This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.
I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.
Next step is to add an API to choose application mode.
BUG=issue1239
Review URL: https://webrtc-codereview.appspot.com/1007006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.
In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.
BUG=
TEST=vie/voe_auto_test, trybots
Review URL: https://webrtc-codereview.appspot.com/1001006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat
TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1003005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
The following targets have been merged into audio_coding_unittests:
* cng_unittests
* g711_unittests
* g722_unittests
* isacfix_unittests
* pcm16b_unittests
Some of them were empty and were created with the assumption they were
needed in order to get code coverage (which was actually not needed).
The following test has been removed since it was empty:
* audio_conference_mixer_unittests
BUG=none
TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated)
Review URL: https://webrtc-codereview.appspot.com/971008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
- Pad packets (empty) were often NACKed even though they were received.
- Padding only frames (empty) were didn't properly update the decoding state,
and would therefore be NACKed even though they were received.
This is a recommit of r3183. Extensive testing suggest that this may have been caused by virtual machine flakiness.
TBR=mflodman@webrtc.org
BUG=1150
Review URL: https://webrtc-codereview.appspot.com/971011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3200 4adac7df-926f-26a2-2b94-8c16560cd09d
For Visual Studio versions older than 2012, we are using a
separate reference output file for windows. (All other platforms
share the same generic reference file.) In VS 2012, the output
matches the generic reference, and not the platform-specific one.
Since, the ResourcePath() method cannot change behavior depending
on compiler version, this fix will short-cut ResourcePath() for
VS 2012 or newer (_MSC_VER >= 1700).
Also made NetEqDecodingTest.TestBitExactnes stop on the first diff.
Once there is a difference, the output is no longer bit-exact, and
the test should be declared a failure.
BUG=
TEST=neteq_unittests on VS2012, try bots
Review URL: https://webrtc-codereview.appspot.com/966028
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3199 4adac7df-926f-26a2-2b94-8c16560cd09d
