Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.
The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.
TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).
I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).
I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.
Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
We encounter a sample-underrun if NetEq is initialized with a sampling rate fs =16000 and receive Opus packets with frame-size less than 5 ms. The reason is as follows.
Let say NetEq buffer has 4 packets of Opus each of size 2.5ms this means that internally timestamp of packets incremented by 80 (internally Opus treated as 32 kHz codec). Given the initial sampling rate of NetEq, at the first time that it wants to fetch packets, it targets to fetch 160 samples. Therefore, it will only extracts 2 packets. Decoding these packets give us exactly 160 samples (5 ms at 32 kHz), however, upon decoding the first packet the internal sampling rate will be updated to 32 kHz. So it is expected that sync buffer to deliver 320 samples while it does only have 160 samples (or maybe few more as it starts with some zeros). And we encounter and under-run.
Even if we ignore the under-run "assert(sync_buffer_->FutureLength() >= expand_->overlap_length())" (neteq_impl.cc::811) is trigered. I'm not sure what happens if we remove this assert perhaps NetEq will work fine in subsequent calls. However the first under-run is blocking ACM2 test to pass.
Here I have a solution to update sample rate as soon as a packet is inserted, if required. It not a very efficient approach as we do the same reset in NetEqImpl::Decode().
It is a bit tricky to reproduce this because the TOT ACM tests do not run ACM2. In https://webrtc-codereview.appspot.com/2192005/ I have a patch to run both ACMs. To reproduce the problem, one can patch that CL and run
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.TestOpus
Note that we would not encounter any problem if NetEq4 is initiated with 32000 Hz sampling rate. You can test this by setting |kNeteqInitSampleRateHz| to 32000 in webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2306004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4896 4adac7df-926f-26a2-2b94-8c16560cd09d
If there is audio in a codec's audio buffer and sample-rate or number of channels change the audio buffer has to reset. Otherwise, the amount of audio in the buffer is misinterpreted any syncronization between 10ms audio blocks and their associated timestamps is lost.
For instance, assume changing from stereo to mono when there is 10ms stereo in the buffer. The "new" codec will interpret this as 20 ms audio, therefore, 2 blocks of 10 ms, but there is only one timestamp. This will results in ACMGenericCodec::in_timestamp_ix_write_ updated to a negative number after an encode is performed.
The drawback with this solution is that if packet-size of the codec is changed then audio buffer is reset wich is not necessary. We accept this as it is a rare case in practice that clients of ACM re-register send codecs to change packet-size.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4887 4adac7df-926f-26a2-2b94-8c16560cd09d
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.
BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2177004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4807 4adac7df-926f-26a2-2b94-8c16560cd09d
First patch set is the same as patch set 3 of http://review.webrtc.org/2237004/
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these
APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these
APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF
detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded.
Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4785 4adac7df-926f-26a2-2b94-8c16560cd09d
-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded. Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2237004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4772 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio
> Enable SetInitialPlayoutDelay on Android.
>
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
>
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2144004TBR=dwkang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.
BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2144004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
