17 Commits

Author SHA1 Message Date
pbos
1ba8d39a9c Remove webrtc/stream.h and unutilized inheritance.
Removes inheritance and a virtual call. Also removes a root header that
would have needed to be moved into a subdirectory otherwise to prevent
circular dependencies.

BUG=webrtc:4243
R=kjellander@webrtc.org, solenberg@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1924793002
Cr-Commit-Position: refs/heads/master@{#12586}
2016-05-02 03:18:36 +00:00
kwiberg
4485ffb58d #include "webrtc/base/constructormagic.h" where appropriate
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.

Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1917043005

Cr-Commit-Position: refs/heads/master@{#12509}
2016-04-26 15:14:48 +00:00
kwiberg
fffa42b57e Replace scoped_ptr with unique_ptr in webrtc/audio/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1706183002

Cr-Commit-Position: refs/heads/master@{#11723}
2016-02-23 18:46:39 +00:00
deadbeef
2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13 20:00:29 +00:00
deadbeef
e591f9377f Storing raw audio sink for default audio track.
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}
2016-01-13 00:45:33 +00:00
Stefan Holmer
3842c5c7f7 Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12 12:55:11 +00:00
Tommi
f888bb58da Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1505253004 .

Cr-Commit-Position: refs/heads/master@{#10995}
2015-12-12 00:37:14 +00:00
solenberg
13725089ef Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1459083007

Cr-Commit-Position: refs/heads/master@{#10788}
2015-11-25 16:16:57 +00:00
solenberg
7add058439 Move some receive stream configuration into webrtc::AudioReceiveStream.
Simplify creation of VoE channels and Call streams in WVoMC.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1454073002

Cr-Commit-Position: refs/heads/master@{#10731}
2015-11-20 17:59:40 +00:00
solenberg
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 23:34:58 +00:00
Henrik Kjellander
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
solenberg
85a0496b8c Implement AudioSendStream::GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1414743004

Cr-Commit-Position: refs/heads/master@{#10424}
2015-10-27 10:35:30 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
2015-10-22 08:49:39 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
2015-10-20 13:01:55 +00:00
pbos
a2f30deea3 Log Call {audio, video} stream deletions.
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1400333002

Cr-Commit-Position: refs/heads/master@{#10286}
2015-10-15 12:22:21 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00