and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
The code previously allowed ipv6 addresses with less than eight sections even without all-zero sections being compacted by a ::.
BUG=webrtc:1028
Review-Url: https://codereview.webrtc.org/2606383003
Cr-Commit-Position: refs/heads/master@{#16108}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found).
The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF.
In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate.
This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2392883002
Cr-Commit-Position: refs/heads/master@{#15129}
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
Reason for revert:
This CL breaks internal dependencies.
Original issue's description:
> New statistics interface for APM
>
> This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
>
> BUG=webrtc:6525
>
> Committed: https://crrev.com/8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4
> Cr-Commit-Position: refs/heads/master@{#14810}
TBR=peah@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2456333002
Cr-Commit-Position: refs/heads/master@{#14814}
This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2433153003
Cr-Commit-Position: refs/heads/master@{#14810}
This CL deletes the old and not used video defines in
engine_configurations.h and pre-pends voice_ to indicate there are only
voice/audio defines left in the file.
BUG=none
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/2401673002 .
Cr-Commit-Position: refs/heads/master@{#14558}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
Reason for revert:
Breaks downstream code
Original issue's description:
> Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
>
> Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
>
> Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
>
> BUG=
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
> Cr-Commit-Position: refs/heads/master@{#14191}
TBR=kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2336123002
Cr-Commit-Position: refs/heads/master@{#14193}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Commit-Position: refs/heads/master@{#14191}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
They always fail, because RED isn't supported.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2055753002
Cr-Commit-Position: refs/heads/master@{#13743}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
This test currently takes 288 seconds to fail if output values are
wrong; there's no point to print the failure hundreds of times.
This change will exit the test early.
R=henrika@webrtc.org
BUG=623538
NOTRY=true
Review-Url: https://codereview.webrtc.org/2097363002
Cr-Commit-Position: refs/heads/master@{#13295}
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/2014973002 .
Cr-Commit-Position: refs/heads/master@{#12925}
Original issue's description:
> Adding a some checks and switching out a few assert for RTC_[D]CHECK.
> These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
>
> BUG=chromium:613482
> NOTRY=true
> (using notry due to offline android_arm64_rel bot)
>
> Committed: https://crrev.com/d36df89d40bde3c62ee5cbff841933e50b3c007b
> Cr-Commit-Position: refs/heads/master@{#12870}
TBR=henrik.lundin@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:613482
Review-Url: https://codereview.webrtc.org/2009253004
Cr-Commit-Position: refs/heads/master@{#12907}
Reason for revert:
Reverting temporarily. Need to fix tests downstream that pass invalid arguments.
Original issue's description:
> Adding a some checks and switching out a few assert for RTC_[D]CHECK.
> These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
>
> BUG=chromium:613482
> NOTRY=true
> (using notry due to offline android_arm64_rel bot)
>
> Committed: https://crrev.com/d36df89d40bde3c62ee5cbff841933e50b3c007b
> Cr-Commit-Position: refs/heads/master@{#12870}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:613482
Review-Url: https://codereview.webrtc.org/2006243002
Cr-Commit-Position: refs/heads/master@{#12874}
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
BUG=chromium:613482
NOTRY=true
(using notry due to offline android_arm64_rel bot)
Review-Url: https://codereview.webrtc.org/2007563002
Cr-Commit-Position: refs/heads/master@{#12870}
