It will be necessary to keep the H264 profile information in
VideoReceiveStream::Decoder. I think it will be easier now and for the
future to just store all of the codec parameters unmodified in
VideoReceiveStream::Decoder instead of extracting a subset of them to an
ad hoc class.
BUG=webrtc:6743,webrtc:5948
Review-Url: https://codereview.webrtc.org/2523773003
Cr-Commit-Position: refs/heads/master@{#15239}
WebRTC.Video.BitrateSentInKbps
WebRTC.Video.MediaBitrateSentInKbps
WebRTC.Video.PaddingBitrateSentInKbps
WebRTC.Video.RetransmittedBitrateSentInKbps
WebRTC.Video.FecBitrateSentInKbps
RtpSender has two StreamDataCounters: for the non-RTX and the RTX stream.
The same counter (for the non-RTX stream) is reported for both the media SSRC and the FlexFEC SSRC.
Bitrate stats are summed for all SSRCs, thus the counter for the non-RTX stream is counted twice.
Do not store the counter for the FlexFEC SSRC.
Do not include info from FlexFEC substreams in VideoSendStream::Stats::ToString (periodically logged during a call).
BUG=webrtc:6774
Review-Url: https://codereview.webrtc.org/2525293002
Cr-Commit-Position: refs/heads/master@{#15238}
Turns out this function is needed by external code.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2532663002
Cr-Commit-Position: refs/heads/master@{#15237}
- Out from modules/utility/ and into modules/audio_device/ios/ - there they are used.
BUG=none
Review-Url: https://codereview.webrtc.org/2526273002
Cr-Commit-Position: refs/heads/master@{#15236}
Reason for revert:
Include fix for downstream import.
Original issue's description:
> Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ )
>
> Reason for revert:
> Breaks downstream import.
>
> Original issue's description:
> > Split out target rtc_media_base from rtc_media
> >
> > The purpose with this CL is to be able to depend on
> > cricket::VideoCodec (webrtc/media/base/codec.h) from other targets
> > without getting cyclic dependencies.
> >
> > BUG=webrtc:6402,webrtc:6337
> >
> > NOTRY=True
> >
> > Committed: https://crrev.com/aae7e7cf35a5bb43ebbaf75396aa7ccc544e920a
> > Cr-Commit-Position: refs/heads/master@{#15137}
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6402,webrtc:6337
>
> Committed: https://crrev.com/0d0d7531b50a78efe7468610395e9dc5f496e2e9
> Cr-Commit-Position: refs/heads/master@{#15139}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6402,webrtc:6337
Review-Url: https://codereview.webrtc.org/2509123003
Cr-Commit-Position: refs/heads/master@{#15235}
Reason for revert:
Breaks downstream projects.
Original issue's description:
> Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
>
> This CL removes RTPPayloadStrategy that is currently used to handle
> audio/video specific aspects of payload handling. Instead, the audio and
> video specific aspects will now have different functions, with linear
> code flow.
>
> This CL does not contain any functional changes, and is just a
> preparation for future CL:s.
>
> The main purpose with this CL is to add this function:
> bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> const webrtc::VideoCodec& video_codec);
> that can easily be extended in a future CL to look at video codec
> specific information.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> Cr-Commit-Position: refs/heads/master@{#15232}
TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2528993002
Cr-Commit-Position: refs/heads/master@{#15234}
This CL removes RTPPayloadStrategy that is currently used to handle
audio/video specific aspects of payload handling. Instead, the audio and
video specific aspects will now have different functions, with linear
code flow.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
The main purpose with this CL is to add this function:
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const webrtc::VideoCodec& video_codec);
that can easily be extended in a future CL to look at video codec
specific information.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2524923002
Cr-Commit-Position: refs/heads/master@{#15232}
The purpose with this CL is to be able to send video codec specific
information down to RTPPayloadRegistry. We already do this for audio
with explicit arguments for e.g. number of channels. Instead of
extracting the arguments from webrtc::CodecInst (audio) and
webrtc::VideoCodec, this CL sends the types unmodified all the way down
to RTPPayloadRegistry.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
In the dependent CL https://codereview.webrtc.org/2524923002/,
RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled
audio/video specific aspects of payload handling. After this CL, we will
know if we get audio or video codecs without any dependency injection,
since we have different functions with different signatures for audio
vs video.
BUG=webrtc:6743
TBR=mflodman
Review-Url: https://codereview.webrtc.org/2523843002
Cr-Commit-Position: refs/heads/master@{#15231}
This perf tests the echo detector in 3 scenarios: standalone, as part of APM with only the echo detector enabled and as part of a normally configured APM.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2517523003
Cr-Commit-Position: refs/heads/master@{#15224}
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.
This test ensures that audio starts pumping. The exact performance is
not that important.
R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True
Review-Url: https://codereview.webrtc.org/2525943003
Cr-Commit-Position: refs/heads/master@{#15223}
The histogram will log a new value every time the AGC changes level_.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2525963002
Cr-Commit-Position: refs/heads/master@{#15222}
When set to true, this adds the files necessary to run memcheck as data dependencies, listed in the .gni files.
This will enable us to run memcheck on swarming.
R=kjellander@chromium.org
BUG=chromium:497757
NOTRY=True
Review-Url: https://codereview.webrtc.org/2510033004
Cr-Commit-Position: refs/heads/master@{#15219}
Change the default value of rtcp-mux policy in RTCConfiguration.
Refactor the peerconnectioninterface and webrtcsession unit tests.
BUG=webrtc:6030
Review-Url: https://codereview.webrtc.org/2043193003
Cr-Commit-Position: refs/heads/master@{#15217}
I forget to remove these when fixing them.
BUG=chromium:636818
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2522023003
Cr-Commit-Position: refs/heads/master@{#15215}
There shouldn't be a need to bind the framebuffer to modify the texture
size. Binding the framebuffer causes a crash on Samsung Galaxy S3 mini
for some reason.
BUG=webrtc:6470
Review-Url: https://codereview.webrtc.org/2524003002
Cr-Commit-Position: refs/heads/master@{#15210}
RTCStatsCollector supports "payloadType", "codec" and "clockRate".
"channels", "parameters" and "implementation" need to be supported
before closing crbug.com/659117.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117, chromium:627816, chromium:657854
NOTRY=True
Review-Url: https://codereview.webrtc.org/2509803004
Cr-Commit-Position: refs/heads/master@{#15207}
Increases readibility and helps with the formating.
BUG=webrtc:6680
Review-Url: https://codereview.webrtc.org/2488973002
Cr-Commit-Position: refs/heads/master@{#15206}
This change adds RotateDesktopFrame(), RotateRect(), RotateSize(),
ReverseRotate() functions, so an implementation can use these free functions to
rotate and copy pixels from one DesktopFrame to another at the same time.
This is the first part of the change to support rotation in DirectX capturer. In
a coming change, these functions will be used in DxgiOutputDuplicator to do the
rotation and copying.
Background,
DirectX APIs always return unrotated data buffer, so we need to rotate it to
match the user-selected rotation. What worse is except for the data buffer,
other variables return by these APIs are all rotated, e.g. output size, monitor
position. So we will eventually not be able to capture the rotated monitors,
because we cannot set their position and size correctly. Though
DXGI_OUTDUPL_DESC provides a DXGI_MODE_ROTATION enumeration to indicate the
output rotation, it does not provide a simple way to rotate an IDXGIResource,
which is the only thing we can get from duplication APIs. A typical user case
here is to use a matrix to transform the IDXGIResource and render it to a
surface. But since we do not render the IDXGIResource at all, we need to
manually rotate it.
BUG=314516
Review-Url: https://codereview.webrtc.org/2500883004
Cr-Commit-Position: refs/heads/master@{#15205}
Previously ProbeController was starting probing as soon as SetBitrates()
is called. As result these probes would often timeout while connection
is being established. Now ProbeController receives notifications about
network route changes. This allows to start probing only when transport
is connected. This also makes it possible to restart probing whenever
transport route changes (will be done in a separate change).
BUG=webrtc:6332
R=honghaiz@webrtc.org, philipel@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2458863002 .
Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
Cr-Original-Commit-Position: refs/heads/master@{#15094}
Cr-Commit-Position: refs/heads/master@{#15204}
Set the |qp_| field in EncodedImage before calling OnEncodedImage.
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2499003002
Cr-Commit-Position: refs/heads/master@{#15199}
the APM parameters to the high-pass filter.
The introduction will be done in three steps:
1) This CL which introduces the new scheme and
changes the code in webrtcvoiceengine.cc to use it.
2) Introduce the scheme into upstream code.
3) Remove the HighPassFilter interface in APM.
BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298
Review-Url: https://codereview.webrtc.org/2415403002
Cr-Commit-Position: refs/heads/master@{#15197}
This new histogram will log a value every time input clipping is
detected. The value is a boolean, with "true" meaning that the gain
was in fact adjusted in response to the detected clipping, and "false"
meaning that adjustment was not allowed due to kClippedLevelMin.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2522543006
Cr-Commit-Position: refs/heads/master@{#15196}
This removes the need of executing Chromium's runhooks.
A selection of hooks (not all) were picked that should be
sufficient for WebRTC's needs.
New dependencies:
* third_party/espresso
* third_party/javax_inject
* tools/clang_format_merge_driver
BUG=webrtc:5578, webrtc:5006, webrtc:6741
TESTED=
rm -rf third_party/android_tools/sdk/extras/google/m2repository/
gclient runhooks
Verified that third_party/android_tools/sdk/extras/google/m2repository/ is downloaded without any license prompt.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2524673002
Cr-Commit-Position: refs/heads/master@{#15195}
This CL re-routes audio through AudioMixer instead of AudioConferenceMixer.
This is done without any modifications to VoiceEngine.
Previously, output audio was polled by an AudioDevice through an AudioTransport
pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the
request for data on to OutputMixer and further to AudioConferenceMixer.
This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice
to have another AudioTransport pointer, which points to an AudioTransportProxy.
The AudioTransportProxy is responsible for feeding mixed data to the
AudioProcessing component for echo cancellation, and to resample the audio data
after AudioProcessing and before it is sent to the AudioDevice.
The set up of the audio path was previously done during VoiceEngine
initialization. Now it is changed in the AudioState constructor.
This list shows where audio-path-related VoiceEngine functionality has been
moved:
OutputMixer --> AudioTransportProxy
VoiceEngineImpl --> AudioState, AudioTransportProxy
SharedData --> AudioState
Channel --> AudioReceiveStream, ChannelProxy, Channel
AudioState owns the new mixer and connects it to AudioTransport and
AudioDevice on initialization.
The audio input source is AudioReceiveStream, which registers itself with the
mixer (which it gets from AudioState) on Start and Stop.
# Since the AudioTransport interface contains non-const references.
NOPRESUBMIT=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2436033002
Cr-Commit-Position: refs/heads/master@{#15193}
The mock is used in a dependent CL https://codereview.webrtc.org/2436033002.
There is also a goal to allow external mixing implementations
(subclasses of webrtc::AudioMixer) and inject them to
PeerConnectionFactory. We think that part of that is an official and
maintained mock.
Summary of changes:
* Created a mixer mock/stub in webrtc/api/test
* Made a target webrtc/api:mock_audio_mixer for it.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2520323002
Cr-Commit-Position: refs/heads/master@{#15190}
