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* Work to support switching transport streams mid-stream and signalling ↵Andreas Huber2010-10-072-2/+4
| | | | | | | discontinuities to the decoder. Change-Id: I7150e5e7342e1117c524856b204aadcb763e06ed related-to-bug: 2368598
* On this particular device the hardware video decoder spits out buffers that ↵Andreas Huber2010-10-072-1/+5
| | | | | | | don't actually contain our video data, so we cannot use them to restore the video frame after suspend/resume. Change-Id: I1b8fe68c1766299844fe84ebbff49cb8b3e4cc7c related-to-bug: 3070094
* Make sure to call AudioTrack::stop() instead of AudioTrack::pause() after ↵Andreas Huber2010-10-051-3/+3
| | | | | | submitting all samples to AudioTrack to make sure those remaining samples are actually played out. Change-Id: Id574a0203efcb5e565f1b0fe77869fc33b9a9d56
* fix [2835280] Add support for cancelling buffers to ANativeWindowMathias Agopian2010-10-041-4/+6
| | | | | | | | | | There is a new ANativeWindow::cancelBuffer() API that can be used to cancel any dequeued buffer, BEFORE it's been enqueued. The buffer is returned to the list of availlable buffers. dequeue and cancel are not mutually thread safe, they must be called from the same thread or external synchronization must be used. Change-Id: I86cc7985bace8b6a93ad2c75d2bef5c3c2cb4d61
* Fixed an issue where the reserved free space in the file writer was larger ↵James Dong2010-10-041-0/+1
| | | | | | | | | | | | | | | than intended The problem was that even though user does not explicitly request the max file size limit via MediaRecorder.setMaxFileSize(), the file writer sets an implicit file size limit if 32-bit file offset is used on user's behalf. The reserved free space is estimated based on the file size, if the file size limit is set by the user. The fix is to add an extra bool to tell the difference between an explit requested file size and an implicit file limit and use that to set the estimated moov box size accordingly. Change-Id: I731aca6c7833aa764ed7b905edb77721577471b3
* Issue 3032913: improve AudioTrack recovery timeEric Laurent2010-09-301-2/+5
| | | | | | | | | | | | This issue showed that when an AudioTrack underruns during a too long period of time and is therefore disabled by audioflinger mixer, it takes an additional delay of up to 3 seconds to recover. This fix adds a simple mechanism to recover immediately when the client application is ready to write data again in the AudioTrack buffer Also throttle warnings on record overflows Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
* Merge "Instead of constantly polling the AudioPlayer to see if it reached ↵Andreas Huber2010-09-281-1/+5
|\ | | | | | | EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens." into gingerbread
| * Instead of constantly polling the AudioPlayer to see if it reached EOS or ↵Andreas Huber2010-09-281-1/+5
| | | | | | | | | | | | | | finished seeking, initiate the notification from the AudioPlayer when the event happens. Change-Id: I43875b6adaf96d4e982ef3dfc3d6c8f7034ac51d related-to-bug: 3036592
* | Merge "Vorbis files may have more samples encoded that should be used, i.e. ↵Andreas Huber2010-09-281-0/+2
|\ \ | |/ | | | | we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files." into gingerbread
| * Vorbis files may have more samples encoded that should be used, i.e. we have ↵Andreas Huber2010-09-281-0/+2
| | | | | | | | | | | | | | to trim samples at the end of the stream. This is crucial for proper looping of some audio files. related-to-bug: 3036592 Change-Id: Ib142b171c829ed74156c0281d9d4543fcc96c802
* | Squashed commit of the following:Andreas Huber2010-09-271-0/+72
|/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 29a4d3effb05a2e074cb0693316ab1977baeb0b6 Author: Andreas Huber <andih@google.com> Date: Mon Sep 27 12:01:32 2010 -0700 Fully working implementation of MPEG2TSWriter (for AAC and AVC sources). Change-Id: I8a32a47565b647bf6c078c520e39565e08ea0d84 commit f4dec4c3899f3be393508e180d6c07e249d3335e Author: Andreas Huber <andih@google.com> Date: Mon Sep 27 10:36:31 2010 -0700 More reliable identification of MPEG2 transport streams. Don't keep scanning forever in case the stream does not have both audio and video tracks. Change-Id: Icc5b4e8be145b2805e8776559546a6818342aea7 commit 4fe3cc942f9b3d3cf54138b828c41214aa916dd2 Author: Andreas Huber <andih@google.com> Date: Mon Sep 27 08:23:39 2010 -0700 test code Change-Id: I16560a17661407d06497f99ff88230724bb898af commit 64d988b24f49f179a90fa677be11c823959e734b Author: Andreas Huber <andih@google.com> Date: Thu Sep 23 14:42:52 2010 -0700 First shot at supporting writing to an MPEG2 transport stream. Change-Id: Ie537939a99fa3ddc0c7661c47c18277584817c74 Change-Id: If78fd034af8f6e8ceac8dbeff96d5ecb3f6b96dc
* Remove stagefright foundation's incompatible logging interface and update ↵Andreas Huber2010-09-211-37/+12
| | | | | | callsites. Change-Id: I45fba7d60530ea0f233ac3695a97306b6dc1795c
* Rename FOCUS_MODE_CONTINUOUS to FOCUS_MODE_CONTINUOUS_VIDEO.Wu-cheng Li2010-09-211-6/+8
| | | | | | | | | | This constant is not public yet. Continuous autofocus should behave differently in still camera and camcorder. In camcorder, lens movement may be more smooth. And the triggers to start a new focus search may be different. If there is a need, FOCUS_MODE_CONTINUOUS_PHOTO can be added in the future. Change-Id: I05df9e491aca37829be3df92a73b952f26c86a4a
* Merge "HW audio encoder expects timestamp via kKeyTime from each input ↵James Dong2010-09-081-0/+1
|\ | | | | | | buffer" into gingerbread
| * HW audio encoder expects timestamp via kKeyTime from each input bufferJames Dong2010-09-081-0/+1
| | | | | | | | | | | | - This fixes media server crashes on droid Change-Id: I7191cadc5275107425ec3ee3d437b2c5295858dc
* | Modify type of some environmental reverb parametersEric Laurent2010-09-081-10/+10
|/ | | | | | | | | Changed type of decay time, reverb delay and reflections delay parameters from signed to unsigned int to match OpenSL ES interface definition. Also fixed some type casts in lvm reverb wrapper. Change-Id: I5ca5e76a87c2590f01f031f3168355586ef22556
* Ogg files can be tagged to be automatically looping, this setting always ↵Andreas Huber2010-09-031-0/+2
| | | | | | | overrides the MediaPlayer's setLooping setting. Change-Id: Ifb564c6cdf6137eac14869f9ca7d471f05a5556a related-to-bug: 2974691
* Remove unused/debugging code from MP4 file writerJames Dong2010-09-032-0/+6
| | | | | | o also makes nal length in the recorded file modifiable at runtime Change-Id: I731b4dde7070d8d9628b36b523a5b2c011c7c2cf
* Better file size estimateJames Dong2010-09-021-0/+1
| | | | | | | | | When the recorded file becomes large, the metadata size can no longer be ignored. This makes it possible to save the recorded file when the storage becomes almost full at the end of the recording session. Change-Id: Ief038080f825c9946ce550949c03e914aec1e31a
* Calculate audio media drift time from AudioSourceJames Dong2010-09-013-1/+3
| | | | | | | | | | | | | | | | | | | The problem was that the time to receive an output buffer from an audio encoder is different because the encoder does not need to read from the source for all output buffers. This leads to large fluctuation in terms of wall clock duration between two neighboring audio sample outputs from the audio encoder. As a result, the media time for the video track after adjustment using the drifting changes wildly sometimes. This patch addresses this issue by only updating the media drift time when an audio source input buffer is read. the wall clock for the audio track is also calculated at the same time when the input audio buffer is read at AudioSource. bug - 2959800 Change-Id: I3174aa182f744784b540f0a7198524d4eee8bd7b
* Remove camera metering mode API.Wu-cheng Li2010-08-301-17/+0
| | | | | | Metering mode is not supported yet. Change-Id: Id6906d6ab0cd1a9dcbc5c303d8d5081b2cda699e
* Merge "ALoopers can now be named (useful to distinguish threads)." into ↵Andreas Huber2010-08-301-0/+6
|\ | | | | | | gingerbread
| * ALoopers can now be named (useful to distinguish threads).Andreas Huber2010-08-271-0/+6
| | | | | | | | Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
* | Workaround for a QCOM issue where the output buffer size advertised by the ↵James Dong2010-08-271-0/+1
| | | | | | | | | | | | | | | | | | | | AVC encoder is occasionally too small. bug - 2882917 Change-Id: Id59d8529084c5689a26f272e0cd3b1e955fd8a30
* | Merge "Suppress the video recording start signal - bug 2950297" into gingerbreadJames Dong2010-08-271-1/+17
|\ \ | |/ |/|
| * Suppress the video recording start signalJames Dong2010-08-261-1/+17
| | | | | | | | | | | | - bug 2950297 Change-Id: I0044d07178691feb904cf81e87c1b6d4b714dc1a
* | fix a race in SF buffer managementMathias Agopian2010-08-261-1/+5
|/ | | | | | also remove some unused code. Change-Id: Iae2c3309b7a08055f3e13a5b866c5c084993e352
* Merge "Added preset reverb." into gingerbreadEric Laurent2010-08-251-1/+2
|\
| * Added preset reverb.Eric Laurent2010-08-241-1/+2
| | | | | | | | | | | | | | Modified lvm reverb wrapper code to expose a preset reverb interface. Also removed debug log from bundle and reverb wrapper. Change-Id: If9b95d91e25a6ff834decdfdda34b17df9b46967
* | Allow sniffers to return a packet of opaque data that the corresponding ↵Andreas Huber2010-08-251-2/+7
| | | | | | | | | | | | | | extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now. Change-Id: Icb77ae3ee95a69c7da25b4d3b8696c0a2d33028a related-to-bug: 2948754
* | fix [2931513] Add support for setting the orientation of an ANativeWindowMathias Agopian2010-08-241-8/+10
|/ | | | | | Also implement support for cropping. Change-Id: Iba5888dd242bf2feaac9e9ce26e404c1f404c280
* Runtime dump support for MediaWriterJames Dong2010-08-232-0/+5
| | | | Change-Id: I10b2c474de612ee4cef4b7c9eae2ee1dd8c2e895
* Merge "Visualizer: replace the FFT implementation with a faster one." into ↵Chia-chi Yeh2010-08-221-1/+0
|\ | | | | | | gingerbread
| * Visualizer: replace the FFT implementation with a faster one.Chia-chi Yeh2010-08-191-1/+0
| | | | | | | | | | | | | | | | | | This implementation uses fixed points instead of floating points. It is slightly inaccurate compared to the old one but still perfect for visualization purpose. It runs 40% faster on passion, 5 times faster on sholes, and of course 14 times faster on sapphire. Change-Id: I1e868417bcffda091becf106a7b941d02813faec
* | Merge "Add camera fps range API." into gingerbreadWu-cheng Li2010-08-201-0/+15
|\ \
| * | Add camera fps range API.Wu-cheng Li2010-08-201-0/+15
| | | | | | | | | | | | | | | | | | | | | Original preview frame rate API assumes the frame rate is fixed. It does not not work with auto frame rate camera. Change-Id: I38f7122ac8ec844ffd63558dc0763ffa17b0926a
* | | Merge "Handle the camera open failure better." into gingerbreadWu-cheng Li2010-08-191-0/+1
|\ \ \
| * | | Handle the camera open failure better.Wu-cheng Li2010-08-191-0/+1
| |/ / | | | | | | | | | | | | | | | Check if camera hardware is NULL to avoid mediaserver crash. Change-Id: Ibde0251f30bdb6b36a5d5380222d7be25ec9449c
* | | Merge "Make MediaWriter stop and pause return errors if necessary" into ↵James Dong2010-08-193-7/+8
|\ \ \ | |/ / |/| | | | | gingerbread
| * | Make MediaWriter stop and pause return errors if necessaryJames Dong2010-08-193-7/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | o Make the API consistent with SF framework, which the MediaSource provides a return status for stop o Also, helps to convey errors that occurred right when a premature stop() is called, leading to a potentially mal-formed output file. Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
* | | Merge "Adding getSupportedPreviewSizes to CameraParameters.DO NOT MERGE" ↵Wu-cheng Li2010-08-191-0/+1
|\ \ \ | | | | | | | | | | | | into gingerbread
| * | | Adding getSupportedPreviewSizes to CameraParameters.DO NOT MERGENipun Kwatra2010-08-191-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Moved functionality to parse string of sizes from getSupportedPictureSizes to parseSizesList. Added getSupportedPreviewSizes which returns a list of supported preview sizes. Change-Id: I41d4f62f9f1641e9e9258aa2ebaeda13ba846c02
* | | | Merge "Adding getSupportedPictureSizes to CameraParameters.DO NOT MERGE" ↵Wu-cheng Li2010-08-191-0/+16
|\ \ \ \ | |/ / / | | | | | | | | into gingerbread
| * | | Adding getSupportedPictureSizes to CameraParameters.DO NOT MERGENipun Kwatra2010-08-191-0/+16
| |/ / | | | | | | | | | | | | | | | | | | | | | Also added a struct 'Size' containing a width and a height field. Modified parse_size to optionally set an end pointer pointing to the character after the found size. Change-Id: I0c95ebf1ad4684721b32165f363db7d4d15a1b19
* | | In the absence of width/height information in the sdp, extract the ↵Andreas Huber2010-08-191-0/+53
|/ / | | | | | | | | | | dimensions from the avc codec specific data. Change-Id: I98c4194593c7e6e24f6fc339c862245111800293
* | Document that autoFocus must be called in auto and macro mode.Wu-cheng Li2010-08-171-2/+6
|/ | | | Change-Id: Ia52f8bc8a75a7473edff50326a4a0467f4295e6a
* Use audio clock as the reference media clockJames Dong2010-08-132-2/+8
| | | | | | | | | | | | o Only do this for realtime applications o Adjust other track clock based on audio clock o Assume other track uses wall clock as the media clock o Use some heuristics to reduce the size of stts box by 2/3. - also o Remove one unused key from MetaData.h Change-Id: Ib9432842627b61795b533508158c25258a527332
* Improve camera documentation.Wu-cheng Li2010-08-121-13/+17
| | | | Change-Id: I3c9e5e6de5ce64b8d7d892483930238fa9cc247c
* Handle large audio lostJames Dong2010-08-101-1/+2
| | | | Change-Id: I2687ad855aac758946954d0b3fe7aff9f7b5ae7c
* Support for extracting G.711 a-law and mu-law audio from WAV files and a ↵Andreas Huber2010-08-091-0/+2
| | | | | | | corresponding software decoder. Change-Id: I92685d09456c220b8c09842defb721bd55b0b9f6 related-to-bug: 2900021
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