| 16e79115e497386eaf010af388627f94314a55a3 |
|
01-Aug-2014 |
Chong Zhang <chz@google.com> |
MediaCodecSource: stop puller from caller's thread instead of looper
Currently CameraSource/AudioSource's stop() and read() are both called
from the puller's looper. This works if source operates normally (i.e.
read() returns regularly before source is stopped), as the stop() will
eventually be handled by the looper. However, if for some reason the
source hang, it will get stuck in read(), and the stop() will never
be processed, which could lead to ANR (in addition to the source hang).
We need to move the source's stop out of the puller's looper. It also
can't be on MediaCodecSource's looper, because the source's stop
synchrounously waits for all outstanding buffers to return, these
are only returned when MediaCodecSource's looper processes the buffer.
This change moves the stop to MediaCodecSource::stop, after encoder
is shutdown.
Bug: 16522726
Change-Id: Ie91f563c5d8a98ab091bf1945af4e51f662b9403
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| d459e303f2f1d34c2b8c1af4f943747b20a8b1b2 |
|
24-Jul-2014 |
Chong Zhang <chz@google.com> |
signal mFrameAvailableCondition when AudioSource is stopped
This allows last read() to exit properly, otherwise it could get
stuck waiting for the condition forever.
Bug: 15849309
Change-Id: I79bba53630e59783a5fc041cfc5d3fdb2545cc2c
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| a5750e0dad9e90f2195ce36f2c4457fa04b2b83e |
|
19-Jun-2014 |
Mark Salyzyn <salyzyn@google.com> |
libstagefright: 64-bit compile warnings
Change-Id: I3d1146714fa23be3d4e696599b6f70cac1f9d28b
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| bce50bfc3846ab008bafa75c5d3f29fd7b5395f7 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter frameCount
It's size_t consistently
Change-Id: I29638ef59ac773218025f2403a3508a307b487e0
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 838b3d8bafa4a781e277870dee4e0390165cff52 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter notificationFrames
It's uint32_t consistently
Change-Id: If8298c7e9aeea2b951fe47b675adbdf48d104846
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 84333e0475bc911adc16417f4ca327c975cf6c36 |
|
08-Feb-2014 |
Andreas Huber <andih@google.com> |
warnings be gone.
Change-Id: Ie3bae3f037730e316d7fca12e7a3527973f752ef
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 1d6fa7af1288b550faabe4ec2cf98684236723db |
|
11-Feb-2014 |
Narayan Kamath <narayan@google.com> |
resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
|
| 377b2ec9a2885f9b6405b07ba900a9e3f4349c38 |
|
03-Feb-2014 |
Kévin PETIT <kevin.petit@arm.com> |
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
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 5f972c031d4061f4f037c9fda1ea4bd9b6a756cd |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::getInputFramesLost() cleanup
Fixed bug that if the binder call failed (for example if the
IAudioFlinger binder is dead), then getInputFramesLost was returning
garbage. Now it correctly returns zero, which is the error value for
this method.
The type declarations for getInputFramesLost were inconsistent:
a mixture of unsigned int, size_t, and uint32_t. Now it returns uint32_t
everywhere, which is what the underlying HAL API returns.
Added a FIXME about the side effect behavior. This will need review
for multi-client.
Change-Id: Ifa2e117a87dbd0c1f2c892a31d1c3dd919bf1a0a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| b65473f4f881ee7c0a24217ceac69514f6c127d5 |
|
17-Oct-2013 |
Vineeta Srivastava <vsrivastava@google.com> |
Fix kAutoRampDurationUs overflow issue
When kAutoRampDurationUs multiplies with mSampleRate, it overflows to int64.
Type cast it to int64 to make sure RampDurationUs properly.
BUg: 11162491
Change-Id: I4f93bc9acc8456e25623a9255ca7a5b206425009
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| e2ffd5b583da9d30d96710b0e8879e90b2b51d30 |
|
13-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord must be used as sp<> only
Bug: 9423855
Change-Id: I78ba8228c60dff11fb466156bb632c5dda45cdaf
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| e33054eb968cbf8ccaee1b0ff0301403902deed6 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| af5dd7753e62353411cf0daf3b513c38818e9662 |
|
02-Oct-2012 |
Andreas Huber <andih@google.com> |
ALooper::GetNowUs() now relies on systemTime instead of gettimeofday.
Change-Id: Ia31eb0940b02581327a8bf51af6df135f9ab6de3
related-to-bug: 7266324
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 082830f92373a1b9e512dbbfb940187ffa1c2c6f |
|
30-Aug-2012 |
Andreas Huber <andih@google.com> |
Prepare for transmitting audio through AudioSource.
AudioSource can now be configured to output buffers timestamped based
on looper time (absolute) instead of based on systemTime() relative to
start time.
Change-Id: I8eca42648eb50033ac4aafbe5daac64a98a40690
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| dd8104cc5367262f0e5f13df4e79f131e8d560bb |
|
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more consistently
In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.
In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.
Remove unused IAudioFlinger::channelCount(audio_io_handle_t).
In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.
Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| a0108697f86d8625eb7ad3f13e422427fe7573ca |
|
30-Jun-2012 |
James Dong <jdong@google.com> |
Fixed a media server crash due to unintialized mRecord member variable
mRecord is not properly initialized if the call to AudioRecord::getMinFrameCount() fails.
media server crashes when the unintialized mRecord object is deleted in AudioSource's destructor.
Change-Id: Ia89222789d044c11c9957a99725bc89f9c709e17
related-to-bug: 6744014
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| f92eec53f886f43e4374a36195be55f2a7bbcf36 |
|
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove AudioRecord record_flags
Change-Id: I021ddcc1bcb63132a4597d13e3d09db2a5f2c628
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| e49f2b424318aa8e830e7a1338e5e32ab82992f9 |
|
13-Jun-2012 |
Eric Laurent <elaurent@google.com> |
stagefright: fix AudioRecord callback buffer size
Make sure that the maximum number of frames passed to
AudioSource by the AudioRecord callback always fits within
the maximum buffer size defined by kMaxBufferSize.
Also make sure that the total AudioRecord buffer size is more
than the minimum required.
Change-Id: I26a1f998e0cf75ac88b02e67ec9d8db3c0cca193
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| ab334fd351ae5a0e18903da123d63e565b536874 |
|
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
audio_channel_in/out_mask_from_count
and avoid ambiguous term "channels" where it might be confusing
as to whether it is a channel mask or channel count
Change-Id: I744fa08ccb6001a98c97bd638d2c9d56836c4234
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 679ab0b0792846a89162ce41c953819d70030112 |
|
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Use AudioRecord::record_flags consistently
Change-Id: I6f369a2b99eb515603bc7d5629a07db2b96783fe
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| b575ddce78d266fa218006f90306158dda5c8f56 |
|
14-Feb-2012 |
James Dong <jdong@google.com> |
Limit the amount of audio record data in each buffer
o The size of each input buffer should be less than or equal to kMaxBufferSize
o related-to-bug: 5977032
Change-Id: I04343169aac3df56694aad4ba7967ec45337ad7e
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| b44c9d2bdc0d5b9cb03254022a58e017b516e9e6 |
|
03-Feb-2012 |
James Dong <jdong@google.com> |
Don't call virtual functions in the destructor for audio and camera source classes
Change-Id: Ia74ffc1c0cbd7971697f5e3c476e340ec5c7727a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| eba51fb3a361f67a6a64d5a16eba6084fe27d60e |
|
23-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_source_t consistently
Was a mix of audio_source_t, uint8_t, and int.
Related fixes:
- fix comments in MediaRecorder.java
- AudioPolicyService server side was not checking source parameter at
all, so if the client wrapper was bypassed, invalid values could be
passed into audio HAL
- JNI android_media_AudioRecord_setup was checking source for positive
values, but not negative values. This test is redundant, since already
checked at Java and now checked by AudioPolicyService also, but might
as well make it correct.
Change-Id: Ie5e25d646dcd59a86d7985aa46cfcb4a1ba64a4a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| fce7a473248381cc83a01855f92581077d3c9ee2 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| a472613aec322e25891abf5c77bf3f7e3c244920 |
|
16-Feb-2011 |
James Dong <jdong@google.com> |
A/V synchronization at the beginning of a recording session
o do not use edts/elst boxes since these optional boxes are ignored
o manipulate the first video/audio frame duration to make sure that the rest
of the audio/video is in sync (ideally, we should only manipulate
the vidoe frame duration, not the audio)
o reduce the initial audio mute/suppression period, which is used to
eliminate the "recording" sound.
bug - 3405882 and 3362703
Change-Id: Ib0acfb4f3843b365157288951dc122b006299c18
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 6b61f4355db1974cd0f0dfaa4effdd7117b9f09b |
|
15-Feb-2011 |
James Dong <jdong@google.com> |
Decouple AudioRecord read and audio encoding
bug - 3313754
Change-Id: I951dd0e21e34aa1412c391f003bc32103d0424b0
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 722555f01ace262c2aba9e1ca5d9794ce30c564f |
|
09-Feb-2011 |
James Dong <jdong@google.com> |
Catch read error from AudioRecord and do not assert
bug - 3439313
Change-Id: Ie29d6e4945978ef27fc3e5849e467d895c7736d3
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| eaae38445a340c4857c1c5569475879a728e63b7 |
|
25-Jan-2011 |
James Dong <jdong@google.com> |
Report errors to applications if AudioRecord->start() fails
bug - 3385198
Change-Id: I86ac8071eb28a538b333e102192193d1b9eda5eb
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 79e23b41fad961008bfde6e26b3c6f86878ca69d |
|
11-Dec-2010 |
James Dong <jdong@google.com> |
Revert "Allows the authoring engine to skip frame."
o Skipping frames could lead to a lot of issues such as I frames is lost etc.
It is not being used anyway.
This reverts commit 53d4e0d58e2d5c18f6e026c705af833b9bdd7aba.
Conflicts:
media/libstagefright/AudioSource.cpp
media/libstagefright/CameraSource.cpp
Change-Id: I3abba1647de48db25bdc369066eb2a7ae4dedec2
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 67e9269eaeab41a6c9a18794ebb32cbd1414381c |
|
14-Sep-2010 |
James Dong <jdong@google.com> |
Fix audio input sample timestamp when audio driver loses audio samples
Change-Id: Ic0f1489f710929af50e7714867ae5153b3242dd8
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 3c3763d2ee1cd1fba7fe522fbaf0faca315d8c2a |
|
09-Sep-2010 |
James Dong <jdong@google.com> |
HW audio encoder expects timestamp via kKeyTime from each input buffer
- This fixes media server crashes on droid
Change-Id: I7191cadc5275107425ec3ee3d437b2c5295858dc
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| d707fcb3e29707ca4a5935c294ef0b38eb5aba5f |
|
02-Sep-2010 |
James Dong <jdong@google.com> |
Calculate audio media drift time from AudioSource
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
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 6e20bdf799a6f4efa6c42121a958634ea32ed5cc |
|
01-Sep-2010 |
James Dong <jdong@google.com> |
Make sure that if initialization fails, AudioSource still behaves well.
Change-Id: I16dfc90bcb8a324d6ee9a38a5a1a31cc094c820a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| f1ae1963f5028a670573b50a9c1cfb504fc426b4 |
|
27-Aug-2010 |
James Dong <jdong@google.com> |
Suppress the video recording start signal
- bug 2950297
Change-Id: I0044d07178691feb904cf81e87c1b6d4b714dc1a
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 47204e1806da9f849464d0cef936851d7e561607 |
|
11-Aug-2010 |
James Dong <jdong@google.com> |
Handle large audio lost
Change-Id: I2687ad855aac758946954d0b3fe7aff9f7b5ae7c
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 46292fb347d72a314d985e34e5e3743d846cb9b6 |
|
30-Jul-2010 |
James Dong <jdong@google.com> |
Add lost frame handling in AudioSource
- Also collect stats on lost audio frames instead of time spent on reading
Change-Id: I6380b143e4fbdcd894491aaae523331e90d0f04f
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 542db5d438988360d491a5add1040a2df9aa90c9 |
|
21-Jul-2010 |
James Dong <jdong@google.com> |
Allows the authoring engine to skip frame.
This is 1st part of the work to allow audio and video resync if
we found out that audio and video are out of sync during authoring
- also fixed a problem in AACEncoder::read() where the buffer acquired
from the buffer group does not release when error out at
reading from source.
Change-Id: I8a2740097fcfdf85e6178869afeb9f3687a99118
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| d3d4e5069e1af0437c4f5a7b4ba344bda5b937af |
|
25-Jun-2010 |
James Dong <jdong@google.com> |
Track maximum amplitude and fix getMaxAmplitude()
- only start to track the max amplitude after the first call to getMaxAmplitude()
Change-Id: I64d3d9ca0542202a8535a211425e8bccceca50fc
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| f60cafe0e6aad8f9ce54660fa88b651ae4e749e6 |
|
19-Jun-2010 |
James Dong <jdong@google.com> |
Audio/video sync during recording (second part)
Change-Id: Iba0b35f57fdeac7ee1da16899406bf4b957a2c8c
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| be6ec71af2d12e2a55f2f0b1b77d3fa5d593a1c7 |
|
15-Jun-2010 |
James Dong <jdong@google.com> |
Remove hard-coded number of audio channels in AudioSource
Change-Id: I5f362252c25e2251bbfa9818b711ee23b4975248
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| a7d1a2dd776bf356c228785a94ba8e0ff6a2ec7f |
|
10-Jun-2010 |
James Dong <jdong@google.com> |
Initial checkin for pause and resume control
Change-Id: Ibdcf7bea5fb66baa81878704ba4091dfcfe382ee
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 365a963142093a1cd8efdcea76b5f65096a5b115 |
|
04-Jun-2010 |
James Dong <jdong@google.com> |
Initial check-in for collecting stats from authoring engine at runtime
Change-Id: I93a9d8bd260efc5e7fc135b726e3f1307c6df794
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| 050b28a593350047845a45a14cc5026221ac1620 |
|
23-Apr-2010 |
James Dong <jdong@google.com> |
Support AAC recording
- Extend the audio recording to AAC format
- Add support for setting some recording parameters
- Add stss box to the meta data in the recorded file
Change-Id: I41167bfd9d70ef9cd33906f8437b39c232b6d3b7
/frameworks/av/media/libstagefright/AudioSource.cpp
|
| e7c9cb48fec02697227bd847cd2e69432659adfd |
|
25-Jan-2010 |
Andreas Huber <andih@google.com> |
Initial checkin of AudioSource and AMRWriter, a pair of classes supporting pure-audio recording in stagefright.
related-to-bug: 2295449
/frameworks/av/media/libstagefright/AudioSource.cpp
|