Instead just use MD5Context type directly. In C++ it is unnecessary to
alias the types using typedef, unline C (where if you don't you have to
spell out struct or enum infront of the user-type everytime you want to make a
variable).
So since WebRTC's base API is C++, it seems unnecessay to keep this
typedef around.
BUG=None
TEST=rtc_unittests --gtest_filter=Md5*
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46799004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8916}
Tested on Linux with the following command lines:
$ gn gen //out/Debug --args='is_debug=true target_cpu="x64" build_with_chromium=false'
$ ninja -C out/Debug frame_analyzer rgba_to_i420_converter
BUG=chromium:461019
TEST=see above
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42999004
Cr-Commit-Position: refs/heads/master@{#8914}
This reverts commit 9509fbfc301dd5412804ce5731afedc81480f2f8.
This is to debug a Chromium issue that WebRTC hangs if there is > 1 PeerConnection active in the browser on Win XP.
BUG=
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43019004
Cr-Commit-Position: refs/heads/master@{#8912}
Pass content_browsertests in Chromium. Performance test result (lower is
better):
C version: 100%
old intrinsics Neon version (with bug): 16.5%
new intrinsics Neon version: 18.0%
asm Neon version: 23.3%
BUG=4002
R=andrew@webrtc.org, jridges@masque.com
Change-Id: Ia0a96ac237216b635fc528f67d39319cdf246281
Review URL: https://webrtc-codereview.appspot.com/46739004
Cr-Commit-Position: refs/heads/master@{#8907}
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
RenderFrame should not modify the I420VideoFrame (and we don't).
This CL changes the declaration of RenderFrame from:
int32_t RenderFrame(const uint32_t streamId, I420VideoFrame& videoFrame)
to:
int32_t RenderFrame(const uint32_t streamId, const I420VideoFrame& videoFrame)
BUG=1128
R=mflodman@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46689005
Cr-Commit-Position: refs/heads/master@{#8902}
This CL avoids changing the mentioned callbacks during a call, to avoid
a potential deadlock when acquiring _sendCritSect and calling
_mediaOpt.SetTargetRates.
Moving the critsect revealed a race for the FEC parameters in RtpVideoSender, so the CL grew a bit to avoid this. I also cleaned up some code here at the same time, but tried to keep it at a minimum since this CL had already increased a lot in size.
BUG=769
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42939004
Cr-Commit-Position: refs/heads/master@{#8899}
We currently hit asserts in AECM where the output of WebRtcSpl_NormW16() on armv7 is incorrect.
I've verified that it outputs -17 for negative values. Internally that means that clz returns 0 after a two's complement operation on a int16_t.
There is a mismatch between the int16_t input and otherwise 32 bit assumptions. Explicitly casting to int32_t makes the two's complement do the correct thing.
The CL also extends the unit tests by running through a larger set of values.
BUG=4486
TESTED=locally on Android Nexus 7 and trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49549004
Cr-Commit-Position: refs/heads/master@{#8897}
This change essentially divides AudioCodingModuleImpl into two parts:
one is the code related to managing codecs, now moved into CodecManager,
and the other is what remains in AudioCodingModuleImpl.
This change also removes AudioCodingModuleImpl::InitializeSender. The
function was essentially no-op, since it was always called immediately
after construction.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51469004
Cr-Commit-Position: refs/heads/master@{#8893}
This is necessary unfortunately since there are a few places where DeRegisterModule does not reliably occur on the same thread.
BUG=4473
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42979004
Cr-Commit-Position: refs/heads/master@{#8891}
We want to crate the illusion that iSAC supports 48000 Hz decoding,
while in fact it outputs 32000 Hz. This is the iSAC fullband mode.
Currently this is (also) handled by higher layers, but in future
changes this will not be the case.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47809004
Cr-Commit-Position: refs/heads/master@{#8889}
All these tests crashed before r8811. These tests should've been with
that change but r8811 was pushed in before to make bots green.
BUG=1788, 1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48669004
Cr-Commit-Position: refs/heads/master@{#8881}
These are problematic when running with the default TSan
settings which has deadlock detection enabled.
Our bots still run with it disabled but we want to be
able to turn it back on, thus this is needed.
BUG=3911,4456
TESTED=
Successfully executed:
GYP_DEFINES="tsan=1 release_extra_cflags=-g use_allocator=none" webrtc/build/gyp_webrtc
ninja -C out/Release rtc_unittests
out/Release/rtc_unittests
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44899004
Cr-Commit-Position: refs/heads/master@{#8879}
Setting the member value output_will_be_muted_ in set_output_will_be_muted() was done before the lock.
This caused a data race.
BUG=4477
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44929004
Cr-Commit-Position: refs/heads/master@{#8877}
Also adds a framework for an AudioManager to be used by both sides (playout and recording).
This initial implementation only does very simple tasks like setting up the correct audio
mode (needed for correct volume behavior). Note that this CL is mainly about modifying
the volume. The added AudioManager is only a place holder for future work. I could have
done the same parts in the WebRtcAudioTrack class but feel that it is better to move these
parts to an AudioManager already at this stage.
The AudioManager supports Init() where actual audio changes are done (set audio mode etc.)
but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the
case here. Hence, the AM now serves as the center for getting audio parameters and then inject
these into playout and recording sides. Previously, both sides acquired their own parameters
and that is more error prone.
BUG=NONE
TEST=AudioDeviceTest
R=perkj@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45829004
Cr-Commit-Position: refs/heads/master@{#8875}
Change internal indexing of registered decoders. It makes sense because payload type is unique, while ACM codec ID may not be. This is a step towards allowing for addition of external decoders.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44869004
Cr-Commit-Position: refs/heads/master@{#8867}
These checks would help catching double-deletes, forgetting to destroy
streams and also catch if VideoEngine has held on to any stale
references.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42929004
Cr-Commit-Position: refs/heads/master@{#8866}
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.
This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.
The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/
BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699004
Cr-Commit-Position: refs/heads/master@{#8861}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3347, 3348, 3353
TESTED=locally on Linux for both fixed and floating point and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44799004
Cr-Commit-Position: refs/heads/master@{#8857}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3348, 3353
TESTED=locally on Linux for both fixed and floating point and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49499004
Cr-Commit-Position: refs/heads/master@{#8853}
