Commit graph

12 commits

Author SHA1 Message Date
Anton Sukhanov
4f08faae82 Introduce MediaTransportConfig
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
2019-05-21 18:58:33 +00:00
Jonas Oreland
a3aa9bd75b Make VideoBitrateAllocatorFactory injectable.
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).

With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.

WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@

Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
2019-04-17 06:17:34 +00:00
Jonas Oreland
6d83592367 Improve handling of packets with unknown ssrc.
Add a feature (gated by field trial) that stores
packets with unknown ssrc in a circular buffer
and replays them once a receive stream with matching
ssrc is created.

This improves situation where media is incoming
but signaling or SetFrameDecryptor is slow.

BUG=webrtc:10405

Change-Id: I7c7b2f4bd96c942c09e96db0cdae4ce5efef2541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127543
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27159}
2019-03-18 11:34:43 +00:00
Amit Hilbuch
e7a5f7bfae Modifying MediaChannel to accept CopyOnWriteBuffer by value.
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.

Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
2019-03-12 23:49:57 +00:00
Steve Anton
e25f595c0a Guard preferred_dscp with the network interface lock
Bug: webrtc:10389
Change-Id: I96112c2135c9c2d545140feeef6345f8a9b81086
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27043}
2019-03-08 23:59:41 +00:00
Steve Anton
ef50b25690 Remove lock in WebRtcVideoEngine
WebRtcVideoEngine is only ever accessed from one thread, so remove
the lock and replace it with ThreadChecker assertions.

Bug: None
Change-Id: I8c34eb6473f0ebaaaafe8a163c3f5d6f19074021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125240
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26957}
2019-03-04 19:17:24 +00:00
Ruslan Burakov
493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api.
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.


Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
2019-02-27 15:08:34 +00:00
Christian Fremerey
6c02541abe Revert "Delete video source proxying in WebRtcVideoSendStream"
This reverts commit b66003ca79.

Reason for revert: Causes bot failures in Chromium, see https://chromium-review.googlesource.com/c/chromium/src/+/1470391

Original change's description:
> Delete video source proxying in WebRtcVideoSendStream
>
> Bug: webrtc:10147
> Change-Id: Ib9f399e79d99f7d8db53fa38ef4b92986913ac26
> Reviewed-on: https://webrtc-review.googlesource.com/c/121569
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26633}

TBR=nisse@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10147
No-Try: True
Change-Id: I80395333d2be8fd3329c0bcdd6ed33d994a01ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/122940
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26672}
2019-02-13 22:42:43 +00:00
Niels Möller
b66003ca79 Delete video source proxying in WebRtcVideoSendStream
Bug: webrtc:10147
Change-Id: Ib9f399e79d99f7d8db53fa38ef4b92986913ac26
Reviewed-on: https://webrtc-review.googlesource.com/c/121569
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26633}
2019-02-11 12:43:31 +00:00
Anders Carlsson
45340ca824 Remove legacy video codec factories.
Removes the deprecated video codec factories and the related flag and
helper classes.

Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
2019-01-14 14:56:40 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
Renamed from media/engine/webrtcvideoengine.h (Browse further)