- Change how the transmission offset is calculated, to
incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
We must use the same clock as in the RTP module to be able to measure
the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.
BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/666006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
We previously had an error when a mono capture device was used with
a stereo codec. This is prevented by avoiding any remixing in
AudioProcessing. Instead, capture side downmixing is now done before
resampling. Upmixing can now be handled properly by AudioCoding,
since the AudioProcessing error condition has been removed.
On the render side, downmixing now occurs before resampling. Ideally
this would be handled still earlier in the chain. Similarly, downmixing
for the AudioProcessing reference data occurs before resampling. This
code has been refactored into RemixAndResample, with a comprehensive
unittest added in output_mixer_unittest.cc.
BUG=issue624
TEST=manually through voe_cmd_test, by using mono and stereo capture
and render devices with mono and stereo codecs. voice_engine_unittest,
voe_auto_test.
Review URL: https://webrtc-codereview.appspot.com/676004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2448 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL change the return int on Process() to meet Google Style. The change affects audio_coding and neteq.
Tests have been changed accordingly and the code has been tested on trybots, vad_unittests, audioproc_unittest, audio_coding_unittests, audio_coding_module_test and neteq_unittests.
BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/663005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2423 4adac7df-926f-26a2-2b94-8c16560cd09d
ACM testing consists of seven individual tests. Up til now we haven't used gtest everywhere, and many of the tests needs some rewriting to follow the style guide.
I've started with this tests, doing formatting, adding the test as a separate test which can now either succeed of fail in a proper way.
Still to do in this test is handling of input file, but that will be changed in a separate CL, because all tests uses the PCMFile class that will be affected by the change.
BUG=none
TEST=audio_coding_module_test, ACM_AUTO_TEST and ACM_TEST_ALL_CODECS.
Review URL: https://webrtc-codereview.appspot.com/646011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2416 4adac7df-926f-26a2-2b94-8c16560cd09d
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/637009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/605007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2363 4adac7df-926f-26a2-2b94-8c16560cd09d
coding.c, to be sure about the changelist. You can go through the test to be sure the changes are tested. However, I don't intend to commi
t the test, as it would be a source of confusion and requires hack to iSAC to be able to run the test. It is basically a one-time test.
The part which not covered in this test is where we limit payload for super-wideband bit-stream. I'll add a test for that as well.
I kept format changes at minimum in all files except isac.c, which was in bad shape, and coding changes were minimum. I'm planning to uplo
ad following patches to this CL where I try to address formatting issues. But I don't intend to change variable names, for the moment.
The refactoring is not yet finished, so you would find part of the code which could be cleaned up, especially KLT transforms in entropy_co
ding.c
Review URL: https://webrtc-codereview.appspot.com/580004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2359 4adac7df-926f-26a2-2b94-8c16560cd09d
Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.
The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 2 files, please take a quick look, if the patch is OK,
I will continue and upload a new patch that covers all of them.
it to all of them.
Review URL: https://webrtc-codereview.appspot.com/566016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2345 4adac7df-926f-26a2-2b94-8c16560cd09d
