Commit Graph

10 Commits

Author SHA1 Message Date
46ad5426b0 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.

.../webrtc/base/atomicops.h:71:8: note:   no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'

Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1505053002

Cr-Commit-Position: refs/heads/master@{#10922}
2015-12-07 22:29:21 +00:00
84f0970d10 Reland of "Create rtc::AtomicInt POD struct."
Relands https://codereview.webrtc.org/1420043008/ with brace initializers
instead of constructors hoping that they won't introduce static
initializers.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1498953002 .

Cr-Commit-Position: refs/heads/master@{#10920}
2015-12-07 22:07:11 +00:00
3c12f4dadb Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
Reason for revert:
Caused static initializers.

BUG=chromium:556866
TBR=tommi@webrtc.org

Original issue's description:
> Create rtc::AtomicInt POD struct.
>
> Prevents accidental non-atomic reads, increments and stores since
> "volatile int" doesn't enforce atomic usage.
>
> BUG=
> R=kwiberg@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/b27f590ece487819c3d1fda400315e582fb975b6
> Cr-Commit-Position: refs/heads/master@{#10657}

TBR=kwiberg@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1453093002

Cr-Commit-Position: refs/heads/master@{#10669}
2015-11-17 11:21:07 +00:00
b27f590ece Create rtc::AtomicInt POD struct.
Prevents accidental non-atomic reads, increments and stores since
"volatile int" doesn't enforce atomic usage.

BUG=
R=kwiberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1420043008

Cr-Commit-Position: refs/heads/master@{#10657}
2015-11-16 19:03:06 +00:00
0b9e29c87d Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 10:12:32 +00:00
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
233bd87d45 Add RemoteEstimatorProxy for capturing receive times
For use when send-side bandwidth estimation is enabled.

Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1290813008

Cr-Commit-Position: refs/heads/master@{#9898}
2015-09-08 20:25:20 +00:00
867fb5224e Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.

BUG=webrtc:4311

Review URL: https://codereview.webrtc.org/1247293002

Cr-Commit-Position: refs/heads/master@{#9670}
2015-08-03 11:38:48 +00:00
e590416722 Moving the pacer and the pacer thread to ChannelGroup.
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}
2015-03-26 10:11:22 +00:00