adding 30% drift to media generator (e.g. audio frame generated every 7ms instead of promised 10ms) works fine
adding 2% drift to video ntp-timestamp-stamper makes A/V sync fail.
BUG=webrtc:5504
R=pbos@webrtc.org,stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1674413004
Cr-Commit-Position: refs/heads/master@{#11556}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
This reverts commit cf3c83e76c273309558c86fda915410f65b7a899.
Reverting EventWrapper split did not fix the issue, re-landing.
BUG=chromium:470013
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49629004
Cr-Commit-Position: refs/heads/master@{#8946}
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}
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.
This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.
R=mflodman@webrtc.org, mflodman
BUG=
Review URL: https://webrtc-codereview.appspot.com/48599004
Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
