Unfortunately, H264 makes it non-trivial to compare video formats for
equality. For every video format besides H264 it's enough to look at the
name, but for H264, we need to dig into the parameters. This logic is
currently in several places, and this CL unifies it to one place.
Bug: webrtc:7925
Change-Id: I83a516b108d6b4d6792fd0bf1d24296916d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/25120
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20853}
This can be used to determine whether the bitrate of a given spatial and temporal layer has been set in the allocation, even if the value it's set to is zero.
GetBitrate still returns 0 if the queried layer does not have the bitrate set.
Bug: webrtc:8479
Change-Id: I1d982e211da9b052fcccdbf588b67da1a4550c60
Reviewed-on: https://webrtc-review.googlesource.com/17440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20852}
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}
This is causing compilation failure on Chrome on Windows using MSVC
64bits.
TBR=andersc@webrtc.org
Bug: chromium:787192
Change-Id: If82290a9d7373385de4820b34f2c7dc9225c3d95
Reviewed-on: https://webrtc-review.googlesource.com/24981
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20821}
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.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: I6dd8677a65f897877fc848aefa7ab37d844e70ed
Reviewed-on: https://webrtc-review.googlesource.com/23573
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20816}
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.
The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.
Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
This is a reland of 154ee1fd8547768a49b7d67ce586ef5d3c5d9ebc
Original change's description:
> Move ulpfec, red, and flexfec codec to video engine
>
> These codecs are currently being added in the internal encoder factory.
> This means that the new injectable video codec factories will miss them.
> This CL moves adding them into the video engine so that both factory
> types will get them.
>
> This CL makes a functional change in that RED, ULPFEC, and FlexFec will
> be placed after both the internal and external codecs. Previously,
> it was placed between the internal and external codecs.
>
> Bug: webrtc:8527
> Change-Id: I5aa7a3ca674f621b17cf3aa095a225c753488e09
> Reviewed-on: https://webrtc-review.googlesource.com/22964
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20700}
TBR=brandt@webrtc.org
Bug: webrtc:8527
Change-Id: I79ced9a909fd424f1308d62e449268dcc9289538
Reviewed-on: https://webrtc-review.googlesource.com/24060
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20749}
Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).
Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20729}
This reverts commit 154ee1fd8547768a49b7d67ce586ef5d3c5d9ebc.
Reason for revert: Breaks AppRTCMobileTest on Android64 (M Nexus5X) at https://build.chromium.org/p/client.webrtc/console
Original change's description:
> Move ulpfec, red, and flexfec codec to video engine
>
> These codecs are currently being added in the internal encoder factory.
> This means that the new injectable video codec factories will miss them.
> This CL moves adding them into the video engine so that both factory
> types will get them.
>
> This CL makes a functional change in that RED, ULPFEC, and FlexFec will
> be placed after both the internal and external codecs. Previously,
> it was placed between the internal and external codecs.
>
> Bug: webrtc:8527
> Change-Id: I5aa7a3ca674f621b17cf3aa095a225c753488e09
> Reviewed-on: https://webrtc-review.googlesource.com/22964
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20700}
TBR=brandtr@webrtc.org,magjed@webrtc.org
Change-Id: I20569ae5aa4e5d794c8f7605ff5d2dd708442ae1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8527
Reviewed-on: https://webrtc-review.googlesource.com/23640
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20707}
These codecs are currently being added in the internal encoder factory.
This means that the new injectable video codec factories will miss them.
This CL moves adding them into the video engine so that both factory
types will get them.
This CL makes a functional change in that RED, ULPFEC, and FlexFec will
be placed after both the internal and external codecs. Previously,
it was placed between the internal and external codecs.
Bug: webrtc:8527
Change-Id: I5aa7a3ca674f621b17cf3aa095a225c753488e09
Reviewed-on: https://webrtc-review.googlesource.com/22964
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20700}
This will help keep ortc dependencies clean in the future, since
gn --check forces us to depend on components from which we include
headers.
cryptoparams.h moves into api/, but ortc appears to think it
should be there anyway. We could consider moving it into the ortc/ api,
but it doesn't appear to be specific to ortc.
Bug: webrtc:6828
Change-Id: Iddae438d10b5e84b2fbc52565364319e20f90613
Reviewed-on: https://webrtc-review.googlesource.com/22660
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20686}
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.
This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520
Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}
VideoEncoderSoftwareFallbackWrapper is updated to take a VideoEncoder as
argument instead relying on built-in SW codecs. The purpose is to make
VideoEncoderSoftwareFallbackWrapper more modular and not depend on
built-in SW encoders.
Bug: webrtc:7925
Change-Id: I99896f0751cfb77e01efd29c97d3bd07bdb2c7c0
Reviewed-on: https://webrtc-review.googlesource.com/22320
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20671}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
Unprefixed LOG macros will be removed on 10/11/2017, this CL just
switch some LOG macros to RTC_LOG.
TBR=magjed@webrtc.org
Bug: webrtc:8452
Change-Id: I103ba7e8a58faaa65a1cf28bd0c72a879956cc16
Reviewed-on: https://webrtc-review.googlesource.com/21960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20627}
This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239.
Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a
NullVideoDecoder that ignores calls instead.
Original change's description:
> Revert "Update internal video decoder factory to new interface"
>
> This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea.
>
> Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051
>
> Original change's description:
> > Update internal video decoder factory to new interface
> >
> > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
> > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
> > is updated to take a VideoDecoder as argument instead of a factory so it
> > can be used with external SW decoders.
> >
> > Bug: webrtc:7925
> > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
> > Reviewed-on: https://webrtc-review.googlesource.com/7301
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20597}
>
> TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/21420
> Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
> Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20605}
TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com
Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/21541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20623}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:6000, webrtc:8396
Change-Id: Ibb6b3f36f14b956c55d4edc934d101cb855b272d
Reviewed-on: https://webrtc-review.googlesource.com/18420
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20622}
This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea.
Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051
Original change's description:
> Update internal video decoder factory to new interface
>
> We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
> updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
> is updated to take a VideoDecoder as argument instead of a factory so it
> can be used with external SW decoders.
>
> Bug: webrtc:7925
> Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
> Reviewed-on: https://webrtc-review.googlesource.com/7301
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20597}
TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/21420
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20605}
We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
is updated to take a VideoDecoder as argument instead of a factory so it
can be used with external SW decoders.
Bug: webrtc:7925
Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
Reviewed-on: https://webrtc-review.googlesource.com/7301
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20597}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace095806a635411edd40fb8490a144e59e63.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
This reverts commit a7678667fc726979ca27d2429689d5735cca425d.
Reason for reland: Fix initializer list constructor.
Original change's description:
> Revert "Add helper functions for migrating to new video codec factories"
>
> This reverts commit 1c9623c70db42550d152c127e01434004087f743.
>
> Reason for revert: Breaks brace initialization:
>
> cricket::VideoDecoderParams params = { "deadbeef" };
>
> I suggest adding an initializer list constructor.
>
> Original change's description:
> > Add helper functions for migrating to new video codec factories
> >
> > This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> > convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> > webrtc::VideoEncoder and webrtc::VideoDecoder.
> >
> > The purpose is to make it as easy as possible for clients to migrate to the new
> > API and allow us to stop depending on the internal SW codecs as soon as possible.
> >
> > There still exists an ugly decoder adapter class in the video engine. The reason
> > is that we need to continue to pass in the |receive_stream_id| decoder params to
> > some legacy clients.
> >
> > Bug: webrtc:7925
> > Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> > Reviewed-on: https://webrtc-review.googlesource.com/15181
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20475}
>
> TBR=magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/17160
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20486}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
Change-Id: Ic825d133b6e1c6e5aad811ba528751dd5ed85e67
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20501}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
This reverts commit 1c9623c70db42550d152c127e01434004087f743.
Reason for revert: Breaks brace initialization:
cricket::VideoDecoderParams params = { "deadbeef" };
I suggest adding an initializer list constructor.
Original change's description:
> Add helper functions for migrating to new video codec factories
>
> This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> webrtc::VideoEncoder and webrtc::VideoDecoder.
>
> The purpose is to make it as easy as possible for clients to migrate to the new
> API and allow us to stop depending on the internal SW codecs as soon as possible.
>
> There still exists an ugly decoder adapter class in the video engine. The reason
> is that we need to continue to pass in the |receive_stream_id| decoder params to
> some legacy clients.
>
> Bug: webrtc:7925
> Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> Reviewed-on: https://webrtc-review.googlesource.com/15181
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20475}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20486}
This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
webrtc::VideoEncoder and webrtc::VideoDecoder.
The purpose is to make it as easy as possible for clients to migrate to the new
API and allow us to stop depending on the internal SW codecs as soon as possible.
There still exists an ugly decoder adapter class in the video engine. The reason
is that we need to continue to pass in the |receive_stream_id| decoder params to
some legacy clients.
Bug: webrtc:7925
Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
Reviewed-on: https://webrtc-review.googlesource.com/15181
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20475}
I want to move away from the old encoder factory interface
cricket::WebRtcEncoderFactory to the new webrtc::VideoEncoderFactory. I
created a new webrtc::SdpVideoFormat that essentially is a subset of the
cricket::VideoCodec variables. E.g. the encoder factories shouldn't have
to assign payload types to the codecs, so the payload is not part of
webrtc::SdpVideoFormat. I also didn't add the "feedback_params" that is
used in cricket::VideoCodec to webrtc::SdpVideoFormat. This is causing
problems now, because the internal encoder factory is adding flexfec
feedback params. To avoid this problem, I add these feedback params in
WebRtcVideoEngine instead, like we do for the other codecs.
Bug: webrtc:7925
Change-Id: I7c6ae8d1e1f47f3631c4804c223ec21da8d73685
Reviewed-on: https://webrtc-review.googlesource.com/15223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20444}
Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
Now decision between using SimulcastEncoderAdapter and using VP8 encoder
is postponed before codec is initialized for VP8 internal codecs. This is done
be new VP8EncoderProxy class. New error code for codec initialization is used
to signal that simulcast parameters are not supported.
Bug: webrtc:7925
Change-Id: I3a82c21bf5dfaaa7fa25350986830523f02c39d8
Reviewed-on: https://webrtc-review.googlesource.com/13980
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20419}
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.
Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
This is pretty useful if you need to debug an SCTP issue. SCTP goes
over DTLS, which is encrypted, which means you need a private key
in order to decrypt a normal packet capture. We don't log this key,
for understandable reasons. So the alternative is to turn on SCTP
verbose logging, then turn the text log into a pcap file.
NOTRY=True
TBR=zhihuang@webrtc.org
Bug: None
Change-Id: If3380d7953ea829b3ae9945326d3c820ce7cc6a3
Reviewed-on: https://webrtc-review.googlesource.com/14561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20380}
