So that we can avoid dependency cycles.
Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
Slight change in functionality: send_stream_ member is no longer moved
to the QueuedTask. This means that a possible race on access to
send_stream_ will not cause nullpointer dereferencing until the posted
task has been run. Most usages of send_stream_ are protected by
thread_checker_, but not DeliverRtcp and EnableEncodedFrameRecording.
This change in behavior should be be able to cause new failures, but it
could potentially make existing race conditions less likely to happen.
Bug: None
Change-Id: Ife42071a4aa2811fcaf2f3ef21ca1888e6640ca3
Reviewed-on: https://webrtc-review.googlesource.com/59800
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22327}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. It will be done in 3 steps:
1. Test/fake_audio_device will be moved to production part of WebRTC
source code and renamed to test_audio_device_module. Old header is
replaced with alias to the new one.
2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice.
3. test/fake_audio_device will be removed.
This CL implements 1st step.
Bug: webrtc:8946
Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f
Reviewed-on: https://webrtc-review.googlesource.com/59740
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22325}
This will give accurate stats for the number of calls
that use video that are using SDES.
Bug: chromium:804275
Change-Id: I35b045a2301fb5267b656b424b9b3482b1b72f9a
Reviewed-on: https://webrtc-review.googlesource.com/60481
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22324}
Trigger on total bitrate change.
Bug: webrtc:8955
Change-Id: I2373a1b7f139c7ea748a9641593e714d6895c8f6
Reviewed-on: https://webrtc-review.googlesource.com/59323
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22323}
This file ended up in the wrong place and prevented building without
SW codecs.
Bug: webrtc:7925
Change-Id: I7909561d96051d5653821130c666ac66938e3edc
Reviewed-on: https://webrtc-review.googlesource.com/59640
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22320}
This updates AppRTC to use addTrack instead of addStream, and removes
the use of onAddStream, because we no longer have to wait for this to be
fired to set the remote track's video renderers.
Bug: webrtc:8869
Change-Id: I1ecae684a9bc4b30512e8c5d717e72b52c589831
Reviewed-on: https://webrtc-review.googlesource.com/57840
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22318}
This adds wrappers to the following native APIs:
- SdpSemantics enum added to the RTCConfiguration
- RtpTransceiver
- PeerConnection.addTrack
- PeerConnection.removeTrack
- PeerConnection.addTransceiver
- PeerConnection.getTransceivers
These APIs are used with the new Unified Plan semantics.
Bug: webrtc:8869
Change-Id: I19443f3ff7ffc91a139ad8276331f09e57cec554
Reviewed-on: https://webrtc-review.googlesource.com/57800
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22317}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This change adds UMA stats that record the format of the remote offered
SDP. There are three classifications for the SDP format:
- Simple: No more than one audio and one video. Should be compatible
with both Plan B and Unified Plan endpoints.
- ComplexPlanB: More than one audio or more than one video in the Plan B
format (e.g., one audio mline with multiple streams).
- ComplexUnifiedPlan: More than one audio or more than one video in the
Unified Plan format (e.g., two audio mlines).
Bug: chromium:811683
Change-Id: If46394edfa6a812ef313d632e27ec27516c18867
Reviewed-on: https://webrtc-review.googlesource.com/57220
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22315}
This was resulting in the currently executing stats callback to be
invoked *again*, possibly ad infinitum, resulting in stack overflow.
Bug: webrtc:8973
Change-Id: Ib3bcc8e6bdd991728214fa242dda2010efc919a7
Reviewed-on: https://webrtc-review.googlesource.com/59464
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22313}
This CL introduces a different rampup behavir for the call startup and after resets
that may occur due to delay changes, clock-drift and audio path glitches.
Bug: chromium:819111, webrtc:8979
Change-Id: Ied1d7896be7f0c69aa6deb61475117021ca6ab09
Reviewed-on: https://webrtc-review.googlesource.com/60002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22312}
This is according to WebRTC spec.
Bug: chromium:819118
Change-Id: I4c0c8e69de812730f9f5c0a1b86d08cf05a472ad
Reviewed-on: https://webrtc-review.googlesource.com/60181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22309}
It holds the same information as codecType, but in different format.
Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
The OWNERS entries for style-guide.md were removed in
https://webrtc-review.googlesource.com/1560, apparently by accident;
add them back.
And while we're at it, add an entry for native-api.md as well.
Bug: none
Change-Id: I924b9e8dab297e668055af3042ac70123e4c00dc
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/60001
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22306}
We use template magic to let them handle both the presence and absence
of the new argument. This will be removed in a later CL, when we can
assume that new argument is always present.
Bug: webrtc:8941
Change-Id: I2d47f7c8572a9f03e742401dcf491b948b161f63
Reviewed-on: https://webrtc-review.googlesource.com/58081
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22301}
This removes the following methods:
- SetAudioSend (directly accessed through MediaChannel now)
- "Early Media" (feature not used)
- GetStats (directly accessed through MediaChannel now)
Bug: None
Change-Id: Ifd075d030b0f5f41e94918979891592a731d5a91
Reviewed-on: https://webrtc-review.googlesource.com/59500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22298}
Thread's constructor calls DoInit, which registers itself with the
MessageQueueManager. This could result in the vptr being read before
the subclass has had a chance to modify it (for example, if another
thread happens to call MessageQueueManager::Clear at the right time).
This is exactly why there's a "DoInit" method, which is intended to be
called by the fully instantiated subclass. This was being done between
MessageQueue/Thread, but not between Thread and its subclasses.
Bug: webrtc:3911
Change-Id: I94d8855da56d9aaf22470ddca12d0b1dd5de249d
Reviewed-on: https://webrtc-review.googlesource.com/59466
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22297}
Just a null ID for now, but future CLs will fix that.
Bug: webrtc:8941
Change-Id: I393af0fef752ca3711421bdaf4b2e41cbe286bcf
Reviewed-on: https://webrtc-review.googlesource.com/58093
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22296}
This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.
Reason for revert: breaks downstream project
Original change's description:
> Separate test/fake_audio_device on API and implementation.
>
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
>
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
>
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}
TBR=kwiberg@webrtc.org,titovartem@webrtc.org
Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
This removes the routing for the deprecated audio control setting
Bug: none
Change-Id: If7a134ee487b80a653ba982768ba74ce2d539e0a
Reviewed-on: https://webrtc-review.googlesource.com/58941
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22288}
These counters will register whether the media sections
used with SDES are for audio, video or data.
Bug: chromium:804275
Change-Id: I1da3bb6625af755c0897bf4cd349655cb283fbb6
Reviewed-on: https://webrtc-review.googlesource.com/59400
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22287}
These tests cannot run on simulators but should be enabled on real device
bots in order to catch regressions or crashes in the iOS codecs.
Bug: webrtc:8950
Change-Id: I8e877aa4368683073fdb4586cd6f4add4a1284ad
Reviewed-on: https://webrtc-review.googlesource.com/59040
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22283}
This provides the empty shell of an AudioGenerator class.
It is intended to be used for debugging purposes and can be inserted
into the APM much like an AecDump. It allows for playing out diagnostic
audio unaffected by codecs and network jitter, while still capturing
API interaction like in a normal call.
NOTRY=True
Bug: webrtc:8882
Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7
Reviewed-on: https://webrtc-review.googlesource.com/53300
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22282}
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.
Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
This removes the routing for the deprecated audio control setting
Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
