- 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
In this CL I have rewritten the way we handle stereo packets in VoE.
Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side).
While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere.
The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed in some of the tests, but I can explain offline.
Reviewers:
Björn - /src/modules/audio_coding
Patrik - /src/modules/rtp_rtcp
Patrik -/src/modules/utility
Henrik A - /src/voice_engine
BUG=410
TEST=voe_cmd_test
Review URL: https://webrtc-codereview.appspot.com/473003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
was incompatibile with the RTP dumps, which want to write both binary
and text data. Writing text data to a file open as "b" isn't actually
an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.
TEST=vie_auto_test
Review URL: http://webrtc-codereview.appspot.com/317005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
General Notes:
1. In general, API structure was not modified and is based on VPLIB.
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length).
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
