We now only set the volume when starting playout if the user has called
SetSpeakerVolume while we weren't playing. This now also ensures it will
actually get set to what the user requested rather than being overridden
by a default value.
Add tests to voe_auto_test.
BUG=6140661
TEST=voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/566006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2188 4adac7df-926f-26a2-2b94-8c16560cd09d
Description:
This CL will temporally define WEBRTC_SVNREVISION as "n/a" because it
could break Chromium if svn is not installed.
The long term solution is a have a script that could deal with it, and
have it support git-svn which is used by most developers.
BUG=496
TEST=build on Linux
Review URL: https://webrtc-codereview.appspot.com/569007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2187 4adac7df-926f-26a2-2b94-8c16560cd09d
I have uploaded this patch for your review. I have done an extensive test to be sure that removing of tables does not create any problem.
The test file, is called test_lpc.c which requires a hack to standard iSAC. The test computes LPC coefficients, then encodes and decodes with old and new (size-reduced) tables. It compares the results is all steps. I have ran the test over large set of files, more then 51 hours of audio, and there was no error.
I tried to do no formatting so the review to be easier, but I know it can be a tricky CL. Hopefully, the test file helps you to be more confident on the CL.
Thanks,... Turaj
In this change list the LPC tables associated with mode 1 & 2 are remoded, and necessary cahnges are made to other files.
The only model allowed is model number 0. Therefore, this CL breaks compatibility with iSAC released prior to 2.4.3. To avoid changing the bit-stream, we still keep the model number in the bit-stream.
entropy_coding.c is cleaned up, especially encoding of LAR had KLT transform of LPC gains which are removed now.
Review URL: https://webrtc-codereview.appspot.com/548004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2186 4adac7df-926f-26a2-2b94-8c16560cd09d
Descritption:
This CL addresses two issues in android surface view render,
1. Uninitlized class members _javaByteBufferObj and _directBuffer which
could cause crash.
2. Using ConvertI420ToRGB565. We should use high level libyuv apis
to help libyuv maintainer.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/566005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2185 4adac7df-926f-26a2-2b94-8c16560cd09d
I'm not sure what is the right thing to do here. That would probably be to call Release() the right amount of times and ensuring that the last call returns 0 (e.g. all references have been released), but I'll leave it up to the CL author to investigate that.
TBR=elham@webrtc.org
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2183 4adac7df-926f-26a2-2b94-8c16560cd09d
Added a test that injects 3 packets and then times out.
Rewrote the packet construction in test to use a builder format.
This makes tests a lot more readable.
Odd behaviour of timeout found; documented as test.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/553004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2161 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the first part of a plan:
- Get basic unit tests for rtcp_receiver.
- Get an unit test for some code inside rtcp_receiver
that touches the TMMBRSet class in hard-to-decipher ways
(rtcp_receiver_help, GetTMMBRSet function, use of memmove()).
- Refactor the TMMBRSet class.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/547005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2159 4adac7df-926f-26a2-2b94-8c16560cd09d
The reason is that this will cause a crash:
TbInterfaces foo = TbInterfaces("blah");
It relies on the generated copy constructor (or assignment operator), which copies the
pointer values from a temporary object. After |foo| in this case has been initialized
with values from the temporary object, the temp goes out of scope and is deleted.
The result is that |foo| has been initialized with pointers do a deleted object.
Also fixing expectations for the return value of VoE Release() calls after I checked
in my change that makes the VoiceEngine per-object ref counted and not per-interface.
Review URL: https://webrtc-codereview.appspot.com/509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2128 4adac7df-926f-26a2-2b94-8c16560cd09d
not per interface. This simplifies things a bit, reduces code and makes it
possible to implement reference counting (if we ever want) without the
static Delete() method. (Reference counted objects are traditionally
implicitly deleted via the last Release())
* Since the reference counting code is now simpler, there's no need for the
RefCount class so I'm removing it.
* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
method. The justification is that it's no longer used and the reason it was there
in the first place was to avoid bugs in third party code, so it's an indication that
something is wrong elsewhere.
* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.
* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.
* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)
BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
