Commit Graph

1474 Commits

Author SHA1 Message Date
2209b90449 Remove WEBRTC_TRACE.
Bug: webrtc:5118
Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d
Reviewed-on: https://webrtc-review.googlesource.com/5382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20114}
2017-10-03 13:20:48 +00:00
849b3aeb71 Move list of supported H264 codecs from InternalEncoderFactory to h264.h
This CL is a clean-up to prepare for adding more supported codecs for the internal H264 SW codec.

Bug: webrtc:8317
Change-Id: If483d05c01c40bbc81cbd1a6aad89961689714ef
Reviewed-on: https://webrtc-review.googlesource.com/4940
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20105}
2017-10-03 09:01:31 +00:00
d4404c232d Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
This reverts commit 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30.

Reason for revert: Breaks Chromium

Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
> 
> (Re-upload of https://codereview.webrtc.org/3020493002/)
> 
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}

TBR=solenberg@webrtc.org,henrika@webrtc.org

Change-Id: Iad03cafb7865f5a22394c3d4d1d3ff3e0fccd4ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4690, webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/5402
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20085}
2017-10-02 15:10:04 +00:00
34cdd2d402 Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
(Re-upload of https://codereview.webrtc.org/3020493002/)

Bug: webrtc:4690, webrtc:7306
Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
Reviewed-on: https://webrtc-review.googlesource.com/5360
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20083}
2017-10-02 15:01:20 +00:00
b0a0207838 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay

Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
2017-10-02 10:47:00 +00:00
bf35298996 Implement temporal layers checkers for vp8
All frames are checked against hard-coded dependency graph 
using new helper class. It's invoked in RTC_DCHECK(). Only 
DefaultTemporalLayers are fully implemented in this CL, checker 
for ScreenshareLayers is not doing anything for now.

Bug: none
Change-Id: I3ec017176d8c25f7572c8f161e52f2ebfac8220f
Reviewed-on: https://webrtc-review.googlesource.com/3740
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20066}
2017-10-02 09:14:59 +00:00
032f410ae0 Delete unneeded includes of pathutils.h
Bug: webrtc:6424
Change-Id: I73b2bc747c67d2fe2ad888dde9c2815a6d9aceaa
Reviewed-on: https://webrtc-review.googlesource.com/4760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20059}
2017-10-02 07:05:19 +00:00
024d8970a7 Add conversion from webrtc::SdpVideoFormat to cricket::VideoCodec
We will have to convert from webrtc::SdpVideoFormat to
cricket::VideoCodec in a couple of places until
cricket::WebRtcVideoEncoderFactory is gone. It will be convenient to
have the conversion logic in a common place.

Bug: webrtc:7925
Change-Id: Ie5e88599f28aeea647e936300c04f9071daffd53
Reviewed-on: https://webrtc-review.googlesource.com/4840
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20048}
2017-09-29 14:45:57 +00:00
d8970dbd42 Delete unneeded includes of fileutils.h
It is now used only by FileRotatingStream.

Bug: webrtc:6424
Change-Id: I216b20baadae836d24c39899efe4cb45c2935f41
Reviewed-on: https://webrtc-review.googlesource.com/4720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20040}
2017-09-29 12:39:09 +00:00
244ad80444 Clean up some bad constructs in media/
We currently suppress warnings for bad constructs in media/. Still, the
warnings are causing problems when trying to include header files from
this directory. This CL cleans up some of the bad constructs.

Bug: None
Change-Id: I808ad39eb23870d20fa5bb05429b50c9078543ae
Reviewed-on: https://webrtc-review.googlesource.com/4541
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20039}
2017-09-29 12:22:57 +00:00
9cf9f758fc Detach SequencedTaskChecker in MediaCodecVideoEncoder::Release.
If this is not done, the RTC_DCHECK_CALLED_SEQUENTIALLY might fire
if the encoder is used on a new VideoStreamEncoder. This happens
after VideoSendStream recreations due to changes in the SDP.

BUG=b/66590444

Change-Id: I086370526afbbe2ba629805f97f89e512ba3f472
Reviewed-on: https://webrtc-review.googlesource.com/4360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20020}
2017-09-28 15:15:21 +00:00
02e7a1981a Remove unnecessary video factory references in PeerConnectionFactory
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.

Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
2017-09-27 14:41:46 +00:00
fc3a2e3393 Remove the VoiceEngineObserver callback interface.
BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019513002
Cr-Commit-Position: refs/heads/master@{#19976}
2017-09-26 16:35:01 +00:00
3b3622fafc Delete member VideoReceiveStream::Config::Rtp::ulpfec.
Replaced with scalars ulpfec_payload_type and red_payload_type.

In particular, ulpfec.red_rtx_payload_type, which duplicated info in
rtx_associated_payload_types, is deleted. This is a followup to cl
https://codereview.webrtc.org/3012963002.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3019453002
Cr-Commit-Position: refs/heads/master@{#19965}
2017-09-26 09:49:21 +00:00
7cd28b9172 Set protected_by_flexfec flag properly in tests.
BUG=none

Review-Url: https://codereview.webrtc.org/3010003002
Cr-Commit-Position: refs/heads/master@{#19921}
2017-09-22 07:26:25 +00:00
4e2deab79c Only return stats for the most recent unsignaled audio stream.
The track-level stats are currently implemented in terms of the stream-
level stats. Which is a problem if multiple unsignaled streams map to the
same track (see bug for more details). This CL fixes the problem
partially, but only returning stats for one of the unsignaled streams.
A better solution would be to return stats for both streams, but update
the track-level stats independently somehow. But that would require more
extensive changes, and it's not yet clear how we want to do it.

BUG=webrtc:8158

Review-Url: https://codereview.webrtc.org/3008373002
Cr-Commit-Position: refs/heads/master@{#19907}
2017-09-20 20:56:21 +00:00
6dc2038d0d Remove VoECodec.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3019433002
Cr-Commit-Position: refs/heads/master@{#19889}
2017-09-18 12:22:39 +00:00
cb728ea83a Fix Gn Untracked headers in webrtc/modules/video_coding.
Fixed following headers in this CL
===================================

src/webrtc/modules/video_coding/sequence_number_util.h
src/webrtc/modules/video_coding/codecs/interface/common_constants.h
src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h

src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h

src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h

src/webrtc/modules/video_coding/test/test_util.h
src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h

Remaining:
===========
src/webrtc/modules/video_coding/include/video_codec_interface.h
src/webrtc/modules/video_coding/include/video_error_codes.h

BUG=webrtc:7620

Review-Url: https://codereview.webrtc.org/3012323002
Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 10:08:08 +00:00
9a2e906b0c Added RTCMediaStreamTrackStats.concealmentEvents
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.

Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
2017-09-18 08:58:06 +00:00
35dee81321 Clean out unused methods from VoiceEngine and VoEBase.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3018523002
Cr-Commit-Position: refs/heads/master@{#19880}
2017-09-18 08:57:01 +00:00
58b0316f3d Expose new video codec factories in the PeerConnectionFactory API
This CL exposes the new type of video codec factories that represent all
video codecs in the PeerConnectionFactory API, i.e. no extra internal SW
video codecs will be added. Clients of the new functions will be
responsible for adding all SW video codecs themselves, and also handling
SW fallback and simulcast.

BUG=webrtc:7925
R=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/3004353002 .
Cr-Commit-Position: refs/heads/master@{#19866}
2017-09-15 17:02:50 +00:00
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00