The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.
Two small but not quite trivial cleanups are included because they
blocked the state_ removal:
- AudioDecoderG722Stereo now inherits directly from AudioDecoder
instead of being a subclass of AudioDecoderG722.
- AudioDecoder now has a CngDecoderInstance member function, which
is implemented only by AudioDecoderCng. This replaces the previous
practice of calling AudioDecoder::state() and casting the result
to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
plainly visible in the AudioDecoder class declaration.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24169005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
Reason: Turns out this does not solve the problem as the buildbots
run into another error later on during collecting gtest output.
The problem is solved by excluding these bots from flakiness dashboard
data instead, in https://codereview.chromium.org/705913002/
> Fix memcheck and dr memory after flakiness dashboard deployment.
>
> I deployed buildbot configuration for uploading data to the
> flakiness dashboard but I didn't predict our Dr Memory and
> Memcheck bots would fail due to the new flag passed to the test.
> Adding the --gtest_output flag to the script will avoid the build
> to fail as a workaround.
>
> TBR=andrew@webrtc.org
> TESTED=Passing test run using:
> src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml
>
> BUG=
>
> Review URL: https://webrtc-codereview.appspot.com/28999004TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7643 4adac7df-926f-26a2-2b94-8c16560cd09d
I deployed buildbot configuration for uploading data to the
flakiness dashboard but I didn't predict our Dr Memory and
Memcheck bots would fail due to the new flag passed to the test.
Adding the --gtest_output flag to the script will avoid the build
to fail as a workaround.
TBR=andrew@webrtc.org
TESTED=Passing test run using:
src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml
BUG=
Review URL: https://webrtc-codereview.appspot.com/28999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7642 4adac7df-926f-26a2-2b94-8c16560cd09d
From r7014 the Android APK tests are designed to be
build from a standalone WebRTC checkout instead of a
Chromium checkout. Because of that, the special handling
for both cases can be removed.
I also don't think we need to use the
base::android::GetExternalStorageDirectory() method since
all devices has a symlink at /sdcard that points
to /storage/emulated/legacy on the Android device.
This essentially reverts the changes in
https://webrtc-codereview.appspot.com/1754005/
plus some minor changes.
BUG=webrtc:3741
TEST=Locally running test_support_unittests APK test on an
Android device using:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s test_support_unittests --verbose --isolate-file-path=webrtc/test/test_support_unittests.isolate
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7632 4adac7df-926f-26a2-2b94-8c16560cd09d
Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...
> Remove the state_ member from AudioDecoder
>
> The subclasses that need a state pointer should declare them---with
> the right type, not void*, to get rid of all those casts.
>
> Two small but not quite trivial cleanups are included because they
> blocked the state_ removal:
>
> - AudioDecoderG722Stereo now inherits directly from AudioDecoder
> instead of being a subclass of AudioDecoderG722.
>
> - AudioDecoder now has a CngDecoderInstance member function, which
> is implemented only by AudioDecoderCng. This replaces the previous
> practice of calling AudioDecoder::state() and casting the result
> to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
> plainly visible in the AudioDecoder class declaration.
>
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/24169005TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30879005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.
Two small but not quite trivial cleanups are included because they
blocked the state_ removal:
- AudioDecoderG722Stereo now inherits directly from AudioDecoder
instead of being a subclass of AudioDecoderG722.
- AudioDecoder now has a CngDecoderInstance member function, which
is implemented only by AudioDecoderCng. This replaces the previous
practice of calling AudioDecoder::state() and casting the result
to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
plainly visible in the AudioDecoder class declaration.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24169005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.
Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.
Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.
BUG=chrome:423985
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.
BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
