170 Commits

Author SHA1 Message Date
Peter Thatcher
c2ee2c86f9 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1229283003 .

Cr-Commit-Position: refs/heads/master@{#9690}
2015-08-07 23:05:42 +00:00
qiangchen
c27d89fdc6 Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.

Review URL: https://codereview.webrtc.org/1225153002

Cr-Commit-Position: refs/heads/master@{#9597}
2015-07-16 17:27:23 +00:00
jbauch
bd38428089 Don't use result of "field_trial::FindFullName" as string reference.
"field_trial::FindFullName" can return "std::string()" which should not
be referenced by the caller.

Review URL: https://codereview.webrtc.org/1238943003

Cr-Commit-Position: refs/heads/master@{#9594}
2015-07-16 11:06:02 +00:00
jbauch
083b73fb95 Use std::string references instead of copying contents.
This CL improves the memory footprint a bit by using string references
instead of creating a copy.

Review URL: https://codereview.webrtc.org/1241973002

Cr-Commit-Position: refs/heads/master@{#9592}
2015-07-16 09:46:43 +00:00
Jelena Marusic
cd6702282a Define Stream base classes
BUG=webrtc:4690

Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.

R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1226123005 .

Cr-Commit-Position: refs/heads/master@{#9591}
2015-07-16 07:30:20 +00:00
pbos
8fc7fa798f Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.

BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1181653002

Cr-Commit-Position: refs/heads/master@{#9586}
2015-07-15 15:03:04 +00:00
noahric
d10a68e797 Don't create unsignalled receive streams for RTX, RED RTX, and ULPFEC packets.
BUG=webrtc:4389

Review URL: https://codereview.webrtc.org/1226093002

Cr-Commit-Position: refs/heads/master@{#9566}
2015-07-10 18:28:02 +00:00
Zeke Chin
71f6f4405c iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.

BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1187573004.

Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 21:35:08 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
2015-05-30 00:21:56 +00:00
Peter Boström
259bd2034c Report ssrc_groups in GetStats().
This was already available in the stats struct, just not filled in.

BUG=4720
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47329004

Cr-Commit-Position: refs/heads/master@{#9308}
2015-05-28 11:40:00 +00:00
Henrik Boström
3b187b9c0c Removed unnecessary includes of webrtcvideocapturer.h
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57469004

Cr-Commit-Position: refs/heads/master@{#9305}
2015-05-28 09:43:45 +00:00
Peter Boström
3548dd2154 Set local SSRCs on receivers added before senders.
Addresses bug where a receiver would report SSRC 1 even though the
endpoint has sending streams.

BUG=4678
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51099004

Cr-Commit-Position: refs/heads/master@{#9262}
2015-05-22 16:48:13 +00:00
Fredrik Solenberg
9a416bd14e Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51879004

Cr-Commit-Position: refs/heads/master@{#9258}
2015-05-22 07:03:48 +00:00
Peter Boström
4d71edef45 Add HW fallback option to software encoding.
Permits falling back to software encoding for unsupported resolutions.

BUG=chromium:475116, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46279004

Cr-Commit-Position: refs/heads/master@{#9227}
2015-05-19 21:09:17 +00:00
Peter Boström
7252a2ba80 Add HW fallback option to software decoding.
Permits falling back to software decoding for unsupported resolutions in
bitstreams.

BUG=4625, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46269004

Cr-Commit-Position: refs/heads/master@{#9209}
2015-05-18 17:41:50 +00:00
Peter Boström
67c9df7982 Base NACK on send codecs.
Addressing discrepancy where NACK used to be set from send codecs in
WebRtcVideoEngine(1), and before this change, from recv codecs in
WebRtcVideoEngine2. This should address that NACK might be sent even if
the remote side does not support it.

BUG=4626
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53409004

Cr-Commit-Position: refs/heads/master@{#9171}
2015-05-11 12:34:53 +00:00
Peter Boström
126c03ea02 Base decision to send REMB on send codecs.
Fixes bug where Chromium would send REMB even though the remote party
doesn't announce support for it (because it was based on local codec
settings instead of remote ones).

BUG=4626
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54389004

Cr-Commit-Position: refs/heads/master@{#9170}
2015-05-11 10:48:08 +00:00
Fredrik Solenberg
d3ddc1b69e Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
BUG=
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49929004

Cr-Commit-Position: refs/heads/master@{#9156}
2015-05-07 15:07:36 +00:00
Fredrik Solenberg
4b60c73e74 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49269004

Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
Peter Boström
81ea54eaac Remove WebRtcVideoEngine.
Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until
build files in Chromium have been modified.

BUG=1695,4566
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48339004

Cr-Commit-Position: refs/heads/master@{#9148}
2015-05-07 09:41:10 +00:00
Alex Glaznev
e433c0ef31 Restore back verbosity logging for camera captured frame.
Helps to debug camera freezes.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46179004

Cr-Commit-Position: refs/heads/master@{#9127}
2015-05-01 20:54:27 +00:00
Erik Språng
efbde3775b Don't use CPU adaptation for screen content in the new API.
BUG=4605
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48309004

Cr-Commit-Position: refs/heads/master@{#9116}
2015-04-29 14:21:32 +00:00
Fredrik Solenberg
23fba1ffa0 Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51749004

Cr-Commit-Position: refs/heads/master@{#9114}
2015-04-29 13:24:10 +00:00
Alex Glaznev
faa6d076b7 Remove a few verbose log messages from webrtcvideoengine2.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49189004

Cr-Commit-Position: refs/heads/master@{#9105}
2015-04-28 16:40:51 +00:00
Erik Språng
143cec1cc6 Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.

BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46069004

Cr-Commit-Position: refs/heads/master@{#9093}
2015-04-28 08:01:14 +00:00
Peter Boström
ee0b00e8a9 Prevent recv-stream reconfig on identical codecs.
Receive streams seem to be reconfigured with identical codecs when
another stream is removed. Preventing this reconfiguration makes sure
that existing streams don't report stats during teardown when the stream
is still supposed to be running.

BUG=1788
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44249004

Cr-Commit-Position: refs/heads/master@{#9059}
2015-04-22 16:40:58 +00:00
Peter Boström
393347ff98 Report receive-side packet loss.
BUG=4558
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48219004

Cr-Commit-Position: refs/heads/master@{#9054}
2015-04-22 12:52:31 +00:00
Henrik Kjellander
7c027b64ae Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46999004

Cr-Commit-Position: refs/heads/master@{#9053}
2015-04-22 11:21:10 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 12:25:42 +00:00
Peter Boström
3c3f646064 Prevent null-stream reconfigs on RTP extensions.
If a codec fails to set (e.g. there's no codec configured), this
prevents a stream reconfigure with an invalid config. Reconfiguring a
stream without correct codec settings causes a CHECK failure.

BUG=chromium:475116
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44219004

Cr-Commit-Position: refs/heads/master@{#9007}
2015-04-15 14:27:39 +00:00
Peter Boström
e432800aeb Enable CPU adaptation by default.
WebRtcVideoEngine2 doesn't support CPU-monitor-based adaptation and as
such requires encoder-time-based CPU adaptation to perform any
adaptation at all.

BUG=4536
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49679004

Cr-Commit-Position: refs/heads/master@{#9001}
2015-04-14 20:45:23 +00:00
Peter Thatcher
77f0e3f7b6 Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48039004

Cr-Commit-Position: refs/heads/master@{#8990}
2015-04-13 17:44:56 +00:00
Peter Boström
e7b221f476 Remove deadlock in WebRtcVideoEngine2.
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.

R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065

Review URL: https://webrtc-codereview.appspot.com/50679004

Cr-Commit-Position: refs/heads/master@{#8985}
2015-04-13 13:34:32 +00:00
Peter Boström
ad1f9b61a3 Remove warning on input frames before config.
Removes log spam for AppRTC when only one client is connected.

BUG=4512
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48019005

Cr-Commit-Position: refs/heads/master@{#8947}
2015-04-08 12:04:06 +00:00
Guo-wei Shieh
64c1e8cda5 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8917}
2015-04-01 22:33:15 +00:00
Minyue
31331cfd2d Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae.

Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.

Review URL: https://webrtc-codereview.appspot.com/47909004

Cr-Commit-Position: refs/heads/master@{#8911}
2015-04-01 14:20:11 +00:00
Guo-wei Shieh
1b1c15cad1 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8905}
2015-04-01 02:42:50 +00:00
Peter Boström
23914fe756 Reject RTP one-byte extension ID 0.
Only accept local identifiers in the range 1-14 inclusive.

BUG=1788, chromium:471328
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50549004

Cr-Commit-Position: refs/heads/master@{#8900}
2015-03-31 13:08:13 +00:00
Peter Boström
dfd53fe26b Raise streams for SetMaxSendBitrates above 2000k.
Fixes b=AS effectively not setting bitrates above 2000k.

BUG=1788,4469
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47839004

Cr-Commit-Position: refs/heads/master@{#8882}
2015-03-27 14:58:17 +00:00
Peter Boström
74d9ed7d85 Report send codec name in GetStats().
BUG=4461
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51439004

Cr-Commit-Position: refs/heads/master@{#8869}
2015-03-26 15:28:43 +00:00
Peter Boström
d6f4c25eed Reject streams reusing simulcast or RTX SSRCs.
BUG=1788, chromium:470122, chromium:470856
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42919004

Cr-Commit-Position: refs/heads/master@{#8868}
2015-03-26 15:23:13 +00:00
Stefan Holmer
e590416722 Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
2015-03-26 10:11:22 +00:00
Peter Boström
d4362cd336 Reject StreamParams with RTX SSRCs not in ssrcs.
BUG=1788, chromium:470122
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44859004

Cr-Commit-Position: refs/heads/master@{#8855}
2015-03-25 13:17:33 +00:00
kwiberg@webrtc.org
eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-24 09:20:19 +00:00
pbos@webrtc.org
a5f6fb53ba Permit single-stream max bitrates above 2000k.
BUG=4463
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49509004

Cr-Commit-Position: refs/heads/master@{#8839}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8839 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 22:30:11 +00:00
pbos@webrtc.org
b9557a9bb7 Fix code to handle crashes for non-VP8.
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.

BUG=1667, 1788
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44789004

Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 19:53:15 +00:00
pbos@webrtc.org
8296ec518b Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44779004

Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 14:28:31 +00:00
perkj@webrtc.org
af612d5e07 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.

Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306

Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47629004

Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:51:44 +00:00
magjed@webrtc.org
afdd5dd372 Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.

Reason for revert: The thread checker in Chromium that crashed has been fixed now.

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/40319004

Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 13:11:43 +00:00
magjed@webrtc.org
b218ff5531 Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.

Reason for revert: Fails on Chromium FYI content_browsertests

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/47529004

Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 15:29:59 +00:00