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
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
Note that,
(1) The main change is a new function PitchFilter() replacing a couple of common code blocks. Next step will be the assembly coding of this function in ARM.
(2) Resulted code is not bit exact with the original. The only reason is replacing two saturation blocks (lines 197 and 208) for the case of "type == 2" with the general case (line 147 and 159). The change makes the code more consistent, and I think the original code might just be a bug. I raised the issue in an email to Turaj and Bjorn last week.
Listening test might be needed. I will send the resulted files to Turaj for this purpose.
(3) I used Astyle to make the code more stylish, but didn't try extra effort to correct all the code style details. Local code style consistency was considered for new code. So this is not a full and final refactor project (will leave that to future refactoring).
Review URL: https://webrtc-codereview.appspot.com/573009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2344 4adac7df-926f-26a2-2b94-8c16560cd09d
Resolve coverity warnings.
14305.
The warning is not really valid. The 'decode' function should be called with a 'mode' variable, where inside the function it is assumed that mode is either zero or one. If mode is taking other values some varibles are used uninitialized. However, this is an internal function and it is always called with either ZERO or ONE. Therefore, the code operates correctly. I made small changes as I beleive it is a bit nicer way.
In ACM:
- Conditions on 'mode' is changed.
Tested with trybots.
BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/564014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2297 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
This CL allows NetEQ to do expand at startup, to make master and slave always go in sync. Before it could happen that master did merge, while slave performed an expand, leading to sync-problems between the channels.
Updating DEPS for new reference files for unittest.
BUG=410
TEST=neteq_unittests
Review URL: https://webrtc-codereview.appspot.com/487005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2055 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
