Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged
This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.
Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22131}
This is a reland of 5897fe27abcbe70f706cc23adc26147e0581f97e.
Adding back CallConfig::kDefaultStartBitrateBps as deprecated.
Also making BitrateContraints::kDefaultStartBitrateBps private to stop
it from being used in other places.
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
Bug: webrtc:8415
Change-Id: Iacfe2d6daedff710832ab89210c7c66d4403c93b
Reviewed-on: https://webrtc-review.googlesource.com/55980
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22123}
This fix will ensure that attempts to send DTMF
events before the channel is opened will return
a failure rather than disappearing the event.
Bug: webrtc:8908
Change-Id: I5044a0398dfd3dfe73b6ae1d48395e9809f81ad4
Reviewed-on: https://webrtc-review.googlesource.com/55480
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22121}
This reverts commit 5897fe27abcbe70f706cc23adc26147e0581f97e.
Reason for revert: Breaking internal builds
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
TBR=nisse@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I598040edba7f1ff8b39d2d9c3c3ceca5627aaa0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/55740
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22106}
This prepares for a CL extracting the bitrate configuration logic from
the Call class.
Also renaming BitrateConfig to BitrateConstraints.
Bug: webrtc:8415
Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
Reviewed-on: https://webrtc-review.googlesource.com/54400
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22104}
The max bitrate is already constrained to the bitrate specified by
ScreenshareLayerConfig, it should not be boosted past that.
Bug: webrtc:8785
Change-Id: If400e829b6bf209e3052e908fcabd65ba2c9457e
Reviewed-on: https://webrtc-review.googlesource.com/53320
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22027}
This class adds a convenience method that allows *sending* a task
to the queue (as opposed to posting). Sending is essentially
Post+Wait, a pattern that we don't want to encourage use of
in production code, but is convenient to have from a testing
perspective and there are already several places in the
source code where we use it.
Change-Id: I6efd1b2257e6c641294bb6e4eb53b0021d9553ca
Bug: webrtc:8848
Reviewed-on: https://webrtc-review.googlesource.com/50441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22022}
Extract the code to produce a stream of frames to its own class,
FakeFrameSource. Use in VideoAdapter unittests, to make the code simpler
and not depend on the deprecated cricket::VideoCapturer.
Bug: webrtc:6353
Change-Id: Ib5c34c6a0bd7f4338650459873ddc94b12d0c569
Reviewed-on: https://webrtc-review.googlesource.com/49740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21995}
The main pieces of this refactor are splitting up the creation of
simulcast layers for screenshare or the normal case, more consistent
naming, renaming streams to layers and trying to be more explicit with
some of the logic. Also added TODOs for future work to put more
application control into creating simulcast streams.
Bug: webrtc:8785
Change-Id: Ibf49fa0cc6d890ff96f8ee11c89d93a2c94119d6
Reviewed-on: https://webrtc-review.googlesource.com/47580
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21989}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
Currently all simulcast streams are set as active by default. This
update takes the values of the rtp encoding parameters and wires those
values down to the VideoSendStream and VideoStreamEncoder, so that the
appropriate simulcast streams can be turned off and on. This includes
adding more application specific controls in the EncoderStreamFactory.
Bug: webrtc:8653
Change-Id: Iaa7da3209cea0f0db72543981a319e319705cb00
Reviewed-on: https://webrtc-review.googlesource.com/47245
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21920}
When a simulcast stream is enabled or disabled, we want this state
change to be reflected properly in the RtpRtcp modules. Each video send
stream can contain multiple rtp_rtcp_modules pertaining to different
simulcast streams. These modules are currently all turned on/off when
the send stream is started and stopped. This change allows for
individual modules to be turned on/off. This means if a module stops
sending it will send a bye message, so the receiving side will not
expect more frames to be sent when the stream is inactive and the
encoder is no longer encoding/sending images.
Bug: webrtc:8653
Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4
Reviewed-on: https://webrtc-review.googlesource.com/42841
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21880}
This CL adds #error to spot where rtc_base/win32.h is unconditionally
included and fixes all the places where it happens.
Bug: webrtc:8814
Change-Id: I3c005acf2cdb58a51f1bcaa4acaeebd272c56660
Reviewed-on: https://webrtc-review.googlesource.com/46060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21840}
This CL adds a GN build flag to include builtin software codecs
(enabled by default).
When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.
Replaces https://webrtc-review.googlesource.com/c/src/+/29203
Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.
Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
This is a reland of 18c4261339dc76b220e7c805e36b4ea6f3dd161d
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=sprang@webrtc.org,stefan@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8630
Change-Id: Ib3df6f9b7158bff362a7ec66fc57e368682c5846
Reviewed-on: https://webrtc-review.googlesource.com/40980
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21688}
Current checks do not allow first frame after fallback to be sent for encode
because of the native checks. However, the rest of the frames are sent and
encoded correctly.
Fallback encoder, although SupportsNativeHandle() is false, can call ToI420()
which converts from native handles, and accordingly can return error or
success. This higher level check seems unnecessary.
Bug: webrtc:8021
Change-Id: I69780cc25eb1e06317ff213e9b80288064e9f1e3
Reviewed-on: https://webrtc-review.googlesource.com/40441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21685}
And wire it up to methods on RTCConfiguration, via MediaConfig::Video.
Bug: webrtc:8504
Change-Id: I30805ee20c11d1d2fe552eb81f16d514db0ba4a8
Reviewed-on: https://webrtc-review.googlesource.com/39786
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21670}
This reverts commit 18c4261339dc76b220e7c805e36b4ea6f3dd161d.
Reason for revert: Broke internal tests
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,shampson@webrtc.org
Change-Id: I0aeb743cbd2e8d564aa732c937587c25a4c49b09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/39883
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21647}
Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
Bug: webrtc:8653
Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
Reviewed-on: https://webrtc-review.googlesource.com/37740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21646}
To align behavior with video receive streams. Configuration of header
extensions happen outside the stream classes (i.e. in Call). Recent
changes stopped recreating streams when extensions changed, but relied
on reconfiguring the stream instead.
Bug: webrtc:4690
Change-Id: I9efe944f94b811c353628d3be34f548f998d0efc
Reviewed-on: https://webrtc-review.googlesource.com/39664
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21642}
GetSimulcastConfig had to be overloaded in order to not break downstream
client tests when the API was changed. Now that the downstream client
has been updated to use the new API, we can remove the overloaded
function.
Bug: webrtc:8630
Change-Id: I5d5d494e0579e60858d6efbb4715761394874b38
Reviewed-on: https://webrtc-review.googlesource.com/38882
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21590}
* VoEBase contains only stub methods (until downstream code is
updated).
* voe::Channel and ChannelProxy classes remain, but are now created
internally to the streams. As a result,
internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
for testing.
* Stream classes share Call::module_process_thread_ for their RtpRtcp
modules, rather than using a separate thread shared only among audio
streams.
* voe::Channel instances use Call::worker_queue_ for encoding packets,
rather than having a separate queue for audio (send) streams.
Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}
In preparation of moving ownership of voe::Channel to the audio stream
classes, semantics for changing configuration properties on the receive
streams need to change, otherwise RTP, audio decoding and NetEq state
will be discarded when streams are recreated. The same pattern as for
AudioSendStream is applied, and the reconfigurable information is kept
to a minimum.
AudioReceiveStream:s may still be recreated when an unsignaled stream
is 'promoted' to signaled state, and the sync label changes at the
same time.
Bug: webrtc:4690
Change-Id: Ibad282965310c3c8174a91e05a659fa3e1827607
Reviewed-on: https://webrtc-review.googlesource.com/38300
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21560}
GetSimulcastConfig should never return an empty vector of VideoStreams, because lower layers in the code expect atleast one VideoStream. It should also never be given input of max_streams equal to 0.
Bug: webrtc:8648
Change-Id: I60f59b3b267a732f07001e4c8a7fa64963802887
Reviewed-on: https://webrtc-review.googlesource.com/38061
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21545}
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}
The macros confuses automatic tooling, Qt Creator fails to identify the
tests defined with the special macros used before.
The value for readers of defining the macros is not obvious either.
Macros can sometime make code more compact and therefore quicker to
overview. However they also increases ambiguity of the code and the
reader will have to look up their definition to know what they do.
In this case I argue that the slight decrease in code size does not
outweigh the cost of lost tooling support.
Bug: None
Change-Id: Ic496fbe1fefdc5acd3f50ec99e2c804bb6065c3d
Reviewed-on: https://webrtc-review.googlesource.com/33540
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21503}
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}
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 is a temporary hack for the iPad Pro 12.9" gen2, which has
non-functional echo cancellation.
Bug: webrtc:8682
Change-Id: I646deeeb4723c4accac6f364c5c76a015791e202
Reviewed-on: https://webrtc-review.googlesource.com/35680
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21417}
Otherwise they're doing exactly the same as Clang bots.
Also fix 64-bit-specific warnings that have sneaked in because we have been testing MSVC build only on 32-bit for a while.
TBR=ehmaldonado@webrtc.org
Bug: webrtc:8664
Change-Id: I875e568d75aa550726f54650c283b288d3f52012
Reviewed-on: https://webrtc-review.googlesource.com/35160
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21414}
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}
The functionality is moved into AudioState.
TBR: henrika@webrtc.org
Bug: webrtc:4690
Change-Id: I015482ad18a39609634f6ead9e991d5210107f0f
Reviewed-on: https://webrtc-review.googlesource.com/34502
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21338}
The functionality is moved into AudioState.
Bug: webrtc:4690
Change-Id: Iee1bfd185566c9507422e8eea8a2cce02baaefe1
Reviewed-on: https://webrtc-review.googlesource.com/33521
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21324}
