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2012-12-07Logging to investigate a crashandroid-cts-4.2_r2android-4.2.2_r1.2android-4.2.2_r1.1android-4.2.2_r1jb-mr1.1-releasejb-mr1.1-devGlenn Kasten
Bug: 6490974 Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
2012-11-29Bug fix for the MediaPlayer::prepare() api.Dylan Powers
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
2012-11-29Fix log spamGlenn Kasten
Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
2012-11-29Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into ↵Andreas Huber
jb-mr1.1-dev
2012-11-29Enable retransmission of UDP packets in case we want to use itAndreas Huber
in our upcoming wfd _sink_ implementation. Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a related-to-bug: 7638155
2012-11-29[wfd] Support a low(er) power state by triggering PAUSE/RESUME.Andreas Huber
Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727 related-to-bug: 7638150
2012-11-28Merge "Reduce the frequency of IDR frames and add intra-fresh mode support ↵James Dong
for WiFi display" into jb-mr1.1-dev
2012-11-28Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi ↵James Dong
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
2012-11-28Added optional intra macroblock refresh support for encodingJames Dong
o related-to-bug: 7524791 Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
2012-11-27Add support for HLS playlists of type 'event'.Andreas Huber
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
2012-11-27Merge "New VHQ resampler" into jb-mr1.1-devGlenn Kasten
2012-11-27New VHQ resamplerGlenn Kasten
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
2012-11-26Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !nullIgor Murashkin
Bug: 7564718 Change-Id: Ie7821cdee57966d88af048759578439a3e6ecb2e
2012-11-16Static AudioTrack plays twice initiallyGlenn Kasten
Bug: 7528721 Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
2012-11-14Merge "Properly signal an error if codec configuration goes wrong." into ↵Andreas Huber
jb-mr1.1-dev
2012-11-14Properly signal an error if codec configuration goes wrong.Andreas Huber
previously any error signaled by setupXXX inside ACodec::configureCodec would be overwritten with the result of setMinBufferSize at the end of the function. Change-Id: Id4beb571ca52ea4646239d0af006e09ce1130268 related-to-bug: 7542181
2012-11-13Various improvements of wifi display codeAndreas Huber
- manually prepend SPS/PPS if encoder doesn't support it - latency improvements - support for "our" method of optional RTP retransmission - improvements to the wfd commandline tool for testing - make it easier to turn on/off suspension of the video pipeline on idle - fixes an issue where an error during encryption would cause a SEGV - add HDCP descriptor if necessary Squashed commit of the following: commit 1115be0ebb3b885b4f1b7dba56761ca013d0ec4a Author: Andreas Huber <andih@google.com> Date: Fri Nov 9 11:32:23 2012 -0800 Better shutdown of wfd -l sessions. Change-Id: Id898a14ae21efd3b065b00a729830063d39195a7 commit 0e7d106dfe4eb6e2640b0b66c65deaba265f7ff0 Author: Andreas Huber <andih@google.com> Date: Thu Nov 8 16:38:55 2012 -0800 No more sending delay, create rtp packets upfront. Change-Id: I809a225f664fdb485c7d9a49a27886601a6a26b2 commit d399e8571b77353d59afb57508dfd2a82c1ef93a Author: Andreas Huber <andih@google.com> Date: Thu Nov 8 14:19:43 2012 -0800 Restore AudioSource buffer size, factor out TimeSeries, make suspending video optional. Change-Id: Ifdfe4d447b901e714abf52856b4641d1d55a5d41 commit f8b649f0b8f917d59f4b8a2e8e6d7db61a684a78 Author: Andreas Huber <andih@google.com> Date: Thu Nov 8 09:34:06 2012 -0800 Pull 480 frames at a time from AudioSource/AudioRecord Change-Id: I1e215abd329faec3da026631122c0f4c800c0ac4 commit 1bc13452eb35eebbba00f5da93fa86535be5db59 Author: Andreas Huber <andih@google.com> Date: Thu Nov 8 08:50:30 2012 -0800 fixed bitrate traffic simulation Change-Id: Ic5efb7cbb0b5d3b4917bc77b8ba73d447249e695 commit 016cdff18e74bdc631a5679e97192645ed095aa2 Author: Andreas Huber <andih@google.com> Date: Wed Nov 7 14:00:03 2012 -0800 resurrected "our" style of retransmission. Change-Id: I34d757aba67428437cb39b8293a9651750ad20d9 commit 384cf1a3c8fb4ec410bdf8fa5722c298e6028f3e Author: Andreas Huber <andih@google.com> Date: Tue Nov 6 09:38:55 2012 -0800 Changes to make wfd work on manta. Change-Id: I7a4e00cf16581fe2146edd1b359af195774090e4 commit 9628f24b22b35f28630d99dda3614babf51bc07e Author: Andreas Huber <andih@google.com> Date: Wed Nov 7 09:15:44 2012 -0800 Patch up rtp timestamps to more accurately measure network jitter. Change-Id: I9502a4615575f97f98a215a13131a89a6ae93c6d commit 7c891a1a24f08bbd50f55be13f7d05f43e807eb8 Author: Andreas Huber <andih@google.com> Date: Tue Nov 6 09:37:24 2012 -0800 Additions to the "wfd" tool to create a local wfd source. Change-Id: I99558653a70fdc703f9d13990b3ce1c4d3ae227a Change-Id: Ia94c63fc390f597014531073485f0cfc53b3418a
2012-11-06Reconcile with jb-mr1-factory-release jb-mr1-release - do not mergeThe Android Open Source Project
Change-Id: Idf4c25115d89397ba668fc290344b2e7c1ea1993
2012-11-06Ignore SIGPIPE during write() to broken pipeGlenn Kasten
This happens occasionally when taking a bugreport. Bug: 6447319 Change-Id: Ia6531a4a3658461f8fd3f7106e7996da7cc5933a
2012-11-01merge in jb-mr1-release history after reset to jb-mr1-devandroid-cts-4.2_r1android-4.2_r1android-4.2.1_r1.2android-4.2.1_r1.1android-4.2.1_r1jb-mr1-releaseThe Android Automerger
2012-11-01Make video encoders' bitrate mode (constant, variable, ...) configurableandroid-sdk-support_r11jb-mr1-devAndreas Huber
through the desired output format. Configure the video encoder to use constant bitrate mode for WFD. Change-Id: Id7bd619598153c13448a9c5acd69d80f8a01f333 related-to-bug: 7459597
2012-10-31merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-31Various tweaks to wfdAndreas Huber
related-to-bug: 7426218 Squashed commit of the following: commit 1553f1a1c66af998674168f7f7a3be23fcb0c794 Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 15:51:27 2012 -0700 Add LPCM, AVC and AVC HRD descriptors as necessary. Change-Id: Ibc836fced0fe37e8a25574c2295e886765b9ea6f commit 4e74db61d2d31ebe239acbdec8f110f88016a4ea Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 15:50:52 2012 -0700 added copyright headers to Sender.{cpp,h} Change-Id: If615ccb8767e32bd83ed1f0f669acc39a72489f6 commit 7144bf8ae68c5cdb8faa6e219547aabbd750f04e Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 15:50:25 2012 -0700 Reenable suspension of the RepeaterSource Change-Id: I765338fcde89c65e4b69be45a5949eba6bcdcf6f commit 812164bcfa0699821d7d8eefcc0dff96b2e2cd08 Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 14:03:50 2012 -0700 Add 2 stuffing bytes to the PES headers for audio tracks. Change-Id: I8b9c634f6a565ab7fa7ecdb610f7d8557e0b139b commit a084a741a63015d47c92d99fcd8b980fe615dc7d Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 13:19:38 2012 -0700 Fix PCM audio packetization in WFD. Change-Id: I99a435f9fe6b4397f24d6c22afae5ae2505ffc14 commit c5cb9369585f701f34bce41534940d5f9b59248f Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 13:19:12 2012 -0700 Support extraction of PCM audio from transport streams. Change-Id: I28a0516756ebcb5587325b6588df013ac871ffb9 commit b0a0512300ae037d6b39c2d04952d34b5fc12b2d Author: Andreas Huber <andih@google.com> Date: Tue Oct 30 08:54:13 2012 -0700 disable suspend of the RepeaterSource Change-Id: Ibf42a98185b0567f817ae582a82e6580f95d3d40 commit 4330e8b7668dc92a6d882b5622c0697cf292d04c Author: Andreas Huber <andih@google.com> Date: Mon Oct 29 14:11:25 2012 -0700 Better handling of datagrams in ANetworkSession reduce unnecessary copy overhead. Change-Id: I2ed8c767274ba07764f03e8d4913041168e5755f commit a44e73c322ba3f2c336f7cc4e1d63d3a74faa75d Author: Andreas Huber <andih@google.com> Date: Mon Oct 29 11:14:47 2012 -0700 Network traffic is now handled on a separate thread. Audio and video are queued to ensure proper A/V interleaving. Scheduled packet sends according to capture timestamps to reduce send-jitter. Change-Id: Ibd6357c1e663086cf87bec0a98f8e54dfdfaa0e5 related-to-bug: 7426218 Change-Id: Ia440129d656c35814abf18df06da50b73d5bb554
2012-10-29merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-26Increase bitrate now that other issues have been resolved.Dave Burke
Bug: 7426218 Change-Id: I67dfa1e4b85f326f355ad0ec5b6c699e87b45564
2012-10-26merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-26Disable silence suppression (causes issues with dongles).Dave Burke
Bug: 7426218 Change-Id: Ie1517a8017bae1f9a9b6c224cd3170dfcc5fb941
2012-10-25merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-24Camera2: Don't hold locks while waiting to sync.Eino-Ville Talvala
Bug: 7409877 Change-Id: Ia3a0bc4f0ab4e19fca868ba04a870cf8e8ee7adb
2012-10-23merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-23Merge "Don't attempt to seek audio and video separately." into jb-mr1-devAndreas Huber
2012-10-23merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-23Don't attempt to seek audio and video separately.Andreas Huber
The widevine extractor doesn't deal too well with that... Change-Id: Iadfeede4fe0c086af788c5639782854e4fbb98ff related-to-bug: 7262386
2012-10-23Merge "Camera2: Increase mode update timeout." into jb-mr1-devEino-Ville Talvala
2012-10-23Camera2: Increase mode update timeout.Eino-Ville Talvala
During camera startup, it might take a few hundred milliseconds before requests start to be dequeued by the HAL. Increase the timeout for synchronizing mode changes and triggers so that triggers near startup don't time out. Bug: 6970465 Change-Id: I9dc35378e8018ec18ae31be874fcb094f8a9a0e9
2012-10-23Merge "stagefright: discard audio at start of capture" into jb-mr1-devEric Laurent
2012-10-22stagefright: discard audio at start of captureEric Laurent
The new camcorder start sound is longer than previous one and we must discard more audio when capture starts. Ideally, camcorder should use synchronous record start. Bug 7394330. Change-Id: I219b4e231aba706776dc7ccc4f1c996eaf22f61a
2012-10-22Increase allowed number of VHQ resamplers to 3Glenn Kasten
Bug: 7378660 Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
2012-10-22merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-22Merge "Fix a typo that caused the high quality resampler to produce garbage" ↵Mathias Agopian
into jb-mr1-dev
2012-10-22WFD use of HDCP is now optional.Andreas Huber
if the sink supports it we'll use HDCP (and fail if necessary), if it doesn't we won't. If an HDCP session is established we'll tell our observer that the connection is secure, otherwise we don't. Change-Id: I7cbef384f2cf0a6ac65801c581eea227b9ef4c46 related-to-bug: 7368436
2012-10-21Fix a typo that caused the high quality resampler to produce garbageMathias Agopian
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
2012-10-20merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-19Merge "AAC DRC reference level for mobile is -16dB" into jb-mr1-devJean-Michel Trivi
2012-10-19Merge "Camera2: Fix potential deadlock" into jb-mr1-devEino-Ville Talvala
2012-10-19merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger
2012-10-19AAC DRC reference level for mobile is -16dBJean-Michel Trivi
Use -16dB as the default DRC reference level when decoding streams with DRC metadata. Bug 7370764 Change-Id: I900cee22f32384a5657fb041b69d42657bcddf09
2012-10-19Merge "Fix valgrind issues" into jb-mr1-devMarco Nelissen
2012-10-18Remove active track when thread goes to standbyGlenn Kasten
Bug: 7369232 Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
2012-10-18merge in jb-mr1-release history after reset to jb-mr1-devThe Android Automerger