback rate on MediaPlayerJean-Michel Trivi2012-02-133-0/+7
| | | | | | | | | | | | | | | | | | 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
* | Move away from MediaDebug and use ADebug insteadJames Dong2012-02-101-36/+0
| | | | | | | | Change-Id: I963a3b6f79a7292891973cbeeaf3378b38629f08
* | Merge "Enable B frame support in MPEG4Writer"James Dong2012-02-081-1/+1
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| * Enable B frame support in MPEG4WriterJames Dong2012-02-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch allows us to automatically detect whether ctts box is needed in MPEG4Writer. MPEG4Writer uses ctts version 0 (non-negative offset value) store the composition time offset on a needed basis. Currently, the size of the ctts box is not optimized. Optimization will be addressed in a subsequent patch. o also changed the private method retrieveDecodingTime(bool) in OMXCodec to getDecodingTime() o related-to-bug: 4232183 Change-Id: Ic6dc7b25ecd258c2506ca4b9c25156e922456e51
* | Use audio_io_handle_t consistently instead of intGlenn Kasten2012-02-083-27/+37
| | | | | | | | | | | | | | | | Other: - add a comment to nextUniqueId - made ThreadBase::mId const, since it is only assigned in constructor. Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
* | Effect UUID inputs passed by pointer are constGlenn Kasten2012-02-083-5/+5
| | | | | | | | Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
* | Merge "Don't double destruct audio_track_cblk_t"Glenn Kasten2012-02-081-0/+1
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| * | Don't double destruct audio_track_cblk_tGlenn Kasten2012-02-031-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | Fortunately audio_track_cblk_t doesn't have a destructor, but for clarity remove the double destruction. Also add warning not to add any virtuals to audio_track_cblk_t. Change-Id: I70ebe1a70460c7002145b2cdf10f9f137396e6f3
* | | Merge "Remove dead mutex in AudioTrack/AudioRecord thread"Glenn Kasten2012-02-082-2/+0
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| * | | Remove dead mutex in AudioTrack/AudioRecord threadGlenn Kasten2012-02-022-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The client callback threads had mutexes called AudioTrackThread::mLock and ClientRecordThread::mLock. These mutexes were only used by start() and stop(), and were unused by the thread itself. But start() and stop() already have their own protection provided by AudioTrack::mLock and AudioRecord::mLock. So the thread mutexes can be removed. Change-Id: I098406d381645d77fba06a15511e179a327848ef
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