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Glenn Kasten [Fri, 28 Feb 2014 17:12:12 +0000 (17:12 +0000)]
Merge "Fix preventing from heap overwrite in capture command"
Narayan Kamath [Thu, 27 Feb 2014 13:18:33 +0000 (13:18 +0000)]
Merge "AArch64: Correction for LP64"
Ying Wang [Wed, 26 Feb 2014 19:18:45 +0000 (19:18 +0000)]
Merge "Use multilib-friendly LOCAL_MODULE_RELATIVE_PATH"
Ying Wang [Wed, 26 Feb 2014 19:07:24 +0000 (11:07 -0800)]
Use multilib-friendly LOCAL_MODULE_RELATIVE_PATH
Change-Id: Ifef0dd59ab317b1799fc899bb2e898bb09d2e5f6
Marcus Oakland [Mon, 17 Feb 2014 08:28:25 +0000 (08:28 +0000)]
AArch64: Correction for LP64
In the Intra16x16PlanePrediction function, a u32 (unsigned 32-bit
integer) was being used for the "i" variable, and being used with a
value of 7 in the loop "for (i = 0, b = 0; i < 8; i++)" to access the
"above[6-i]" location where "above" is defined as "u8 *". Because "i"
was unsigned there was no sign extension on use with the __LP64__
64-bit pointer, so rather than 0xFFFFFFFF being treated as -1, it was
treated as 4,292,967,295 and that resulted in a SIGSEGV at an invalid
address. By changing the type of "i" to an i32 (signed 32-bit integer)
the expected sign extension occurs and the value is treated as -1.
This change fixes android.media.cts.DecoderTest#testCodeBasicH264 on
64-bit platforms
Change-Id: I85df58b0dc1d39f89ab421d04ab5481356520f0c
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Glenn Kasten [Tue, 25 Feb 2014 19:36:22 +0000 (19:36 +0000)]
Merge "Avoid a building error"
Marco Nelissen [Thu, 20 Feb 2014 17:21:06 +0000 (17:21 +0000)]
Merge "LiveSession: Use the actual, possibly redirected url as base in the M3U"
Ryszard Grzesica [Fri, 17 Jan 2014 10:40:16 +0000 (11:40 +0100)]
Fix preventing from heap overwrite in capture command
There was heap usage error in case of stop of audio framework while
capturePoint was negative. Pointer to reply data was moved but
final silence write was done using original buffer size.
Now silence set is done at the beginning under the condition
that framework has stopped.
Change-Id: I7dab1e922f1813e5fbfd4a64c8b0d15d9465520c
Colin Cross [Fri, 14 Feb 2014 01:04:44 +0000 (01:04 +0000)]
Merge "libeffects: don't set module path"
Colin Cross [Fri, 14 Feb 2014 00:39:11 +0000 (16:39 -0800)]
libeffects: don't set module path
Setting the module path breaks multilib builds where there may
be two different module paths for 32-bit and 64-bit, and there
is no need to set it to the default.
Change-Id: I95d62b5573a0bb0ab6c97649041dfc4309cbc8b9
Glenn Kasten [Thu, 13 Feb 2014 15:59:58 +0000 (15:59 +0000)]
Merge "Replace size_t in shared memory by uint32_t"
Glenn Kasten [Tue, 28 Jan 2014 19:03:28 +0000 (11:03 -0800)]
Replace size_t in shared memory by uint32_t
Eventually we may want to use uint64_t, but will need to confirm atomicity.
Bug:
12381724
Change-Id: Ia2c591d262d22b47b6f7dab4b9d9faa14b86d865
Lajos Molnar [Tue, 11 Feb 2014 21:31:40 +0000 (21:31 +0000)]
Merge "avcenc: Only do startcode escaping if the next byte requires it"
Martin Storsjo [Wed, 7 Aug 2013 11:57:20 +0000 (14:57 +0300)]
avcenc: Only do startcode escaping if the next byte requires it
Section 7.4.1 in the H.264 standard says that the only valid bytes
to follow a sequence that starts with 0x000003 are 0x00, 0x01,
0x02 or 0x03.
This makes EncodeDecodeTest pass properly when decoding using
OMX.google.h264.decoder, which is strict about the forbidden
escape sequences.
Change-Id: Ice113d9b934015003ea9cb10d0b21cee4d18d774
Narayan Kamath [Tue, 11 Feb 2014 13:35:01 +0000 (13:35 +0000)]
Merge "Make frameworks/av 64-bit compatible"
Kévin PETIT [Mon, 3 Feb 2014 12:35:36 +0000 (12:35 +0000)]
Make frameworks/av 64-bit compatible
Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
Colin Cross [Mon, 10 Feb 2014 22:34:16 +0000 (22:34 +0000)]
Merge "frameworks/av: fix errors inside ALOGV"
Colin Cross [Fri, 7 Feb 2014 04:29:44 +0000 (20:29 -0800)]
frameworks/av: fix errors inside ALOGV
Fix errors exposed by adding compile-time checking to disabled ALOGVs.
Change-Id: Ie06db81d422bb4eee7dfc10abb8d03001627af4c
Glenn Kasten [Mon, 10 Feb 2014 18:05:01 +0000 (18:05 +0000)]
Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
Glenn Kasten [Mon, 10 Feb 2014 18:04:54 +0000 (18:04 +0000)]
Merge "AudioTrack: Never try to use the fast path if resampling is required"
Lajos Molnar [Mon, 10 Feb 2014 17:45:10 +0000 (17:45 +0000)]
Merge "avcenc: Update video port parameters in the base class"
Lajos Molnar [Mon, 10 Feb 2014 17:41:54 +0000 (17:41 +0000)]
Merge "ChromiumHTTPDataSource: Keep track of the redirected URL"
Lajos Molnar [Mon, 10 Feb 2014 17:34:23 +0000 (17:34 +0000)]
Merge "SoftVPXEncoder: Set the frame size on the output port as well"
Marco Nelissen [Mon, 10 Feb 2014 17:12:50 +0000 (17:12 +0000)]
Merge "M3UParser: Fix typo in
8883a38a308"
Martin Storsjo [Mon, 10 Feb 2014 17:09:59 +0000 (19:09 +0200)]
M3UParser: Fix typo in
8883a38a308
Change-Id: I09f8deb40b8b34efd4bfcfab6866b7780f8bae96
Marco Nelissen [Mon, 10 Feb 2014 16:42:46 +0000 (16:42 +0000)]
Merge "M3UParser: Skip query strings when looking for the last slash in a URL"
Colin Cross [Fri, 7 Feb 2014 21:23:06 +0000 (21:23 +0000)]
Merge "frameworks/av: fix errors inside ALOGV"
Colin Cross [Fri, 7 Feb 2014 04:29:44 +0000 (20:29 -0800)]
frameworks/av: fix errors inside ALOGV
Fix errors exposed by adding compile-time checking to disabled ALOGVs.
Change-Id: I9602a4a485dffa3caad732c2a19ec0e41a0ac65b
Martin Storsjo [Wed, 5 Feb 2014 17:49:05 +0000 (19:49 +0200)]
AudioFlinger: Remove code for supporting resampling in fast tracks
This isn't used at the moment.
Change-Id: I4e0fb2af5f7d959dbafd5ddb7defa1c6b8e8636a
Martin Storsjo [Fri, 22 Nov 2013 15:05:05 +0000 (17:05 +0200)]
M3UParser: Skip query strings when looking for the last slash in a URL
Change-Id: I72d3a5e11fef9bbd75b291bc490c9cab1dce58da
Glenn Kasten [Wed, 5 Feb 2014 16:37:48 +0000 (16:37 +0000)]
Merge "audioflinger: conform inline ASM to UAL"
synergy dev [Tue, 4 Feb 2014 11:38:33 +0000 (06:38 -0500)]
audioflinger: conform inline ASM to UAL
Clang requires some inline ASM to conform to the UAL standards (Unified Assembler Language).
This fixes a small issue in this inline asm to allow building.
Change-Id: Ifd9b1814343ab5ade636b9401a21d575559dac16
Lajos Molnar [Mon, 3 Feb 2014 17:53:00 +0000 (17:53 +0000)]
Merge "correct one logic error in decide whether should render or not"
Jianzheng Zhou [Tue, 14 Jan 2014 09:55:16 +0000 (17:55 +0800)]
correct one logic error in decide whether should render or not
Change-Id: Ie41663f6fd5a7d983279f14a2228cb57231771bf
Signed-off-by: Jianzheng Zhou <jianzheng.zhou@freescale.com>
Martin Storsjo [Fri, 31 Jan 2014 11:30:15 +0000 (13:30 +0200)]
AudioTrack: Never try to use the fast path if resampling is required
Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
Eric Laurent [Fri, 31 Jan 2014 00:59:47 +0000 (00:59 +0000)]
Merge "fix deadlock issues that arise when there are simultaneous effect control interface calls to proxy and to non sub-effect wrappers(eg., bundlewrapper) from audioflinger Also, return NO_ERROR when CMD_OFFLOAD succeeds"
jpadmana [Thu, 14 Nov 2013 11:50:52 +0000 (17:20 +0530)]
fix deadlock issues that arise when there are simultaneous
effect control interface calls to proxy and to
non sub-effect wrappers(eg., bundlewrapper) from audioflinger
Also, return NO_ERROR when CMD_OFFLOAD succeeds
Whenever there are parallel calls to proxy and non sub-effects wrappers,
some of the calls are not completed. This is due to deadlock arsing out
of Proxy waiting for the subeffect call to return and subeffect waiting
for proxy to release lock.
The call flow is changed to a cleaner and simple one - Proxy gets the
aeli(effect library info) of subeffects during the EffectGetSubEffects()
call. Therby, proxy will manage the sub effects by itself rather than
going through effects factory.
Signed-off-by: jpadmana <jayashree.r.padmanaban@intel.com>
Bug:
12424044
Change-Id: I16852222f1d0e94e433a19177729323a4bb1c090
Eric Laurent [Fri, 31 Jan 2014 00:58:30 +0000 (00:58 +0000)]
Merge "audioflinger: fix static track end detection"
Brian Carlstrom [Thu, 30 Jan 2014 21:51:01 +0000 (21:51 +0000)]
Merge "frameworks/av: Rename persist.sys.dalvik.vm.lib to allow new default"
Brian Carlstrom [Thu, 30 Jan 2014 21:14:01 +0000 (13:14 -0800)]
frameworks/av: Rename persist.sys.dalvik.vm.lib to allow new default
Bug:
12798969
Change-Id: I2db14a7ee28db2449ec6e2384ade21944284528d
Eric Laurent [Sat, 21 Dec 2013 01:36:01 +0000 (17:36 -0800)]
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug:
12013676.
Change-Id: Iad72c921fa18d34811abf7d1073890c093a27725
Glenn Kasten [Wed, 29 Jan 2014 17:48:44 +0000 (17:48 +0000)]
Merge "AudioMixer: Remove tracks from enabledTracks after reseting outTemp"
Gaurav Kumar [Mon, 6 Jan 2014 05:27:18 +0000 (10:57 +0530)]
AudioMixer: Remove tracks from enabledTracks after reseting outTemp
If any track goes through AudioMixer::process__genericNoResampling, and
its getnextbuffer returns NULL, Then that track is removed by AudioMixer
from enabledTracks.
Thus if all tracks getnextbuffer return NULL, Then this function doesn't
reset outTemp and last buffer in AudioFlinger's mMixBuffer will be
repeated and noise is observed.
Remove tracks from enabledTracks after reseting outTemp to zero, so that
process__genericNoResampling will reset outTemp and noise won't appear.
Bug:
12450065
Change-Id: I1ccac7ee4a3bf7fd930254356c072099e11e7c19
Signed-off-by: Gaurav Kumar <gaurav.kumar@broadcom.com>
Signed-off-by: Pierre Couillaud <pierre@broadcom.com>
Colin Cross [Tue, 28 Jan 2014 00:43:21 +0000 (00:43 +0000)]
Merge "frameworks/av: convert LOCAL_MODULE_PATH to LOCAL_MODULE_RELATIVE_PATH"
Colin Cross [Sat, 25 Jan 2014 05:00:32 +0000 (21:00 -0800)]
frameworks/av: convert LOCAL_MODULE_PATH to LOCAL_MODULE_RELATIVE_PATH
LOCAL_MODULE_PATH doesn't work for multiarch builds, replace it
with LOCAL_MODULE_RELATIVE_PATH.
Change-Id: I4e4ceec61d026bbe74ba604554c06104bde42e5e
Eric Laurent [Mon, 27 Jan 2014 17:19:48 +0000 (17:19 +0000)]
Merge "Only increase the counter when we do take a wakelock"
Eric Laurent [Mon, 27 Jan 2014 17:10:34 +0000 (17:10 +0000)]
Merge "Do partial reads in MemoryLeakTrackUtil"
Oscar Rydhé [Tue, 22 Jan 2013 10:09:54 +0000 (11:09 +0100)]
Do partial reads in MemoryLeakTrackUtil
Do partial read in MemoryLeakTrackUtil dumpMemoryAddresses
to avoid using more memory than what is allocated.
Change-Id: I94feb4e00647407f938571167b981c7371f39e3d
Jimmy Dalqvist [Tue, 14 Jan 2014 13:13:52 +0000 (14:13 +0100)]
Only increase the counter when we do take a wakelock
We keep track on how many wakelocks we have taken. We always
just take one real wakelock but increase / decrease the
counter every time we try to acquire / release a wakelock.
The counter is always increased even if the power manager is
not ready, leading to an incorrect counter that could cause a
crash when we try to release it.
Make sure we only increase the counter when a wakelock, real
or counted, is taken.
Change-Id: Iad940e052694932f1dad8a1a71fa63601d289d6a
Marco Nelissen [Fri, 17 Jan 2014 21:23:17 +0000 (21:23 +0000)]
Merge "Long mp3 metadata displays as corrupted file after transfer"
Marco Nelissen [Fri, 17 Jan 2014 17:08:51 +0000 (17:08 +0000)]
Merge "Added support for ID3v2 meta data in 3gp files"
Marco Nelissen [Fri, 17 Jan 2014 16:04:36 +0000 (16:04 +0000)]
Merge "Fix compile error in NuPlayerRenderer debug print"
Oscar Rydhé [Thu, 27 Jan 2011 13:01:24 +0000 (14:01 +0100)]
Added support for ID3v2 meta data in 3gp files
Added support for parsing ID3v2 meta data from
the ID32 chunk in 3gp files. The priority will be
3gpp -> ID3v2 -> iTunes per field.
Change-Id: I0282ecab58e3e5fa6bd738078d562c8bb8ce00ed
Marco Nelissen [Wed, 15 Jan 2014 22:12:04 +0000 (22:12 +0000)]
Merge "Avoid jumps to faulty position after seeks"
Eino-Ville Talvala [Wed, 15 Jan 2014 20:26:02 +0000 (20:26 +0000)]
Merge "libcameraservice: Fix build in ISO C++11 mode"
Nick Kralevich [Sat, 11 Jan 2014 04:20:10 +0000 (04:20 +0000)]
Merge "Fix c++11 narrowing"
Glenn Kasten [Fri, 10 Jan 2014 17:22:39 +0000 (17:22 +0000)]
Merge "Frameworks: AudioFlinger: Fix effects memory leak"
Martin Storsjo [Fri, 22 Nov 2013 15:57:31 +0000 (17:57 +0200)]
LiveSession: Use the actual, possibly redirected url as base in the M3U
This fixes playback of HLS streams where the M3U playlists use
relative paths and the main playlist is opened via a redirect.
Change-Id: I787e1c050daddc566be4c8e3f84803d3f91eec46
Marco Nelissen [Thu, 9 Jan 2014 17:43:40 +0000 (17:43 +0000)]
Merge "HLS: Fixed rounding error with decimal segment duration"
Narayan Kamath [Wed, 8 Jan 2014 08:56:10 +0000 (08:56 +0000)]
Merge "Change M4OSA_ERR_CREATE to return unsigned integer."
Oscar Rydhé [Tue, 15 Oct 2013 07:54:08 +0000 (09:54 +0200)]
Fix compile error in NuPlayerRenderer debug print
When activating verbose log prints in NuPlayerRenderer the build
fails because a variable have changed but the log print hasn't been
updated.
Change-Id: I3089b087d296c37dfe6379d7e75d5892912fef96
Ashok Bhat [Thu, 2 Jan 2014 14:54:57 +0000 (14:54 +0000)]
Change M4OSA_ERR_CREATE to return unsigned integer.
While M4OSA_ERR is defined as M4OSA_UInt32, MOSA_ERR_CREATE
is defined to return M4OSA_Int32. This leads to signed/unsigned
comparison warnings. M4OSA_ERR_CREATE has been changed to return
M4OSA_UInt32 to fix this issue.
Change-Id: I71a5c50a95c7f296469604b486a1d3969d302a3f
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Eric Laurent [Sat, 21 Dec 2013 01:19:56 +0000 (01:19 +0000)]
Merge "stagefright: do not offload LD-AAC decoding"
Eric Laurent [Fri, 6 Dec 2013 19:51:42 +0000 (11:51 -0800)]
stagefright: do not offload LD-AAC decoding
For now, do not offload LD and ELD AAC decoding because there
is no way to know if it is supported by the audio DSP implementation.
The longer term fix will be to have mapMimeToAudioFormat() use the
audio object type in track metadata to refine the AAC format and the
audio HAL list supported AAC profiles.
Change-Id: Iaa88ecf3f4ae42ad48c1b42a9b007dd80eb88147
Eric Laurent [Sat, 21 Dec 2013 01:18:46 +0000 (01:18 +0000)]
Merge "stagefright: fix offloading HE-AAC sampling rate."
Yin Liu [Tue, 4 Dec 2012 08:19:53 +0000 (09:19 +0100)]
Long mp3 metadata displays as corrupted file after transfer
Cut the metadata to 1 Byte and return in function
getObjectPropertyList in order to show it properly on a PC.
Change-Id: Iefacf9fa86c20ece2572e6d95d35877a94066fe7
Eric Laurent [Fri, 6 Dec 2013 19:16:54 +0000 (11:16 -0800)]
stagefright: fix offloading HE-AAC sampling rate.
Fix HE AAC SBR sampling rate reading for explicit
signaling.
Bug:
11697128.
Change-Id: Ifec0ab9d48d9293f6774ec1efd9da9445994cb7c
Oscar Rydhé [Tue, 17 Dec 2013 08:53:18 +0000 (09:53 +0100)]
HLS: Fixed rounding error with decimal segment duration
If segment duration is specified with decimal value only the
integer value will be used, causing the stream duration to
be wrong.
Reported to Android public issue tracker:
https://code.google.com/p/android/issues/detail?id=56223
Change-Id: I34fb7a81af6ad3d9a214228cfe3724636ebf5ab5
Roger1 Jonsson [Thu, 11 Oct 2012 08:32:33 +0000 (10:32 +0200)]
Avoid jumps to faulty position after seeks
When seeking multiple times it is possible that some seeks
are discarded in AwesomePlayer, which causes unwanted jumps
to faulty positions.
The reason is that a seek flag is reset twice in AwesomePlayer.
At first when the video seek is completed and then again when the
audio seek is completed. If a new seek is made after the previous
video seek completed but before the previous audio seek completed,
the new seek position is discarded by the previous audio seek
completion.
This fix makes sure that the seek flag is reset only when video
has completed the seek.
Change-Id: I8f8741d4cb8682345f1d1855bbad57c05f4e3c8d
Nick Kralevich [Wed, 18 Dec 2013 04:12:18 +0000 (04:12 +0000)]
Merge "libeffects: do not use GNU old-style field designators"
synergy dev [Wed, 18 Dec 2013 01:48:51 +0000 (17:48 -0800)]
libeffects: do not use GNU old-style field designators
Avoiding the use of GCC extensions improves code portability
Change-Id: I9edbedc5c8ad4aa46ca54bc2e28280441431a530
Eric Laurent [Tue, 17 Dec 2013 22:06:50 +0000 (22:06 +0000)]
Merge "libstagefright: Delay release of wakelock in TimedEventQueue"
Haynes Mathew George [Fri, 6 Dec 2013 19:31:57 +0000 (11:31 -0800)]
libstagefright: Delay release of wakelock in TimedEventQueue
Delay release of wakelock in the TimedEventQueue to
after an event has been processed.
This ensures AP shutdown does not happen while an event
is ready but hasn't been processed yet.
Bug:
11976087.
Change-Id: I71a5f3ac4a57e1d05dd5d9ab5c6f91ed7bb64c87
Eric Laurent [Tue, 17 Dec 2013 22:06:10 +0000 (22:06 +0000)]
Merge "audioflinger: check for condition before waiting"
Haynes Mathew George [Wed, 4 Dec 2013 05:26:02 +0000 (21:26 -0800)]
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
has already been satisfied before waiting.
Bug:
11824817
Change-Id: Ic8c2090d521ecd6a30b76ee75635258d35eb1eff
Eric Laurent [Tue, 17 Dec 2013 22:05:19 +0000 (22:05 +0000)]
Merge "AudioTrack: fix position callback after restore"
Eric Laurent [Wed, 27 Nov 2013 22:29:13 +0000 (14:29 -0800)]
AudioTrack: fix position callback after restore
When restoring an AudioTrack, the next position callback point
should not be modified and set ahead of current buffer head.
Otherwise, as frames are dropped, the new position is never reached
and an application relying on position callbacks to reload the buffer
would be stalled.
Bug:
11868603.
Change-Id: I93b2a311642a0c89944b78bcc0482d4ceed98ae4
Eric Laurent [Tue, 17 Dec 2013 22:04:08 +0000 (22:04 +0000)]
Merge "update offloaded audio track sampling rate"
Eric Laurent [Sat, 27 Jul 2013 00:16:50 +0000 (17:16 -0700)]
update offloaded audio track sampling rate
AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
Glenn Kasten [Tue, 17 Dec 2013 16:26:41 +0000 (16:26 +0000)]
Merge "Increase kFastTrackMultiplier from 1 to 2"
Andy McFadden [Wed, 11 Dec 2013 21:30:41 +0000 (21:30 +0000)]
Merge changes Ia684fde5,I58fcb526
* changes:
Fix the help text
screenrecord fixes
Andy McFadden [Wed, 11 Dec 2013 21:29:49 +0000 (21:29 +0000)]
Merge "Add "--bugreport" option to screenrecord"
Andy McFadden [Wed, 20 Nov 2013 00:48:50 +0000 (16:48 -0800)]
Fix the help text
Pesky bloggers.
(cherry-pick from Ia8677b054423db292a34e28337431b57804df259)
Change-Id: Ia684fde52697ea78fca79de958ef8b31a50e68ba
Andy McFadden [Tue, 19 Nov 2013 20:50:17 +0000 (12:50 -0800)]
screenrecord fixes
Fixes to issues identified during code review.
(cherry-pick from I2203694acb5c0544878f64f4347d29ad1a0725c4)
Change-Id: I58fcb5264fc17b26fac4b03f95d35262e9e199e2
Andy McFadden [Fri, 18 Oct 2013 14:31:41 +0000 (07:31 -0700)]
Add "--bugreport" option to screenrecord
The --bugreport option adds two visible features: (1) a timestamp
overlay that (mostly) matches logcat, making it easier to match what
appears in the video with what's in the log, and (2) an "info page"
at the start of the video that shows the system configuration.
Enabling this option adds an additional composition step,
increasing the overhead of screenrecord. Depending on the device
and circumstances, this may be unnoticeable or very pronounced.
If --bugreport is not enabled, the overhead of screenrecord is
unchanged.
We also now track device orientation changes. This is currently
detected by polling surfaceflinger, which is suboptimal. As a
result, we detect the rotation too late, and get a weird mixed
frame before the start of the animation for 90-degree changes.
Also, allow the bit rate to be specified as e.g. "4M" for 4Mbps.
Also, --rotate is now deprecated.
Bug
11220305
Bug
11136964
(cherry pick from Ibb94b81d2f73547b95d7a47e027da75fab187a4f)
Change-Id: I829a91aaca5ab82a07c14172d9e188ec38f14e57
The Android Open Source Project [Thu, 5 Dec 2013 20:38:14 +0000 (12:38 -0800)]
Glenn Kasten [Tue, 3 Dec 2013 17:06:43 +0000 (09:06 -0800)]
Increase kFastTrackMultiplier from 1 to 2
Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
Marco Nelissen [Thu, 28 Nov 2013 17:26:16 +0000 (17:26 +0000)]
Merge "NuPlayer: Use a software renderer when using software codecs"
Marco Nelissen [Tue, 26 Nov 2013 02:10:18 +0000 (02:10 +0000)]
Merge "Fix SIGABRT when playing mp4 file"
Marco Nelissen [Mon, 25 Nov 2013 16:39:56 +0000 (08:39 -0800)]
Fix SIGABRT when playing mp4 file
If the track duration was shorter than the segment duration, the calculated
encoder padding would be negative, resulting in a crash.
b/
11823061
https://code.google.com/p/android/issues/detail?id=62610
Change-Id: I3989ad88caea38d212b61355c15aec13382c6116
Marco Nelissen [Mon, 25 Nov 2013 16:34:06 +0000 (16:34 +0000)]
Merge "Fixed data offset at parsing IPMP Descriptors"
Martin Storsjo [Fri, 22 Nov 2013 16:11:58 +0000 (18:11 +0200)]
ChromiumHTTPDataSource: Keep track of the redirected URL
This makes the code actually match an existing comment in
DrmInitialization, which claimed that mURI was the redirected
URL and not the original one.
Change-Id: I0a5cc65f520f1482ff91320ae78af84a8a681ee3
Martin Storsjo [Sat, 10 Aug 2013 20:13:30 +0000 (23:13 +0300)]
SoftVPXEncoder: Set the frame size on the output port as well
This makes sure the MediaCodec output MediaFormat contains
the right width and height.
Change-Id: Ic97af3b5850ebaf563533c3d1cae992be3e4d074
Ichitaro Kohara [Thu, 6 Jun 2013 03:19:17 +0000 (12:19 +0900)]
Fixed data offset at parsing IPMP Descriptors
MPEG4Extractor::parseDrmSINF() miscalculated data offset in parsing
IPMP Descriptors. This commit makes it to take in 2 bytes which is
the size of IPMPS_Type field.
Change-Id: I42cbb6793af9d9b2f14dbfdd7a616096002793f9
The Android Open Source Project [Fri, 22 Nov 2013 18:35:20 +0000 (10:35 -0800)]
Martin Storsjo [Fri, 15 Nov 2013 10:22:23 +0000 (12:22 +0200)]
avcenc: Update video port parameters in the base class
This makes sure that the right parameters are returned on both
input and output ports if queried after setting.
This also makes sure that the output MediaFormat from the
MediaCodec class contains the right video size.
Change-Id: I0667b3b4c6bb90331ad0ae7d51388e1bca3d1bbd
Marco Nelissen [Fri, 15 Nov 2013 22:21:32 +0000 (14:21 -0800)]
am
24605338: Merge "Fix metadata access" into klp-dev
* commit '
246053380a9f628405a29a055a3f1f4fba13ed5b':
Fix metadata access
Marco Nelissen [Fri, 15 Nov 2013 21:58:45 +0000 (21:58 +0000)]
Merge "Fix metadata access" into klp-dev
Marco Nelissen [Fri, 15 Nov 2013 21:49:58 +0000 (13:49 -0800)]
Fix metadata access
Metadata string pointers become invalid after setting more metadata,
so don't cache them.
b/
11692062
Change-Id: Iaf1afb24cf53f7fa36f49ce759355693494076e5
Eric Laurent [Fri, 15 Nov 2013 21:09:54 +0000 (13:09 -0800)]
am
d7e59228: audioflinger: do not use raw pointer for tracks
* commit '
d7e59228caad3867794d847f6bf163c6495e9506':
audioflinger: do not use raw pointer for tracks