4 Commits

Author SHA1 Message Date
Taylor Brandstetter
0cd086b70e Adding codecs to the RtpParameters returned by an RtpSender.
Contains every field except for sdpFmtpLine.
Setting a reordered list of codecs is not yet supported.

R=glaznev@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1885473004 .

Cr-Commit-Position: refs/heads/master@{#12453}
2016-04-20 23:23:22 +00:00
deadbeef
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
kjellander
1afca73055 Change to WebRTC license in webrtc/media
This was decided to be done in a separate CL from the move
that took place in https://codereview.webrtc.org/1587193006/

BUG=webrtc:5420
NOTRY=True
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1676923002

Cr-Commit-Position: refs/heads/master@{#11520}
2016-02-08 04:46:50 +00:00
kjellander
a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00