This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
When using WebRTC in iOS this Warning is shown for every single call even if stereo is not being used at all.
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Bug: webrtc:10146
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Reviewed-on: https://webrtc-review.googlesource.com/c/85283
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26133}
This removes the dependency on sstream which will reduce binary size.
Bug: webrtc:8982
Change-Id: I153390f2d8baf3e15875c39d1f5027b3575bca31
Reviewed-on: https://webrtc-review.googlesource.com/c/115748
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26127}
Previously we were setting the property again in it's setter. This is
obviously not a great idea. CL 109641 changed ivar accesses in blocks
to property accesses and this bug got introduced there.
Bug: webrtc:10110, webrtc:10127, webrtc:9971
Change-Id: I01abb0885b3bfc91fb741d82d1ece015ee9d3b58
Reviewed-on: https://webrtc-review.googlesource.com/c/116062
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26124}
This wires up the "state change" event and defines an observer
class that can be used by clients.
Bug: chromium:907849
Change-Id: I3cba2dc051a56280fb958f139f29cbb0022a39c6
Reviewed-on: https://webrtc-review.googlesource.com/c/114884
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26120}
The ANA frame length controller requires the provided frame lengths supported by the encoder to be ordered. A data structural guarantee of such was in an earlier version but was accidentally removed since https://codereview.webrtc.org/2429503002. This CL uses std::set to ensure that again.
Change-Id: Ia197dbf6a34f02506e81c9f49d6cd60e4cdacef4
BUG: webrtc:6303
Reviewed-on: https://webrtc-review.googlesource.com/c/115946
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26119}
This change adds Simulcast negotiation to media session offers/answers.
Next step is to add negotiation logic to PeerConnection.
Bug: webrtc:10075
Change-Id: Iea3a1084c16058f0efbc974cf623ec05c3c7a74f
Reviewed-on: https://webrtc-review.googlesource.com/c/115790
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26115}
Currently media transport can't log events to event log, but it should (things like bitrate estimates, goog cc logging, etc). This change make RtcEventLog available inside media transport.
Bug: webrtc:9719
Change-Id: I89a3b727049ccadc11c26c1d26ebaee3a1172556
Reviewed-on: https://webrtc-review.googlesource.com/c/115789
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26106}
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.
The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().
Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.
Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
This reverts commit d1208c26b1cdb536fdec942207033711101d5d26.
Reason for revert: This cl causes the crashing issue as in
chromium:916961 at starting desktop capture on Windows.
Original change's description:
> Desktop capturer: Add OnDisplayChanged callback
>
> This adds support for a new DesktopCapturer::Callback method
> OnDisplayChanged that is sent at the start of a desktop capture
> session and whenever the display geometry changes.
>
> This cl adds the basic structure to call this api at the start
> of the capture session. Currently Windows only.
>
> A follow-up cl will add support to call this whenever the display
> geometry changes.
>
> Bug: webrtc:10122, chromium:915411
> Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
> Reviewed-on: https://webrtc-review.googlesource.com/c/114020
> Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26053}
TBR=jamiewalch@chromium.org,braveyao@webrtc.org,braveyao@chromium.org,garykac@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10122, chromium:915411, chromium:916961
Change-Id: Id0471e01bb90bb5accdf58262ae2b130cf343ecd
Reviewed-on: https://webrtc-review.googlesource.com/c/115433
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26095}
SSRCs are specified twice in calls to the RtpVideoSender constructor.
Once in the first argument of ssrcs, and then again in the RtpConfig
ssrcs variable. Resolving to reference the variable in the RtpConfig.
Bug: None
TBR: stefan@webrtc.org
Change-Id: I53528140166a53f3558f950d5662b7d3d6b8c822
Reviewed-on: https://webrtc-review.googlesource.com/c/114910
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26094}
The new controller behaves mostly like before, but increases the target
rate on timer update rather than when feedback is received. This makes
the behavior easier to predict. It also uses a duration parameter to
track the increase, removing the meed for the minimum rate increase
constants that exists in the previous solution.
Bug: webrtc:9718
Change-Id: Iae31a9ba2d6474a8236f8eb72f86ff434f1d1fc6
Reviewed-on: https://webrtc-review.googlesource.com/c/114681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26088}
Perf bots can be removed from this config because they will not perform compilation anymore.
Note that Linux64 Builder already exists.
Bug: chromium:908001
Change-Id: I3d2de332083bc0e7054fa09f8814c6500fad9ee4
Reviewed-on: https://webrtc-review.googlesource.com/c/115413
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26087}
This test was added in
https://webrtc-codereview.appspot.com/15789004/, which looks like it
was early on in adding support for capture and encode with native
textures on Android. Since then, the VideoFrame/VideoFrameBuffer
interfaces have emerged and texture support has changed a lot (now
using VideoFrameBuffer with type kNative).
The test only exercises the parts of the video pipeline before
VideoStreamEncoder::pre_encode_callback_, which doesn't care at all
about the type of the corresponding VideoFrameBuffer. That's not so
useful, and since it blocks removal of pre_encode_callback, let's
delete this old test.
Bug: webrtc:9864, chromium:362437
Change-Id: I2eb6c4c48557883309fd6431bc25528441c83078
Reviewed-on: https://webrtc-review.googlesource.com/c/115411
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26086}
