Commit Graph

12 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
267527459b Remove cricket::VideoCodec with, height and framerate properties
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.

This cl require some upstream projects to change first.

BUG=webrtc:5332

Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
2016-10-24 08:21:24 +00:00
11a9cbfa50 Refactoring: move ownership of RtcEventLog from Call to PeerConnection
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.

This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).

BUG=webrtc:6393

Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
2016-10-07 18:53:15 +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
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
7f216b71aa Renames TransportController worker_thread to network_thread.
function suffix '_w' changes to '_n'

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

Review URL: https://codereview.webrtc.org/1895813003 .

Cr-Commit-Position: refs/heads/master@{#12694}
2016-05-12 07:20:43 +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
67cf2c1294 Removing preference field from cricket::Codec.
This field only existed as an implementation detail for getting the
codecs sorted, so it doesn't need to be in the public interface.
It cluttered the code and undesirably affected codec comparisons,
causing the video encoder to be reconfigured if a codec's preference
changed but nothing else did.

BUG=webrtc:5690

Review URL: https://codereview.webrtc.org/1845673002

Cr-Commit-Position: refs/heads/master@{#12349}
2016-04-13 17:07:24 +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
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