- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
When https://crbug.com/webrtc/8734 is resolved this setup should be
valid and CreateOffer() and SetLocalDescription() should work, but
currently it doesn't. It probably fails because both senders are
assigned the same ID (the track ID).
EXPECT-ing the current behavior with a TODO referencing the bug.
Bug: webrtc:8734
Change-Id: If2a9cc9b0be12c39def83b0e219e1ca82dbd7d65
Reviewed-on: https://webrtc-review.googlesource.com/39041
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21654}
This version of stream stats will iterate over senders and
receivers and note which streams they think they know about,
rather than iterating over streams.
This means that streams mentioned in AddTrack() are also
included, and that only tracks actually attached are included
for those streams.
Bug: webrtc:8616
Change-Id: I4e704b1a47a152812f23a448cf1a6bc3af1ffafa
Reviewed-on: https://webrtc-review.googlesource.com/39262
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21609}
Similar to the change for RtpReceivers, this removes the BaseChannel
methods that would just proxy calls to the MediaChannel and instead
gives the MediaChannel directly to the RtpSenders to make the calls
directly.
Bug: webrtc:8587
Change-Id: Ibab98d75ff1641e902281ad9e31ffdad36caff35
Reviewed-on: https://webrtc-review.googlesource.com/38983
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21608}
This is a reland of e357a4dd4e3b015f8281813f246de793589bd537
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org
Bug: webrtc:8673
Change-Id: I610302efc5393919569b77e3b59aa3384a9b88a5
Reviewed-on: https://webrtc-review.googlesource.com/38842
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21589}
This reverts commit e357a4dd4e3b015f8281813f246de793589bd537.
Reason for revert: Looks like it's breaking some downstream projects.
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
Change-Id: I621c10236c02be01d82f4660168f0323b85e24af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8673
Reviewed-on: https://webrtc-review.googlesource.com/38681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21586}
This generates stats IDs for Track stats (which
represents stats on the attachment of a track to
a PeerConnection) from being SSRC-based to being
based on an ID that is allocated when connecting the
track to the PC.
This is a prerequisite to generating stats before
the PeerConnection is connected.
Bug: webrtc:8673
Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
Reviewed-on: https://webrtc-review.googlesource.com/38360
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21582}
When Unified Plan semantics are set, PeerConnection will fire OnAddTrack
according to the WebRTC spec. OnRemoveTrack will never be fired and will
be deprecated in the future.
Bug: webrtc:7600
Change-Id: Idfaada65b795b5fb9fe4844cff036d52ea90da17
Reviewed-on: https://webrtc-review.googlesource.com/38122
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21564}
This change corrects PeerConnection behavior under Unified
Plan semantics to:
- Set the RtpSender id to be the track ID if created with AddTrack.
- Put the RtpSender id in the SDP as part of the MSID.
- Set the RtpReceiver id to be the track part of the MSID
when created via SetRemoteDescription.
Also, the RtpSender constructors have been simplified to defer
mutable state (in this case, setting BaseChannels) to method calls.
Bug: webrtc:8721
Change-Id: Idc80965e2df7a803b8bbeec1d96de9ad95391cce
Reviewed-on: https://webrtc-review.googlesource.com/38480
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21563}
Currently, the RtpReceivers take a BaseChannel which is (mostly)
just used for proxying calls to the MediaChannel. This change
removes the extra layer and moves the proxying logic to RtpReceiver.
Bug: webrtc:8587
Change-Id: I01b0e3d57b4629e43d9d148cc94d6dd2941d320e
Reviewed-on: https://webrtc-review.googlesource.com/38120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21562}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.
Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
This is a reland of d2b912aed132c751919ed286439fb39bbd714dda
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Bug: webrtc:8630
Change-Id: I7429d9e270c9ecb4dfaf6aef85d3055c47658631
Reviewed-on: https://webrtc-review.googlesource.com/35600
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21524}
AddTrack is just a legacy wrapper for the new AddTrack method, so
calling the new AddTrack method should do everything that the old one
does.
Bug: None
Change-Id: I272a9e9584c470d54243377c1307b786f41c660d
Reviewed-on: https://webrtc-review.googlesource.com/37546
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21508}
Bug: webrtc:8714
Change-Id: I107b7d5908cf6efe7f114522959da4503450d7fc
Reviewed-on: https://webrtc-review.googlesource.com/37724
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21506}
VideoSourceInterface is clearly an integral part of
mediastreaminterface.h already, so moving that interface to api makes
sense. This also resolves a circular dependency in call/.
Bug: webrc:6828
Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c
Reviewed-on: https://webrtc-review.googlesource.com/37500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21498}
This is the old-style-stats equivalent of CL 34360.
Bug: webrtc:8616
Change-Id: I12573eb305a8f1ecf8134b87ab14e33eaec5ba22
Reviewed-on: https://webrtc-review.googlesource.com/37080
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21497}
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
This was causing ICE pings to continue going out on PeerConnections
that use DataChannels, even after closing the PeerConnection.
This CL adds a two-line fix, and an integration test that will catch
this and similar issues.
Bug: webrtc:7655
Change-Id: I589a2a1aaf6433c1d65be69a1267e1b52a33534b
Reviewed-on: https://webrtc-review.googlesource.com/37145
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21488}
This will allow stats to be generated when AddTrack() is used.
It also exposes a ClearStatsCache() call on the PC to allow enforcement
of cache lifetime restrictions.
Bug: webrtc:8616
Change-Id: If47b967ce9e40fa768303e6f5f54fe74db2cc7a4
Reviewed-on: https://webrtc-review.googlesource.com/34360
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21468}
This change adds support to PeerConnection's CreateOffer/
CreateAnswer/SetLocalDescription/SetRemoteDescription for
Unified Plan SDP mapping to/from RtpTransceivers. This behavior
is enabled using the kUnifiedPlan SDP semantics in the
PeerConnection configuration.
Bug: webrtc:7600
Change-Id: I4b44f5d3690887d387bf9c47eac00db8ec974571
Reviewed-on: https://webrtc-review.googlesource.com/28341
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21442}
The whole memory block is filled with "memcpy" afterwards.
Bug: None
Change-Id: Ieb7fe2513184a6abbf6c61f3582590f4bcd16f1f
Reviewed-on: https://webrtc-review.googlesource.com/35980
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21420}
This changes all internal code to use the media_description() helper
for ContentInfo along with the as_audio, as_video, and as_data casting
methods introduced in a previous CL. Reduces the total number of
pointer static_casts in pc/ from 351 to 122.
Bug: webrtc:8620
Change-Id: I996f49b55f1501c758a9e5223e30539a9f8d4eac
Reviewed-on: https://webrtc-review.googlesource.com/35921
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21419}
This makes the following changes:
- Replaces ContentDescription with its only subclass,
MediaContentDescription
- Adds helpers to cast a MediaContentDescription to its
audio, video, and data subclasses.
- Changes ContentInfo.type to a new enum, MediaProtocolType.
Bug: webrtc:8620
Change-Id: I5eb0811cb16a51b0b9d73ecc4fe8edc7037f1aed
Reviewed-on: https://webrtc-review.googlesource.com/35100
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21401}
This reverts commit d2b912aed132c751919ed286439fb39bbd714dda.
Reason for revert: broke internal tests
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,pthatcher@webrtc.org,shampson@webrtc.org
Change-Id: If82810072e21818ae452a0fc3f984d44e5dac70c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8630
Reviewed-on: https://webrtc-review.googlesource.com/35540
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21399}
I followed the wiring path for the max bitrate.
Doc:
https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
Bug: webrtc:8630
Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
Reviewed-on: https://webrtc-review.googlesource.com/30380
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21397}
Bug: webrtc:8620
Change-Id: I9b0b6d8dc9bda366e925dda9a5b92fc4e3fd9f43
Reviewed-on: https://webrtc-review.googlesource.com/35003
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21378}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This is a reland of 4770fd935ac92400487bddd3b755753572e6d692
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
Bug: webrtc:8636
Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8
Reviewed-on: https://webrtc-review.googlesource.com/34586
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21371}
The stats collectors would only ever call this on the signaling
thread, so they might as well just ask the voice/video channel
their transport_name directly.
Bug: None
Change-Id: I8dd36210ff22d222b45b5f5e22c253f601cdc79e
Reviewed-on: https://webrtc-review.googlesource.com/34581
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21339}
This reverts commit 4770fd935ac92400487bddd3b755753572e6d692.
Reason for revert: breaks downstream projects
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: Ia72c6d7f185a95b21fd0aec90e7fdc00cb1fb423
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8636
Reviewed-on: https://webrtc-review.googlesource.com/34600
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21335}
The JsepTransport class is moved to pc/ and the utility methods and
enums are moved to where they are used.
With JsepTransport moved to pc/, JsepTransport can depend on objects in
pc/ including RtpTranport, SrtpTransport etc.
Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
Bug: webrtc:8636
Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
Reviewed-on: https://webrtc-review.googlesource.com/33701
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21333}
Previously, the RTCStatsCollector needed to ask the voice/video
channel for its transport name in order to generate transport
level stats. That would happen on the networking thread which was
unsafe because the voice/video channel could have disappeared in
the duration of the asynchronous thread hop from the signaling
thread to the networking thread. This changes the networking stats
code to check a saved map that tracks the transport name for each
voice/video channel.
Bug: None
Change-Id: I1f03ba8c0526eaa4419f660f18b8b9da62c3f932
Reviewed-on: https://webrtc-review.googlesource.com/33660
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21332}
