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
|