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Glenn Kasten [Mon, 13 Jan 2014 18:37:17 +0000 (10:37 -0800)]
Document locking rules for mFlags, and fix discrepancies
Change-Id: Id45ba544cc84133ed5e578fb4fd8a11b62211dc1
Glenn Kasten [Tue, 14 Jan 2014 20:36:18 +0000 (20:36 +0000)]
Merge "Fix race in AudioTrack::getParameters()"
Glenn Kasten [Tue, 14 Jan 2014 17:33:33 +0000 (17:33 +0000)]
Merge "Improve error logging for getOutputSamplingRate"
Glenn Kasten [Tue, 14 Jan 2014 17:11:51 +0000 (17:11 +0000)]
Merge "Remove obsolete AudioTrack::processStreamEnd()"
Glenn Kasten [Mon, 13 Jan 2014 17:59:51 +0000 (09:59 -0800)]
Fix race condition in AudioRecord::pause followed by start
Bug:
11148722
Change-Id: Ia1e14133d73ac301fe06a047e70a573911822630
Glenn Kasten [Mon, 13 Jan 2014 18:25:53 +0000 (10:25 -0800)]
Improve error logging for getOutputSamplingRate
Change-Id: I3b52402a663b27efe1d7c6a4f684521f33f3ff8f
Glenn Kasten [Mon, 13 Jan 2014 18:29:08 +0000 (10:29 -0800)]
Fix race in AudioTrack::getParameters()
mOutput is protected by mLock.
Change-Id: Id02e627062855ca60f28bd8961b1d5f44939c727
Glenn Kasten [Mon, 13 Jan 2014 18:22:41 +0000 (10:22 -0800)]
Remove obsolete AudioTrack::processStreamEnd()
Change-Id: I7c01b9d2e109acf8c393d2c3b7b1985f6647d96c
Nick Kralevich [Sat, 11 Jan 2014 05:51:27 +0000 (05:51 +0000)]
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e311b15c: Merge "Fix c++11 narrowing"
* commit '
081559cc5afb6c8f2f9847e1de739f66a5a07a6b':
Fix c++11 narrowing
Nick Kralevich [Sat, 11 Jan 2014 05:48:40 +0000 (05:48 +0000)]
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e311b15c: Merge "Fix c++11 narrowing"
* commit '
514ac2bae9fc32327cac3ba22c4862e8b5ef259e':
Fix c++11 narrowing
Nick Kralevich [Sat, 11 Jan 2014 04:25:46 +0000 (20:25 -0800)]
am
e311b15c: Merge "Fix c++11 narrowing"
* commit '
e311b15cf821f65f68af2cdfe01e864cbc9a86ce':
Fix c++11 narrowing
Nick Kralevich [Sat, 11 Jan 2014 04:20:10 +0000 (04:20 +0000)]
Merge "Fix c++11 narrowing"
Andy Hung [Fri, 10 Jan 2014 19:28:54 +0000 (19:28 +0000)]
Merge "Improve dynamic audio resampler filter generation"
Glenn Kasten [Fri, 10 Jan 2014 17:34:37 +0000 (17:34 +0000)]
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61540b5f: Merge "Frameworks: AudioFlinger: Fix effects memory leak"
* commit '
0ba9f9b07d47e9e6b374798f397fe961256a6029':
Frameworks: AudioFlinger: Fix effects memory leak
Glenn Kasten [Fri, 10 Jan 2014 17:32:40 +0000 (17:32 +0000)]
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61540b5f: Merge "Frameworks: AudioFlinger: Fix effects memory leak"
* commit '
d8d45edf6a2714fff051d67aa1ac83d74bb0c6e0':
Frameworks: AudioFlinger: Fix effects memory leak
Glenn Kasten [Fri, 10 Jan 2014 17:26:02 +0000 (09:26 -0800)]
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61540b5f: Merge "Frameworks: AudioFlinger: Fix effects memory leak"
* commit '
61540b5fe82cad1c6baa018f02bd2554e62e4364':
Frameworks: AudioFlinger: Fix effects memory leak
Glenn Kasten [Fri, 10 Jan 2014 17:22:39 +0000 (17:22 +0000)]
Merge "Frameworks: AudioFlinger: Fix effects memory leak"
Marco Nelissen [Thu, 9 Jan 2014 17:56:45 +0000 (17:56 +0000)]
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5ca94d2f: Merge "HLS: Fixed rounding error with decimal segment duration"
* commit '
9dfe2ae13ef557a3b6c245bc02be8b5c71ef3fa9':
HLS: Fixed rounding error with decimal segment duration
Marco Nelissen [Thu, 9 Jan 2014 17:54:13 +0000 (17:54 +0000)]
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5ca94d2f: Merge "HLS: Fixed rounding error with decimal segment duration"
* commit '
586dda1d0845612af88e3f4ffc46ed38e304ef85':
HLS: Fixed rounding error with decimal segment duration
Marco Nelissen [Thu, 9 Jan 2014 17:50:14 +0000 (09:50 -0800)]
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5ca94d2f: Merge "HLS: Fixed rounding error with decimal segment duration"
* commit '
5ca94d2f3c4662aed7b66a97b77eb2d1948464ad':
HLS: Fixed rounding error with decimal segment duration
Glenn Kasten [Thu, 9 Jan 2014 17:44:58 +0000 (17:44 +0000)]
Merge "Cleanup AudioTrack::getMinFrameCount error handling"
Marco Nelissen [Thu, 9 Jan 2014 17:43:40 +0000 (17:43 +0000)]
Merge "HLS: Fixed rounding error with decimal segment duration"
Glenn Kasten [Thu, 9 Jan 2014 17:41:02 +0000 (17:41 +0000)]
Merge "Cleanup AudioSystem::getInputBufferSize error handling and caching"
Glenn Kasten [Thu, 9 Jan 2014 17:31:14 +0000 (17:31 +0000)]
Merge "Cleanup error handling in AudioSystem get methods"
Andy Hung [Fri, 3 Jan 2014 20:30:41 +0000 (12:30 -0800)]
Improve dynamic audio resampler filter generation
Improve dynamic audio resampler filter generation speed by 2x.
The resulting filters should be the same (excepting roundoff).
Also added check for upsampling sample rate changes to share
previously generated filters.
Modify the profiling to improve output format and sampling result
reliability.
Change-Id: I9aa6b914fd552a63f79dd4a95945df2f8275772a
Signed-off-by: Andy Hung <hunga@google.com>
Haynes Mathew George [Sat, 28 Dec 2013 00:09:28 +0000 (16:09 -0800)]
audioflinger: update track ready condition
Signal track ready if the track isStopping().
Bug:
12423190
Change-Id: Ie279995d5f90fa8211a20dfbeacc1cf9b921d0bf
Glenn Kasten [Wed, 8 Jan 2014 16:53:44 +0000 (08:53 -0800)]
Cleanup AudioTrack::getMinFrameCount error handling
Guarantee to return a non-zero frameCount for return status NO_ERROR;
Return the correct specific status_t if any of the AudioSystem APIs fail,
instead of the generic NO_INIT.
API change: getMinFramCount no longer defaults to zero on error, so
callers _must_ check the return status. This change makes
getMinFrameCount more like other APIs. All known callers were reviewed,
and they do check the return status.
Change-Id: I4a8342a75ee89a068c23c84b8380ed9d1b968507
Glenn Kasten [Wed, 8 Jan 2014 16:56:06 +0000 (08:56 -0800)]
Cleanup error handling in AudioSystem get methods
Don't return zero sample rate or frame count without an error.
Change-Id: I052d841080ed33e4f081ae9825a2f33dff444fb9
Jean-Michel Trivi [Wed, 8 Jan 2014 19:00:19 +0000 (19:00 +0000)]
Merge "Support more channel configurations in MPEG4Extractor"
Glenn Kasten [Wed, 8 Jan 2014 16:58:53 +0000 (08:58 -0800)]
Cleanup AudioSystem::getInputBufferSize error handling and caching
Previously, if the IAudioFlinger::getInputBufferSize failed,
it would return NO_ERROR but a zero buffer size value, which could
confuse the caller. Now it returns BAD_VALUE in this case.
Also it would still cache the zero buffer size. Now it does
not cache on failure.
Removed over-initialization of the cache globals.
Change-Id: I6835fcb56fe52535e018fc8c0c242115221b5d85
Jean-Michel Trivi [Wed, 18 Dec 2013 23:47:49 +0000 (15:47 -0800)]
Support more channel configurations in MPEG4Extractor
Add support for streams having an audio specific configuration with the
channel_configuration equal to zero.
Add support for 6.1 and 7.1 channel configurations.
Bug
9428126
Change-Id: Iaac2516139093579c52095d4f74ae4428f8e368a
Narayan Kamath [Wed, 8 Jan 2014 09:48:22 +0000 (09:48 +0000)]
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99044adc: Merge "Change M4OSA_ERR_CREATE to return unsigned integer."
* commit '
f41d9bc92410784fc12fa15a544282c06da76008':
Change M4OSA_ERR_CREATE to return unsigned integer.
Narayan Kamath [Wed, 8 Jan 2014 09:45:25 +0000 (09:45 +0000)]
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99044adc: Merge "Change M4OSA_ERR_CREATE to return unsigned integer."
* commit '
f60cfa7faa0bdde0a4f302f1272d6aa869588cc4':
Change M4OSA_ERR_CREATE to return unsigned integer.
Narayan Kamath [Wed, 8 Jan 2014 09:01:54 +0000 (01:01 -0800)]
am
99044adc: Merge "Change M4OSA_ERR_CREATE to return unsigned integer."
* commit '
99044adc6e209d31e9c5308d960e2bd5f6999f85':
Change M4OSA_ERR_CREATE to return unsigned integer.
Narayan Kamath [Wed, 8 Jan 2014 08:56:10 +0000 (08:56 +0000)]
Merge "Change M4OSA_ERR_CREATE to return unsigned integer."
Glenn Kasten [Wed, 8 Jan 2014 00:21:03 +0000 (00:21 +0000)]
Merge "Remove unnecessary defaults for parameters in AudioSystem"
Andy McFadden [Fri, 20 Dec 2013 21:40:34 +0000 (13:40 -0800)]
Provide raw H.264 output from screenrecord
This adds an experimental (undocumented) "--raw" flag. If set, we
output an H.264 byte stream rather than a .mp4 file.
If the filename is "-", we send the output to stdout. If stdout is a
tty, we reconfigure it to avoid CRLF line termination over adb.
Bug
12239887
Change-Id: I00ceb628bd885916eaf4658ea7f08f620ad74c03
Eric Laurent [Tue, 7 Jan 2014 02:06:07 +0000 (02:06 +0000)]
Merge "audioflinger: fix static track end detection"
Jean-Michel Trivi [Mon, 6 Jan 2014 23:39:26 +0000 (23:39 +0000)]
Merge "Update AAC decoder wrapper to latest FDK API for output channel count"
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>
Andy Hung [Mon, 30 Dec 2013 18:34:29 +0000 (10:34 -0800)]
Fix for Change 396851
Fix a typo in a variable name.
Change-Id: I2555f729fc22b9c158ae488c8cefde029fa244cb
Signed-off-by: Andy Hung <hunga@google.com>
Andy Hung [Mon, 30 Dec 2013 18:17:31 +0000 (18:17 +0000)]
Merge "Audio resampler update to add S16 filters"
Andy Hung [Mon, 9 Dec 2013 20:12:46 +0000 (12:12 -0800)]
Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
Jean-Michel Trivi [Wed, 18 Dec 2013 17:43:21 +0000 (09:43 -0800)]
Update AAC decoder wrapper to latest FDK API for output channel count
Rename decoder parameter for the maximum number of PCM output channels,
according to FDK AAC decoder interface, as defined in aacdecoder_lib.h
Bug
9428126
Change-Id: I2f0f6ca848bdbc8657d8dea589b03238245c0eaf
Zhijun He [Thu, 26 Dec 2013 19:15:29 +0000 (19:15 +0000)]
Merge "Camera2 API: fix front facing camera flip issue"
Zhijun He [Thu, 26 Dec 2013 18:38:46 +0000 (10:38 -0800)]
Camera2 API: fix front facing camera flip issue
Camera stream transform calculation should take camera facing into account.
For example, front facing camera preview stream should be horizontally or
vertically flipped.
Bug:
12300670
Change-Id: Ib497f0b8c3e65974de05d4f0aca3c51e99717c3d
Eric Laurent [Sat, 21 Dec 2013 01:37:03 +0000 (01:37 +0000)]
am
42d89e5d: am
b904d53f: am
8115f4b2: Merge "stagefright: do not offload LD-AAC decoding"
* commit '
42d89e5de5f401c7e81e1961061d07ae490c9d29':
stagefright: do not offload LD-AAC decoding
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
Eric Laurent [Sat, 21 Dec 2013 01:32:26 +0000 (01:32 +0000)]
am
b904d53f: am
8115f4b2: Merge "stagefright: do not offload LD-AAC decoding"
* commit '
b904d53ffaaafddb2f9b6f469b73d2a3bec3dd0d':
stagefright: do not offload LD-AAC decoding
Eric Laurent [Sat, 21 Dec 2013 01:30:20 +0000 (01:30 +0000)]
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1777ed30: Merge "stagefright: fix offloading HE-AAC sampling rate."
* commit '
5f7483eb6a03b3a14283c1ef09ed65bd3e015b96':
stagefright: fix offloading HE-AAC sampling rate.
Eric Laurent [Sat, 21 Dec 2013 01:28:59 +0000 (01:28 +0000)]
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5a086733: am
1777ed30: Merge "stagefright: fix offloading HE-AAC sampling rate."
* commit '
5a086733f8a7472bee85b371e9d02874c9317f3b':
stagefright: fix offloading HE-AAC sampling rate.
Eric Laurent [Sat, 21 Dec 2013 01:26:47 +0000 (17:26 -0800)]
am
8115f4b2: Merge "stagefright: do not offload LD-AAC decoding"
* commit '
8115f4b206dbd04c620f062feb3c7785e2732ab3':
stagefright: do not offload LD-AAC decoding
Eric Laurent [Sat, 21 Dec 2013 01:23:27 +0000 (17:23 -0800)]
am
1777ed30: Merge "stagefright: fix offloading HE-AAC sampling rate."
* commit '
1777ed30bfb3b9c1edc037a1b5482d5ab8f28b9b':
stagefright: fix offloading HE-AAC sampling rate.
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."
Glenn Kasten [Fri, 20 Dec 2013 23:47:09 +0000 (23:47 +0000)]
Merge "Fix some (but not all) unused parameter warnings"
Glenn Kasten [Fri, 20 Dec 2013 23:45:54 +0000 (23:45 +0000)]
Merge "Add versions of get/SetParameters without I/O handle"
Glenn Kasten [Fri, 20 Dec 2013 15:35:48 +0000 (15:35 +0000)]
Merge "Fix compile warning / incomplete initialization"
Glenn Kasten [Fri, 20 Dec 2013 00:34:04 +0000 (16:34 -0800)]
Fix some (but not all) unused parameter warnings
Change-Id: Ia99e23a0b46db3f3e6aa46f9018e63c14f4af369
Glenn Kasten [Fri, 20 Dec 2013 00:35:06 +0000 (16:35 -0800)]
Remove unnecessary defaults for parameters in AudioSystem
Change-Id: I0ee7bc13cf64f50b1ea780f4d99899aed20421a0
Glenn Kasten [Fri, 20 Dec 2013 00:35:18 +0000 (16:35 -0800)]
Add versions of get/SetParameters without I/O handle
This is a step towards hiding I/O handles from application level,
as much as possible.
Change-Id: I30f4171d5dcf77f8e8eb332ce2e9245b30f5f2e1
Glenn Kasten [Thu, 19 Dec 2013 17:09:33 +0000 (09:09 -0800)]
Fix compile warning / incomplete initialization
Change-Id: Ib4accf99be800988e081f96222e1ee73538221ec
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
Glenn Kasten [Wed, 18 Dec 2013 15:55:48 +0000 (15:55 +0000)]
Merge "Fix bug in test-resample's AudioBufferProvider"
Nick Kralevich [Wed, 18 Dec 2013 05:54:13 +0000 (05:54 +0000)]
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1f95555c: Merge "libeffects: do not use GNU old-style field designators"
* commit '
33425f660affa39da98aeb9735b82cc00dbd47a0':
libeffects: do not use GNU old-style field designators
Nick Kralevich [Wed, 18 Dec 2013 05:50:54 +0000 (05:50 +0000)]
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1f95555c: Merge "libeffects: do not use GNU old-style field designators"
* commit '
d497b648ccb316e3fbda4c081f7c3010318bbcd9':
libeffects: do not use GNU old-style field designators
Nick Kralevich [Wed, 18 Dec 2013 04:17:23 +0000 (20:17 -0800)]
am
1f95555c: Merge "libeffects: do not use GNU old-style field designators"
* commit '
1f95555c69219180f792ac742cc0e386496c29e6':
libeffects: do not use GNU old-style field designators
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
Lajos Molnar [Wed, 18 Dec 2013 00:40:25 +0000 (00:40 +0000)]
Merge "stagefright: Fix issue with tracking media format in packet source"
Glenn Kasten [Wed, 18 Dec 2013 00:14:04 +0000 (16:14 -0800)]
Fix bug in test-resample's AudioBufferProvider
The contract for AudioBufferProvider::releaseBuffer() was missing.
Bug:
12194314
Change-Id: I2fcf75e7b8eaf6db34f360206d79457a04a73565
Glenn Kasten [Tue, 17 Dec 2013 23:22:08 +0000 (15:22 -0800)]
Add ability to read .wav files to test-resample
Previously test-resample could only read .raw files, and the input
sample rate had to be specified. Now the input sample rate is derived
from the input file. This also allows us to read 8-bit PCM files,
and other formats such as floating-point in the future.
However, the ability to read raw files is lost.
A workaround is to use sox or equivalent on the host.
Change-Id: Icd06b4d02482b3ad07bf03979f46860e68d38ad9
Glenn Kasten [Tue, 17 Dec 2013 22:49:17 +0000 (14:49 -0800)]
Use libsndfile to write .wav files
This will reduce code duplication, and allow us take advantage of more
advanced capabilities of libsndfile in the future.
Change-Id: I25fa2b6d0c21e325aeaf05bda62cf7aab0c5deb4
Glenn Kasten [Tue, 17 Dec 2013 23:56:46 +0000 (23:56 +0000)]
Merge "Fix several test-resample BufferProvider bugs"
Lajos Molnar [Tue, 17 Dec 2013 22:10:46 +0000 (14:10 -0800)]
stagefright: Fix issue with tracking media format in packet source
Media format in AnotherPacketSource is now tracked across discontinuities.
This fixes a bug where format was set on queueAccessUnit and cleared on
dequeueAccessUnit, thereby allowing it to remain cleared.
Change-Id: I20975a630443f4a223a2b4344e8244f34b9560b9
Signed-off-by: Lajos Molnar <lajos@google.com>
Bug:
12060952
Eric Laurent [Tue, 17 Dec 2013 22:21:24 +0000 (22:21 +0000)]
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274cc85d: Merge "libstagefright: Delay release of wakelock in TimedEventQueue"
* commit '
b73d48783665bcf4fe3282e469ccb0fc89a7a1f9':
libstagefright: Delay release of wakelock in TimedEventQueue
Eric Laurent [Tue, 17 Dec 2013 22:21:10 +0000 (22:21 +0000)]
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f81f7a52: Merge "audioflinger: check for condition before waiting"
* commit '
58be57a04de323fa1c7cc4c1bf42f785d12056a7':
audioflinger: check for condition before waiting
Eric Laurent [Tue, 17 Dec 2013 22:21:09 +0000 (22:21 +0000)]
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645e4397: Merge "AudioTrack: fix position callback after restore"
* commit '
c63662a23c156ad6bacaf5d3524adbc2914dd712':
AudioTrack: fix position callback after restore
Eric Laurent [Tue, 17 Dec 2013 22:21:08 +0000 (22:21 +0000)]
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51d166ec: am
9060d498: Merge "update offloaded audio track sampling rate"
* commit '
9c1969c854266216e7885e02d1cfffc62b16ced4':
update offloaded audio track sampling rate
Eric Laurent [Tue, 17 Dec 2013 22:17:31 +0000 (22:17 +0000)]
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274cc85d: Merge "libstagefright: Delay release of wakelock in TimedEventQueue"
* commit '
35da47e5d55392e6adbc97efaf181cb1dd259c04':
libstagefright: Delay release of wakelock in TimedEventQueue
Eric Laurent [Tue, 17 Dec 2013 22:17:31 +0000 (22:17 +0000)]
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f81f7a52: Merge "audioflinger: check for condition before waiting"
* commit '
e4301da6b830b8ae3e27b6f095c1a96bed0b69ac':
audioflinger: check for condition before waiting
Eric Laurent [Tue, 17 Dec 2013 22:17:30 +0000 (22:17 +0000)]
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38138bb1: am
645e4397: Merge "AudioTrack: fix position callback after restore"
* commit '
38138bb1816e49f3f4e73e5bff2affe3d24a96fc':
AudioTrack: fix position callback after restore
Eric Laurent [Tue, 17 Dec 2013 22:17:29 +0000 (22:17 +0000)]
am
51d166ec: am
9060d498: Merge "update offloaded audio track sampling rate"
* commit '
51d166ec7985949fa69262f213a4162708ebe81e':
update offloaded audio track sampling rate
Glenn Kasten [Tue, 17 Dec 2013 21:54:29 +0000 (13:54 -0800)]
Fix several test-resample BufferProvider bugs
Previously getNextBuffer always returned the same data address over
and over. Now it correctly returns the right portion of the input buffer.
Previously getNextBuffer always returned the total number of frames in
the input, which might be larger than the size requested by the caller,
and/or larger than the number of remaining input frames. It also always
returned successfully, even when there should be no frames available.
This violates the contract for getNextBuffer. Now getNextBuffer will
return the maximum of the number of frames requested, and the number of
remaining frames available. If that maximum is zero, getNextBuffer will
return an error instead.
Previously releaseBuffer would silently allow releasing more frames than
were actually gotten, which violates the contract for releaseBuffer.
Now releaseBuffer checks for this and logs a message if it happens.
Add 'v' (verbose) option to log buffer provider calls.
Bug:
12194314
Change-Id: I9b915e954b3612a07ef271da8652486b8875e0fd
Eric Laurent [Tue, 17 Dec 2013 22:11:15 +0000 (14:11 -0800)]
am
274cc85d: Merge "libstagefright: Delay release of wakelock in TimedEventQueue"
* commit '
274cc85dcb255185838705a91dba00efa52bf436':
libstagefright: Delay release of wakelock in TimedEventQueue
Eric Laurent [Tue, 17 Dec 2013 22:11:14 +0000 (14:11 -0800)]
am
f81f7a52: Merge "audioflinger: check for condition before waiting"
* commit '
f81f7a52d4720f441197f75918d2b2c05d41ab45':
audioflinger: check for condition before waiting
Eric Laurent [Tue, 17 Dec 2013 22:11:14 +0000 (14:11 -0800)]
am
645e4397: Merge "AudioTrack: fix position callback after restore"
* commit '
645e43977d3aabc5addab022d772accc32fd5bdc':
AudioTrack: fix position callback after restore
Eric Laurent [Tue, 17 Dec 2013 22:11:13 +0000 (14:11 -0800)]
am
9060d498: Merge "update offloaded audio track sampling rate"
* commit '
9060d498be0c54d3caf84e2dbf1ba62516b3e76e':
update offloaded audio track sampling rate
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 16:35:48 +0000 (16:35 +0000)]
am
5c474238: am
5893736d: am
a67c7653: Merge "Increase kFastTrackMultiplier from 1 to 2"
* commit '
5c474238bc579224ec8f7267d05651b0aa934100':
Increase kFastTrackMultiplier from 1 to 2
Glenn Kasten [Tue, 17 Dec 2013 16:33:47 +0000 (16:33 +0000)]
am
5893736d: am
a67c7653: Merge "Increase kFastTrackMultiplier from 1 to 2"
* commit '
5893736dc1776008026e660526d34943ea888fd6':
Increase kFastTrackMultiplier from 1 to 2