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}
The delay_ms stat in AudioprocessStats should be an Optional, because its value is not always computed. This CL changes it to an optional.
Bug: webrtc:8569
Change-Id: I42fd7a86b975c766b685444bf1829511f790da2a
Reviewed-on: https://webrtc-review.googlesource.com/33320
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21293}
The new estimator uses the timestamps attached to EncodedImage, and is
taken from the reverted cl
https://webrtc-review.googlesource.com/c/src/+/23720.
Bug: webrtc:8504
Change-Id: I273bbe3eb6ea2ab9628c9615b803a379061ad44a
Reviewed-on: https://webrtc-review.googlesource.com/31380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21289}
The title of Dock window on OSX10.12 is still 'Dock' with layer number 20.
So the removed codes here is reduntant.
Also fix a wrong comment.
Bug: webrtc:8460
Change-Id: I72d4c8f5741a1ccb00aa45897f11e85af8d24e05
Reviewed-on: https://webrtc-review.googlesource.com/33123
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21287}
This reverts commit 59283e4c66d038a00923736685457f4b53f922fe.
Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.
Sample error log:
[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
>
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
>
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
>
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
This reverts commit f711428898cb2f03d4d09738ba751cf0d316c631.
Reason for revert: Breaks downstream projects.
Original change's description:
> Use std::fstream instead of rtc::File to write perf results + rename flag.
>
> Use std::fstream instead of rtc::File to write perf results.
> On Android, when I use rtc::File, the results are not written for some reason.
>
> Also rename the flag to '--chartjson_result_file'.
>
> Bug: webrtc:8566
> Change-Id: I32215e2233e18690c41050dfd35ac77e01d11f35
> Reviewed-on: https://webrtc-review.googlesource.com/32001
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21225}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8566
Change-Id: I55611592c3171152cee97e64bff35a0d62cea510
Reviewed-on: https://webrtc-review.googlesource.com/33080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21283}
Also cleaned up a bit in RtpFrameReferenceFinder.
Bug: chromium:762556
Change-Id: Ib08d2e7ce4b146b359ce9ba823f3aa15776c71bc
Reviewed-on: https://webrtc-review.googlesource.com/32301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21282}
This CL removes the following GN variables: rtc_build_libyuv,
rtc_libyuv_dir (as requested in webrtc:7906).
It also removes some unneeded dependencies on //third_party/libyuv.
WebRTC targets were using public_deps to depend on //third_party/libyuv
and this created a build graph where targets that were depending on
//third_party/libyuv were not declaring the dependency to GN because
they were somehow getting it from another target that was exposing
//third_party/libyuv header files even if it wasn't directly depending
on it.
Bug: webrtc:8605, webrtc:7906
Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb
Reviewed-on: https://webrtc-review.googlesource.com/32201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21275}
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Though hacky, this is very similar to currently used implementation
in the RTCPSender::SendFeedbackPacket
Bug: webrtc:8239
Change-Id: I237b422ae1594dede78cb63daa4aa42b6774d6fe
Reviewed-on: https://webrtc-review.googlesource.com/32680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21274}
This is the first step towards removing this function from the interface. Once all implementers of the interface remove their implementations this can be removed.
Bug: webrtc:8572
Change-Id: Ia8f7f1b6949a482787df67b193d4cf999142e06e
Reviewed-on: https://webrtc-review.googlesource.com/27620
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21271}
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: I59bd51f0f5e2f8c187dff9fcf003a24c35ed037f
Reviewed-on: https://webrtc-review.googlesource.com/32600
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21267}
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}
When GetSources is called with non-existing ssrc, it will log the
error and return an empty RtpSource list instead of hitting the DCHECK.
Bug: chromium:793699
Change-Id: I30bebb657de32f87f9c82920fa0b19403893791f
Reviewed-on: https://webrtc-review.googlesource.com/32860
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21258}
