rtp_rtcp_format is lighter build target than rtc_media_base and
a more natural place to keep rtp parsing functions.
Bug: None
Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34504}
This builds on a few other CLs that avoid recreating the audio receive
streams on config changes and removes redundant config state in
WebRtcAudioReceiveStream, constructs the embedded receive stream in the
initializer list and keeps it const.
Bug: webrtc:11993
Change-Id: Iad28e0170bee6bf1e08713a89af7c81435b4265e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222100
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34317}
This is to be consistent with how things work on the video side but
also much less drastic than the current implementation. Aim is to
remove RecreateAudioReceiveStream(), which would improve efficiency
as well as allow for specific handling of the cases that currently
trigger recreation.
Bug: webrtc:11993
Change-Id: Ia81a5e66d44e41ea4eb2bff800e0b1583821c96a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34282}
This reverts commit 8a18e5b3c954a3f9cc006c90356a3d850bcc352f.
Reason for revert: Removing the problematic DCHECK.
Original change's description:
> Revert "Remove AudioReceiveStream::Reconfigure() method."
>
> This reverts commit e2561e17e29e62c02731f1d214d7ee5ffdaeb941.
>
> Reason for revert: Speculative revert: breaks an downstream project
>
> Original change's description:
> > Remove AudioReceiveStream::Reconfigure() method.
> >
> > Instead, adding specific setters that are needed at runtime:
> > * SetDepacketizerToDecoderFrameTransformer
> > * SetDecoderMap
> > * SetUseTransportCcAndNackHistory
> >
> > The whole config struct is big and much of the state it holds, needs to
> > be considered const. For that reason the Reconfigure() method is too
> > broad of an interface since it overwrites the whole config struct
> > and doesn't actually handle all the potential config changes that might
> > occur when the config changes.
> >
> > Bug: webrtc:11993
> > Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34252}
>
> TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11993
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34253}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:11993
Change-Id: I0d3bf9abdcdc8d3f9259d014e6074a5e6b6cc73c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221747
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34255}
This reverts commit e2561e17e29e62c02731f1d214d7ee5ffdaeb941.
Reason for revert: Speculative revert: breaks an downstream project
Original change's description:
> Remove AudioReceiveStream::Reconfigure() method.
>
> Instead, adding specific setters that are needed at runtime:
> * SetDepacketizerToDecoderFrameTransformer
> * SetDecoderMap
> * SetUseTransportCcAndNackHistory
>
> The whole config struct is big and much of the state it holds, needs to
> be considered const. For that reason the Reconfigure() method is too
> broad of an interface since it overwrites the whole config struct
> and doesn't actually handle all the potential config changes that might
> occur when the config changes.
>
> Bug: webrtc:11993
> Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34252}
TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34253}
Instead, adding specific setters that are needed at runtime:
* SetDepacketizerToDecoderFrameTransformer
* SetDecoderMap
* SetUseTransportCcAndNackHistory
The whole config struct is big and much of the state it holds, needs to
be considered const. For that reason the Reconfigure() method is too
broad of an interface since it overwrites the whole config struct
and doesn't actually handle all the potential config changes that might
occur when the config changes.
Bug: webrtc:11993
Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34252}
Pending messages on network thread for MediaChannel, will be dropped
when the MediaChannel object is deleted (without blocking).
Remove MessageHandler inheritance from Channel since Post-ing to the
network thread has been removed from there.
Copy/pasted code for SendRtp/SendRtcp in WebRtcVideoChannel and
WebRtcVoiceMediaChannel consolidated in MediaChannel.
Bug: webrtc:11993
Change-Id: I05320eb7f86b98adba50ca5eb8b76b78f4111263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217720
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33955}
This CL mostly adds plumbing to get awareness of the network thread
to the media channel classes. Currently this pointer is only used
to DCHECK that `SetInterface` for the `NetworkInterface` pointer, is
called on the network thread. Follow up changes will establish that
most of the methods are called on the network thread and the mutex
in the MediaChannel base class, can be removed.
Most of the changes in the CL are in channel_unittest.cc. They're mostly
around updating the tests to incorporate the network thread in ways
that reflect how the classes are used in production. Another change is
to use accessor methods for the media channel instances instead of
caching potentially dangling pointers.
Bug: webrtc:11993
Change-Id: I8e2ed1bc23724e238554dbce386789d69660f7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217682
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33951}
There's a bit of copy/pasted code in the channel code, which is
making moving network traffic consistently over to the network thread
a bit trickier than it needs to be, so I'm also updating variable
names used in Set[Local|Remote]Content_w to be more explicitly the same
and make it clear that the code is copy/pasted (and future updates can
consolidate more of it).
Also removing some code from the video/voice media channels that's
effectively dead code (vector + registration methods that aren't needed)
Bug: webrtc:12705
Change-Id: I2e14e69fbc489a64fc1e8899aaf1cfc979fe840b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215978
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33847}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.
The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.
This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.
This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.
This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.
Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
This is to allow FEC to be encoded at the lowest bitrate.
Bug: chromium:1086942
Change-Id: I1d30276a9a2aaa80016250dc786d5d867ba6cd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212501
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33539}
Ensures that the value read by the audio thread is well-defined.
Bug: b/176104610
Change-Id: I15d1901522be79703b3dc188fbe03c752be09a60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212442
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33503}
This is a reland of d48a2b14e7545d0a0778df753e062075c044e2a1
The diff of the reland (what caused the tsan error) can be seen
by diffing patch sets 2 and 3. Essentially I missed keeping the calls
to the transport controller on the worker thread. Note to self to add
thread/sequence checks to that code so that we won't have to rely on
tsan :)
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
Bug: webrtc:11993, webrtc:12430
Change-Id: I4fccaa418d22c2087a55bbb3ddbb25fac3b4dfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33153}
This reverts commit d48a2b14e7545d0a0778df753e062075c044e2a1.
Reason for revert: TSan tests started to fail constantly after this CL (it looks like it is flaky and the CQ was lucky to get green). See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25042/overview.
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: Id87cf9cbcc8ed58e74d755a110f0ef9dd980e298
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205525
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33145}
This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
Functionality wise, there should be no change with this CL, aside
from updating tests to anticipate OnPacketReceived to handle the packet
asynchronously (as already was the case via BaseChannel).
This only removes the network->worker hop out of the BaseChannel
class into the WebRTC MediaChannel implementations. However, it updates
the interface contract between BaseChannel and MediaChannel to align
with how we want things to work down the line, i.e. avoid hopping to
the worker thread for every rtp packet.
The following steps will be to update the video and voice channel
classes to call Call::DeliverPacket on the network thread and only
handle unsignalled SSRCs on the worker (exception case).
Bug: webrtc:11993
Change-Id: If0540874444565dc93773aee89d862f3bfc9c502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33040}
Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.
This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.
Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}
API to injecting a heavy audio processing operation into WebRTC audio capture pipeline
Bug: webrtc:12003
Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32291}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
this makes the association between opus and red a bit more obvious.
Also it allows access to the opus payload type which might be
used in the fmtp line in a future CL
BUG=webrtc:11640
Change-Id: I04e0648aedf049d103e3c3481c8712dfc9b79538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31868}
only enables RFC 2198 redundancy if it has a higher preference
than Opus. This means it not used by default but can be
chosen with setCodecPreferences.
BUG=webrtc:11640
Change-Id: I84ff2ca518da70440297a667dedba5cf4484eed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178742
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31830}
This CL changes the style of logging for an API which is essential when
WebRTC is used in Chrome. By changing the format, we can more easily
tie in (search for tags etc.) logs from WebRTC with logs in Chrome.
See e.g.
https://chromium-review.googlesource.com/c/chromium/src/+/2093443
for more details.
I decided to use a new private method to avoid using rtc::StringBuilder.
The idea was to make the log statements less complex and more condensed.
Tbr: mbonadei
Bug: webrtc:11493
Change-Id: I46b4a933ad62ac1db376743b4a41b62c5f8c6ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172841
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31808}
When enabled:
- Creates an audio network adapter config that is passed to audio send
stream.
- Configures a lower default min bitrate.
All parameters can be configured via a field trial that can also force
enable the audio network adaptor (this is mainly intended for testing).
Bug: chromium:1086942
Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31565}
negotiates the RED codec for opus audio behind a field trial
WebRTC-Audio-Redundancy
This adds the following line to the SDP:
a=rtpmap:someid RED/48000/2
To test start Chrome with
--force-fieldtrials=WebRTC-Audio-Red-For-Opus/Enabled
BUG=webrtc:11640
Change-Id: I8fa9fb07d03db5f90cdb08765baaa03d3d0458cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176372
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31562}
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.
This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.
The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.
The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
surface of APM.
2) Those files anyway needed to be moved to a separate build-
target to avoid a circular build-file dependency caused by
the other changes in this CL
Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in ChannelSend, where the frame
transformation will occur in the follow-up CL.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30946}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
