Commit Graph

8 Commits

Author SHA1 Message Date
87d7d77700 Add new codec for FlexFEC.
This CL does nothing except adding new strings and enums corresponding to
the new codec.

R=stefan@webrtc.org
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2470103002
Cr-Commit-Position: refs/heads/master@{#14943}
2016-11-07 11:04:03 +00:00
06c8e1eaa7 Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
Reason for revert:
Breaks H264 for external encoders in WebRTC as well as breaking H264 interop with e.g. Edge.

Original issue's description:
> H264 codec: Check profile-level-id when matching
>
> For the H264 video codec, comparing the codec name is not enough
> for determining a match. The profile-level-id must also match.
> This CL:
>  - Specializes the VideoCodec::Matches function with extra logic for
>    matching H264 codecs.
>  - Adds unittests for matching H264 video codecs.
>  - Removes the unnecessary CodecTest fixture class.
>
> BUG=webrtc:6337,chromium:645599
>
> Committed: https://crrev.com/68979ab7dd971ab6e983b23c83154ba05e183fb8
> Cr-Commit-Position: refs/heads/master@{#14546}

TBR=kthelgason@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6337,chromium:645599,webrtc:6552,webrtc:6402

Review URL: https://codereview.webrtc.org/2440123002 .

Cr-Commit-Position: refs/heads/master@{#14759}
2016-10-25 07:54:04 +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
9fa49759e5 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped.
- Rename the data codec payload types to end with "PlType" instead of "Id", for consistency.

BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2397413002
Cr-Commit-Position: refs/heads/master@{#14581}
2016-10-08 20:02:51 +00:00
68979ab7dd H264 codec: Check profile-level-id when matching
For the H264 video codec, comparing the codec name is not enough
for determining a match. The profile-level-id must also match.
This CL:
 - Specializes the VideoCodec::Matches function with extra logic for
   matching H264 codecs.
 - Adds unittests for matching H264 video codecs.
 - Removes the unnecessary CodecTest fixture class.

BUG=webrtc:6337,chromium:645599

Review-Url: https://codereview.webrtc.org/2347863003
Cr-Commit-Position: refs/heads/master@{#14546}
2016-10-06 09:15:56 +00:00
6f8d686d35 Remove use of RtpHeaderExtension and clean up
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension

The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.

Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.

BUG= webrtc:5895

Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
2016-05-26 18:25:04 +00:00
hta
a6b99448ee Generate FMTP parameters for the H.264 codec.
This CL generates FMTP parameters that allow H.264 interoperation
with Firefox for the default codec list.

BUG=chromium:591971

Review URL: https://codereview.webrtc.org/1880963002

Cr-Commit-Position: refs/heads/master@{#12333}
2016-04-12 17:29:20 +00:00
f475277547 Rename constants files in webrtc/{media,p2p}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.

To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}

This CL will require coordinating landing a roll in Chromium.

BUG=webrtc:4256
NOTRY=True

Review URL: https://codereview.webrtc.org/1750593002

Cr-Commit-Position: refs/heads/master@{#11842}
2016-03-02 13:42:35 +00:00