Commit Graph

19 Commits

Author SHA1 Message Date
3cf8ece954 Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
Reason for revert:
This CL probably broke Chromium FYI.

Original issue's description:
> Stop caching supported codecs in WebRtcVideoEngine2
>
> We currently cache the result of GetSupportedCodecs in a member variable
> |video_codecs_| in WebRtcVideoEngine2. This means we need to keep
> |video_codecs_| and the result of GetSupportedCodecs in sync, which is
> error prone. It's simpler to just call GetSupportedCodecs when we need
> it, and we actually end up making fewer calls, so it's faster as well.
> This CL also returns all std::vectors by-value instead of by-ref. Move
> semantic together with in-place filtering of codecs actually end up with
> fewer copies, and it's also simpler to not return references.
>
> BUG=webrtc:6337
>
> Committed: https://crrev.com/9f71ec5a3e3175751f4475b126cfda89767363f2
> Cr-Commit-Position: refs/heads/master@{#15007}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2489173004
Cr-Commit-Position: refs/heads/master@{#15014}
2016-11-10 11:36:57 +00:00
9f71ec5a3e Stop caching supported codecs in WebRtcVideoEngine2
We currently cache the result of GetSupportedCodecs in a member variable
|video_codecs_| in WebRtcVideoEngine2. This means we need to keep
|video_codecs_| and the result of GetSupportedCodecs in sync, which is
error prone. It's simpler to just call GetSupportedCodecs when we need
it, and we actually end up making fewer calls, so it's faster as well.
This CL also returns all std::vectors by-value instead of by-ref. Move
semantic together with in-place filtering of codecs actually end up with
fewer copies, and it's also simpler to not return references.

BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2492473002
Cr-Commit-Position: refs/heads/master@{#15007}
2016-11-10 07:45:20 +00:00
cb56065c62 Add support for GCM cipher suites from RFC 7714.
GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".

If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).

BUG=webrtc:5222, 628400

Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
2016-08-04 12:20:38 +00:00
14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +00:00
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
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}
2016-06-30 07:59:49 +00:00
1895526c61 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

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
05b9803c8e Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2059403002
Cr-Commit-Position: refs/heads/master@{#13135}
2016-06-14 15:59:54 +00:00
dedfd28a52 Support for two audio codec lists down into WebRtcVoiceEngine.
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.

This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
2016-06-14 14:12:46 +00:00
6c87a67b63 Do not create a temporary transport channel when using max-bundle
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.

This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.

BUG=webrtc:5856

Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
2016-05-18 00:49:58 +00:00
c1513ee1a3 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API.
The caller can set a negative or zero file size to avoid using a limit.
BUG=

Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
2016-05-13 15:30:44 +00:00
33b01f2162 Adds network thread to rtc::BaseChannel
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.

BUG=webrtc:5645
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1903393004 .

Cr-Commit-Position: refs/heads/master@{#12690}
2016-05-11 17:55:41 +00:00
3102294fc0 Replace scoped_ptr with unique_ptr in webrtc/pc/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1783263002

Cr-Commit-Position: refs/heads/master@{#11961}
2016-03-11 22:18:26 +00:00
c11b184837 Remove CaptureManager and related calls in ChannelManager.
Removed unused screencast APIs.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1757843003

Cr-Commit-Position: refs/heads/master@{#11896}
2016-03-08 01:35:46 +00:00
Per
fb45d170c0 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/)
I readded virtual bool Pause(bool paused) for now with a dummy implementation since Chrome remoting override this method.

Original cl description:

Removed unused cricket::VideoCapturer methods:

void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
bool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()

This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.

There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.

BUG=webrtc:5426
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1744153002 .

Cr-Commit-Position: refs/heads/master@{#11809}
2016-02-29 11:07:45 +00:00
74622e0613 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ )
Reason for revert:
Breaks remoting::protocol::WebrtcVideoCapturerAdapter::Pause'

See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3689/steps/compile/logs/stdio

Original issue's description:
> Removed unused cricket::VideoCapturer methods:
>
> void UpdateAspectRatio(int ratio_w, int ratio_h);
> void ClearAspectRatio();
> ool Pause(bool paused);
> Restart(const VideoFormat& capture_format);
> MuteToBlackThenPause(bool muted);
> IsMuted() const
> set_square_pixel_aspect_ratio
> bool square_pixel_aspect_ratio()
>
> This cl also remove the use of messages and posting of state change.
> Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
> It does not add restrictions on what thread frames are delivered on though.
>
> There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/e9c0cdff2dad2553b6ff6820c0c7429cb2854861
> Cr-Commit-Position: refs/heads/master@{#11773}

TBR=magjed@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1740963002

Cr-Commit-Position: refs/heads/master@{#11777}
2016-02-26 10:54:43 +00:00
e9c0cdff2d Removed unused cricket::VideoCapturer methods:
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
ool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()

This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.

There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1733673002

Cr-Commit-Position: refs/heads/master@{#11773}
2016-02-26 07:36:22 +00:00
Per
a50924115e This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting.
BUG=webrtc:5426
R=pthatcher@webrtc.org
TBR=pthatcher@webrtc.org for moved channelmanager....

Review URL: https://codereview.webrtc.org/1689923002 .

Cr-Commit-Position: refs/heads/master@{#11597}
2016-02-12 12:31:08 +00:00
65c7f67f09 Fix license headers in webrtc/pc
This was not done in https://codereview.webrtc.org/1691463002/
in order to preserve Git history when moving the files.

BUG=webrtc:5419
TBR=pthatcher@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1693773002

Cr-Commit-Position: refs/heads/master@{#11593}
2016-02-12 08:05:07 +00:00
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00