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}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
* 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}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: I25e460b7a848c765369ce881f8833081eedf2558
Reviewed-on: https://webrtc-review.googlesource.com/23600
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21575}
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}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
It shouldn't be empty. As it was before it printed
RESULT min_test_bitrate_no_allocation_strategy: = 80 kbps
Whereas now it prints
RESULT min_test_bitrate_no_allocation_strategy: min_bitrate= 80 kbps
Bug: webrtc:7156
Change-Id: Ie86e3912d296d6d7bd6936d1709df9d2dc7fc143
Reviewed-on: https://webrtc-review.googlesource.com/38040
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21529}
All of the PlaysOutAudioAndVideoInSync* tests were reporting metrics under
the same name ("sync_convergence_time/synchronization") so that only one of
the tests (whichever ran last) had its metrics reported to the dashboard,
while the others were silently ignored.
I added a suffix to differentiate between them.
Bug: webrtc:8566
Change-Id: Ia51f0441d28b202581c5b22ef5ea683091557ab8
Reviewed-on: https://webrtc-review.googlesource.com/36541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21501}
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 change allows work to be done in parallel for lower level implementation and wiring/exposing multiple simulcast layer's encoding parameters at the api interface.
Bug: webrtc:8653
Change-Id: I89c9a6af0786134771d28526056759bd63213a0a
Reviewed-on: https://webrtc-review.googlesource.com/32902
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21375}
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
instead of pair of pointer + size.
it removes hidden memcpy in RtpPacketReceived::Parse:
RtpPacketReceived keeps a reference to a CopyOnWriteBuffer. By
passing it the same CopyOnWriteBuffer that was created by
BaseChannel, one allocation and memcpy is avoided.
Bug: None
Change-Id: I5f89f478b380fc9aece3762d3a04f228d48598f5
Reviewed-on: https://webrtc-review.googlesource.com/23761
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21143}
Header files base/videosinkinterface.h and base/videosourceinterface.h
were not part of any target (because they cause 2 dependency cycles).
This CL uncomment them so GN can keep dependencies under control, the
2 dependency cycles will be removed as part of webrtc:6828.
Bug: webrtc:6828
Change-Id: I5c5580facc010ba619e105a9b8a572ac70169a01
Reviewed-on: https://webrtc-review.googlesource.com/27621
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20970}
This makes it visible that there are no side effects and no dependency on BitrateAllocator.
Bug: None
Change-Id: I3d54ea545e694ae8303860114ddb3ce7569ecb14
Reviewed-on: https://webrtc-review.googlesource.com/26920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20933}
1. Bitrate allocation strategy unit test
2. Perf test determining minimal supported bitrate with and without strategy
Bug: webrtc:8243
Change-Id: Idf675fbadddb66c77b2582052d6497971eb99ad6
Reviewed-on: https://webrtc-review.googlesource.com/4880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20886}
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
GetBandwidthObserver should be used instead as it exposes a smaller interface.
Bug: webrtc:8415
Change-Id: I29ca795657e205186d7ebd929e756038a294b5f7
Reviewed-on: https://webrtc-review.googlesource.com/23900
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20871}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
