History log of /frameworks/av/media/libstagefright/AudioSource.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
4f998cdef97b9c027f145b2da5c48278e19c3d33 27-Oct-2016 Hangyu Kuang <hkuang@google.com> media: Use audio timestamp for recording.

Test: Manual test.

Bug: 32243250
Change-Id: Iac9009acf1c86388e835975f83f91d40cbffb22f
/frameworks/av/media/libstagefright/AudioSource.cpp
bd83e4b3e77b31e089832bcfbebde086392216c9 26-Oct-2016 Hangyu Kuang <hkuang@google.com> media: Adjust the first video/audio sample time in slow motion.

Test: Manual test.

Bug: 32243250
Change-Id: I9469d6f9929ca67f2f12e658382bc76e7236b256
/frameworks/av/media/libstagefright/AudioSource.cpp
b2379ba0a32638bae2ea0460644f68cf5a0967ce 24-May-2016 Eric Laurent <elaurent@google.com> Fix android.permission.CAPTURE_AUDIO_OUTPUT permission check

Make sure we use the correct PID and UID when checking
android.permission.CAPTURE_AUDIO_OUTPUT permission from audio
policy service.

Bug: 28772898
Change-Id: I8f0c45bf7a5728b4364a451eefb6c025892ae9a5
/frameworks/av/media/libstagefright/AudioSource.cpp
49d9f9a0ce21bcb8670019ff1365d1015da32e19 08-Apr-2016 Marco Nelissen <marcone@google.com> Merge "AudioSource: initialize variables" into mnc-dev am: d31f7db am: 96beb1b am: 503a93a
am: 62ccd40

* commit '62ccd40deb94ea58ca6ea870d46dbc002dd31175':
AudioSource: initialize variables

Change-Id: I3cdbb76aea2c199f06a03e24972f1a469cff26c5
96beb1b1359fe5cd4a9f2babb18fef443ea0afdd 08-Apr-2016 Marco Nelissen <marcone@google.com> Merge "AudioSource: initialize variables" into mnc-dev
am: d31f7db

* commit 'd31f7dbe92d7f54466ea962d3477c0ff293849c5':
AudioSource: initialize variables

Change-Id: I32890ae78f27060cbc1bea4e2f1a49fca29fe7e1
f88d1d8e63442d09303ca1090e1ee12e22040500 05-Apr-2016 Marco Nelissen <marcone@google.com> AudioSource: initialize variables

to prevent info leak

Bug: 27855172
Change-Id: I3d33e0a9cc5cf8a758d7b0794590b09c43a24561
/frameworks/av/media/libstagefright/AudioSource.cpp
78bd91b15ee8ea5aa2ab5a8cad7e892cb2d01c1b 26-Feb-2016 Lajos Molnar <lajos@google.com> stagefright: set pcm-encoding for raw audio tracks

Bug: 27172163
Change-Id: I4db7ace4e8cbf01939dedefbb1c54730e393530f
/frameworks/av/media/libstagefright/AudioSource.cpp
fa4303dcbeac79452f35c078c8008f5c2e7622b7 13-Aug-2015 Chong Zhang <chz@google.com> adjust audio timestamp to account for AudioRecord latency

bug: 22953017
Change-Id: Iccd1bb406ff68aa8bc3ccec35c8128625894f6ae
/frameworks/av/media/libstagefright/AudioSource.cpp
46d26dd29195450db15704e84d65740628a821fb 16-Jul-2015 Chong Zhang <chz@google.com> MediaRecorder: enable audio for slow motion recording

bug:22460747

Change-Id: I23f773ecf7b86928a8eacd25a481636e645affbe
/frameworks/av/media/libstagefright/AudioSource.cpp
3e98ecd18c906dc3ac2ff1a890f0b3163447272d 18-May-2015 Glenn Kasten <gkasten@google.com> Clear mAudioRecord and mAudioTrack on set() failure

Bug: 20818955
Bug: 20890069
Change-Id: I243205b99b198eb1b5a1a027b27cbb7d5aaaa8db
/frameworks/av/media/libstagefright/AudioSource.cpp
be71aa29a3c86d2e01cd17839d2a72ab09a1bce5 28-Apr-2015 Svet Ganov <svetoslavganov@google.com> Respect the record audio app op - media

Change-Id: I3a97977b6e9a09355e2008f780d22d480fb7308b
/frameworks/av/media/libstagefright/AudioSource.cpp
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