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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
3ba5b9e5c6d300738ee4eaef5b1d73b197755d97
/
media
/
base
/
codec.cc
2c9ebef
Use Abseil container algorithms in media/
by Steve Anton
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
039743e
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
by Niels Möller
· 6 years ago
6e8e299
Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
by Oleh Prypin
· 6 years ago
80cd25b
Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
3c7d599
Replace _stricmp with absl::EqualsIgnoreCase
by Niels Möller
· 6 years ago
98badbc
Add VP9 profile negotiation to SDP
by Emircan Uysaler
· 7 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 7 years ago
b7d9d83
Implement RtpCodecParameters::parameters
by Florent Castelli
· 7 years ago
88c9956
Remove internal media/base/ stringstream usages
by Jonas Olsson
· 7 years ago
634a777
Add RRTR parameter to media engine and pass it to video receive stream
by Ilya Nikolaevskiy
· 7 years ago
a680a6a
Enable and fix chromium clang warnings in sdk/android targets.
by Paulina Hensman
· 7 years ago
9c1fb1e
Consider packetization-mode when matching H264 codecs
by Steve Anton
· 7 years ago
523589d
Create common helper method for comparing video formats
by Magnus Jedvert
· 7 years ago
7880758
Optional: Use nullopt and implicit construction in /media
by Oskar Sundbom
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
024d897
Add conversion from webrtc::SdpVideoFormat to cricket::VideoCodec
by Magnus Jedvert
· 7 years ago
244ad80
Clean up some bad constructs in media/
by Magnus Jedvert
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/media/base/codec.cc]
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 8 years ago
e702b30
Adding C++ versions of currently spec'd "RtpParameters" structs.
by deadbeef
· 8 years ago
9aa9688
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
by hta
· 8 years ago
243a0a7
Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
by hta
· 8 years ago
e59647b
This approach passes packetization mode to the encoder as part of
by hta
· 8 years ago
0928a3c
Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ )
by magjed
· 8 years ago
10165ab
Unify VideoCodecType to/from string functionality
by magjed
· 8 years ago
0d0d753
Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ )
by magjed
· 8 years ago
aae7e7c
Split out target rtc_media_base from rtc_media
by magjed
· 8 years ago
f823ede
Negotiate H264 profiles in SDP
by magjed
· 8 years ago
3663c52
Provide move semantic for cricket::Codec and subclasses
by magjed
· 8 years ago
87d7d77
Add new codec for FlexFEC.
by brandtr
· 8 years ago
1e45cc6
Replace WebRtcVideoEncoderFactory::VideoCodec with cricket::VideoCodec
by magjed
· 8 years ago
06c8e1e
Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
by Magnus Jedvert
· 8 years ago
2675274
Remove cricket::VideoCodec with, height and framerate properties
by perkj
· 8 years ago
061ea0d
Remove VideoCodec resolution validation.
by Per
· 8 years ago
68979ab
H264 codec: Check profile-level-id when matching
by magjed
· 8 years ago
0cd086b
Adding codecs to the RtpParameters returned by an RtpSender.
by Taylor Brandstetter
· 9 years ago
67cf2c1
Removing `preference` field from `cricket::Codec`.
by deadbeef
· 9 years ago
1afca73
Change to WebRTC license in webrtc/media
by kjellander
· 9 years ago
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
[Renamed (99%) from talk/media/base/codec.cc]
ba4c0e4
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
by stefan
· 9 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 9 years ago
25702cb
Misc. small cleanups.
by pkasting
· 9 years ago
43edf0f
Require negotiation to send transport cc feedback over RTCP.
by stefan
· 9 years ago
e62202f
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
by Shao Changbin
· 10 years ago
d324546
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
by pkasting@chromium.org
· 10 years ago
cce874b
Fix libjingle_media_unittest codec comparison issue
by guoweis@webrtc.org
· 10 years ago
bc6961f
Make webrtc 50 KB smaller by not inlining Codec.
by guoweis@webrtc.org
· 10 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 11 years ago
ff1b1bf
When creating an answer, takes the codec preference from the offer.
by wu@webrtc.org
· 11 years ago
fbd1328
(Auto)update libjingle 69555283-> 69567902
by buildbot@webrtc.org
· 11 years ago
b5a22b1
Revert r6110 and r6109.
by pbos@webrtc.org
· 11 years ago
17911dc
(Auto)update libjingle 66798415-> 66813165
by buildbot@webrtc.org
· 11 years ago
0df2ea0
Rollback of r6108
by henrike@webrtc.org
· 11 years ago
a7f70a4
Initialize bitrates in ValidateCodecFormat.
by pbos@webrtc.org
· 11 years ago
d266a20
Initial wiring of new webrtc API in libjingle.
by pbos@webrtc.org
· 11 years ago
28e2075
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk
by henrike@webrtc.org
· 12 years ago