86 Commits

Author SHA1 Message Date
guidou
716d7ac5c1 Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
Reason for revert:
Suspect of breaking Chrome FYI bots.

See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder

Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
 #include "third_party/webrtc/video_encoder.h"
                                              ^

Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570

TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881

Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
2017-04-03 16:15:52 +00:00
ilnik
c42f540570 Move video_encoder.h and video_decoder.h to /api and create GN targets for them
BUG=webrtc:5881
# Because PRESUBMIT ignores LINT blacklist for moved files and these
# headers have some not easy to resolve issues.
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2780943003
Cr-Commit-Position: refs/heads/master@{#17511}
2017-04-03 15:37:32 +00:00
solenberg
9a5f032227 Remove VoEHardware interface usage.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2737633002
Cr-Commit-Position: refs/heads/master@{#17248}
2017-03-15 13:14:12 +00:00
kjellander
a2ef4f94e4 Enable GN check for webrtc/media
BUG=webrtc:6828, webrtc:7245
NOTRY=True

Review-Url: https://codereview.webrtc.org/2716143002
Cr-Commit-Position: refs/heads/master@{#17067}
2017-03-06 14:04:55 +00:00
solenberg
76377c55b7 Remove usage of VoEAudioProcessing from WVoE/MC.
Calling APM and TransmitMixer directly instead.

BUG=webrtc:4690
TBR=peah@webrtc.org

Review-Url: https://codereview.webrtc.org/2681033010
Cr-Commit-Position: refs/heads/master@{#16734}
2017-02-21 08:54:31 +00:00
ossu
11bfc53cd9 Fixed a couple of build-flag dependent tests of webrtcvoiceengine.
The codecs expected by HasCorrectCodecs now depends which codecs were
enabled by build flags.

SendSideBweWithOverheadTest.MinAndMaxBitrate now expects different
values for min bitrate depending on if we support 120 ms frames for
Opus.

BUG=b/35415435

Review-Url: https://codereview.webrtc.org/2691343008
Cr-Commit-Position: refs/heads/master@{#16643}
2017-02-16 13:37:06 +00:00
minyue
2e03c66119 Adding build switch for Opus that supports 120ms ptime.
BUG=webrtc:7097

TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o

Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
2017-02-02 01:31:11 +00:00
mbonadei
9aa3f0a200 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reason for revert:
Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file)

Original issue's description:
> Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
>
> Reason for revert:
> This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio
>
> Original issue's description:
> > Moving webrtc.gni up one level from build/
> >
> > BUG=webrtc:7030
> >
> > Review-Url: https://codereview.webrtc.org/2651543003
> > Cr-Commit-Position: refs/heads/master@{#16241}
> > Committed: 35a32700fc
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2657563002
> Cr-Commit-Position: refs/heads/master@{#16244}
> Committed: 69dc7dbe24

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2654773002
Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 14:58:22 +00:00
mbonadei
69dc7dbe24 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
Reason for revert:
This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio

Original issue's description:
> Moving webrtc.gni up one level from build/
>
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2651543003
> Cr-Commit-Position: refs/heads/master@{#16241}
> Committed: 35a32700fc

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2657563002
Cr-Commit-Position: refs/heads/master@{#16244}
2017-01-24 13:14:35 +00:00
mbonadei
35a32700fc Moving webrtc.gni up one level from build/
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2651543003
Cr-Commit-Position: refs/heads/master@{#16241}
2017-01-24 12:49:35 +00:00
pbos
2197e91860 Remove dead code for GtkVideoRenderer.
Pulls in unnecessary GTK dependencies that breaks the chromium GTK3
build. This removes the last of webrtc/media/devices.

BUG=chromium:668446

Review-Url: https://codereview.webrtc.org/2646793008
Cr-Commit-Position: refs/heads/master@{#16198}
2017-01-21 02:13:07 +00:00
noahric
be850e1b1d Clear out cached codecs when calculating new codec lists.
Without this, every time WebRtcVideoEngine2 calls supported_codecs(),
the codec list grows.

BUG=webrtc:7020

Review-Url: https://codereview.webrtc.org/2639423006
Cr-Commit-Position: refs/heads/master@{#16178}
2017-01-20 09:07:26 +00:00
nisse
61f31ee376 Delete unused rtpdump code in media/base.
Reading and writing RTP files is implemented elsewhere,
in test/rtp_file_reader.cc and test/rtp_file_writer.cc;
that code is untouched by this cl.

BUG=webrtc:6974

Review-Url: https://codereview.webrtc.org/2633453002
Cr-Commit-Position: refs/heads/master@{#16046}
2017-01-13 13:55:08 +00:00
michaelt
566d820e00 Update smoothed bitrate.
BUG=webrtc:6443

Review-Url: https://codereview.webrtc.org/2546493002
Cr-Commit-Position: refs/heads/master@{#16036}
2017-01-12 18:17:38 +00:00
deadbeef
953c2cea5e Reland of: Separating SCTP code from BaseChannel/MediaChannel.
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.

SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.

Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
  processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.

BUG=None

Review-Url: https://codereview.webrtc.org/2564333002
Cr-Original-Commit-Position: refs/heads/master@{#15906}
Committed: 67b3bbe639
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15973}
2017-01-09 22:53:41 +00:00
deadbeef
c0dad89bed Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
Reason for revert:
Hitting DCHECK in chromium's WebrtcTransportTest.TerminateDataChannel and WebrtcTransportTest.DataStreamLate. Will investigate and reland.

Original issue's description:
> Separating SCTP code from BaseChannel/MediaChannel.
>
> The BaseChannel code is geared around RTP; the presence of media engines,
> send and receive streams, SRTP, SDP directional attribute negotiation, etc.
> It doesn't make sense to use it for SCTP as well. This separation should make
> future work both on BaseChannel and the SCTP code paths easier.
>
> SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
> directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
> doesn't get confused with webrtc::DataChannel any more.
>
> Beyond just moving code around, some consequences of this CL:
> - We'll now stop using the worker thread for SCTP. Packets will be
>   processed right on the network thread instead.
> - The SDP directional attribute is ignored, as it's supposed to be.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2564333002
> Cr-Commit-Position: refs/heads/master@{#15906}
> Committed: 67b3bbe639

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2614813003
Cr-Commit-Position: refs/heads/master@{#15908}
2017-01-05 04:28:21 +00:00
deadbeef
e9bbde5830 Removing hybriddataengine.h from BUILD.gn
It was deleted in this CL, but the BUILD.gn file wasn't updated:
https://codereview.webrtc.org/2564333002/

BUG=None
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2612963002
Cr-Commit-Position: refs/heads/master@{#15907}
2017-01-05 03:52:10 +00:00
deadbeef
67b3bbe639 Separating SCTP code from BaseChannel/MediaChannel.
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.

SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.

Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
  processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.

BUG=None

Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15906}
2017-01-05 02:38:02 +00:00
deadbeef
40610e24ce Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
This allows building without SCTP support (and even building/running
tests). The "HAVE_SCTP" define has been functional for a while, but there
wasn't any easy way to turn it on/off.

NOTRY=True
BUG=webrtc:6933

Review-Url: https://codereview.webrtc.org/2593313002
Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 18:53:38 +00:00
magjed
a3f2d30182 Remove media/base header files from rtc_media target
Chromium has now been updated, so we can remove the base headers from
rtc_media.

BUG=None

Review-Url: https://codereview.webrtc.org/2590813002
Cr-Commit-Position: refs/heads/master@{#15712}
2016-12-20 13:26:59 +00:00
magjed
dd40702357 Move VideoDecoder::Create() logic to separate internal video decoder factory
The goal with this CL is to move implementation details out from the
webrtc root (webrtc/video_decoder.h) to simplify the dependency graph.
Another goal is to streamline the creation of VideoDecoders in
webrtcvideoengine2.cc; it will now have two factories of the same
WebRtcVideoDecoderFactory type, one internal and one external.

Specifically, this CL:
 * Removes webrtc::VideoDecoder::DecoderType and use webrtc::VideoCodecType
   instead.
 * Removes 'static VideoDecoder* Create(DecoderType codec_type)' and
   moves the create function to the internal decoder factory instead.
 * Removes video_decoder.cc. webrtc::VideoDecoder is now just an
   interface without any static functions.

BUG=webrtc:6743

Review-Url: https://codereview.webrtc.org/2521203002
Cr-Commit-Position: refs/heads/master@{#15350}
2016-12-01 08:27:35 +00:00
aleloi
a8eb756a34 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.

Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.

transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.

NOTRY=True

BUG=webrtc:5589, webrtc:5878, webrtc:6785

Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 15:02:19 +00:00
magjed
0928a3cf0f Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ )
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}
2016-11-25 08:40:26 +00:00
magjed
f6acc2a710 Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
The class VideoDecoderSoftwareFallbackWrapper is an implementation
detail of webrtc/media/engine/webrtcvideoengine2.cc and should not be
directly under webrtc/video_decoder.h. The main purpose is to improve
the dependency graph in WebRTC so that VideoDecoderSoftwareFallbackWrapper
can depend on cricket::VideoCodec.

The test for VideoDecoderSoftwareFallbackWrapper is also moved from
webrtc/video/video_decoder_unittest.cc to
webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc.

BUG=webrtc:6743
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2518263003
Cr-Commit-Position: refs/heads/master@{#15180}
2016-11-22 09:43:06 +00:00
magjed
509e4fe8e6 Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
Reason for revert:
The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change:
   function removeVideoCodec(offerSdp) {
-    offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
-                                'a=rtpmap:100 XVP8/90000\r\n');
+    offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/,
+                                'a=rtpmap:$1 XVP8/90000\r\n');
     return offerSdp;
   }

Original issue's description:
> Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
>
> Reason for revert:
> Breaks chromium.fyi test:
> WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
>
> Original issue's description:
> > Stop using hardcoded payload types for video codecs
> >
> > This CL stops using hardcoded payload types for different video codecs
> > and will dynamically assign them payload types incrementally from 96 to
> > 127 instead.
> >
> > This CL:
> >  * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> >    webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> >    internally supported software codecs instead. The purpose is to
> >    streamline the payload type assignment in webrtcvideoengine2.cc which
> >    will now have two encoder factories of the same
> >    WebRtcVideoEncoderFactory type; one internal and one external.
> >  * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> >    instead.
> >  * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> >    moves the create function to the internal encoder factory instead.
> >  * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> >    interface without any static functions.
> >  * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> >    the internal and external codecs and assigns them payload types
> >    incrementally from 96 to 127.
> >  * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> >    what payload types will be used.
> >
> > BUG=webrtc:6677,webrtc:6705
> > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> > Cr-Commit-Position: refs/heads/master@{#15135}
>
> TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6677,webrtc:6705
>
> Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591
> Cr-Commit-Position: refs/heads/master@{#15140}

TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705

Review-Url: https://codereview.webrtc.org/2511933002
Cr-Commit-Position: refs/heads/master@{#15148}
2016-11-18 09:34:14 +00:00
magjed
eacbaea920 Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
Reason for revert:
Breaks chromium.fyi test:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec

Original issue's description:
> Stop using hardcoded payload types for video codecs
>
> This CL stops using hardcoded payload types for different video codecs
> and will dynamically assign them payload types incrementally from 96 to
> 127 instead.
>
> This CL:
>  * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
>    webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
>    internally supported software codecs instead. The purpose is to
>    streamline the payload type assignment in webrtcvideoengine2.cc which
>    will now have two encoder factories of the same
>    WebRtcVideoEncoderFactory type; one internal and one external.
>  * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
>    instead.
>  * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
>    moves the create function to the internal encoder factory instead.
>  * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
>    interface without any static functions.
>  * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
>    the internal and external codecs and assigns them payload types
>    incrementally from 96 to 127.
>  * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
>    what payload types will be used.
>
> BUG=webrtc:6677,webrtc:6705
> R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> Cr-Commit-Position: refs/heads/master@{#15135}

TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705

Review-Url: https://codereview.webrtc.org/2513633002
Cr-Commit-Position: refs/heads/master@{#15140}
2016-11-17 16:52:06 +00:00
magjed
0d0d7531b5 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

Review-Url: https://codereview.webrtc.org/2508163002
Cr-Commit-Position: refs/heads/master@{#15139}
2016-11-17 15:51:35 +00:00
magjed
aae7e7cf35 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

Review-Url: https://codereview.webrtc.org/2471573003
Cr-Commit-Position: refs/heads/master@{#15137}
2016-11-17 15:29:37 +00:00
Magnus Jedvert
42043b9587 Stop using hardcoded payload types for video codecs
This CL stops using hardcoded payload types for different video codecs
and will dynamically assign them payload types incrementally from 96 to
127 instead.

This CL:
 * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
   webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
   internally supported software codecs instead. The purpose is to
   streamline the payload type assignment in webrtcvideoengine2.cc which
   will now have two encoder factories of the same
   WebRtcVideoEncoderFactory type; one internal and one external.
 * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
   instead.
 * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
   moves the create function to the internal encoder factory instead.
 * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
   interface without any static functions.
 * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
   the internal and external codecs and assigns them payload types
   incrementally from 96 to 127.
 * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
   what payload types will be used.

BUG=webrtc:6677,webrtc:6705
R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2493133002 .

Cr-Commit-Position: refs/heads/master@{#15135}
2016-11-17 15:08:47 +00:00
aleloi
10111bc495 Passed AudioMixer to AudioState::Config.
This is a refactoring change in preparation for enabling AudioMixer
with the goal to have a small CL as possible for passing audio through
the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/
enables the mixer.

An object of class AudioState is shared across different webrtc audio
connections. It is created in tests and in
WebRTCVoiceEngine. AudioState is constructed by passing a Config
struct, where one argument is scoped_refptr<AudioMixer>.

Populating this field has now been mandatory. Tests and
WebRTCVoiceEngine create and pass either a AudioMixerImpl.
WebRTCVoiceEngine passes a real AudioMixer, which is
currently unused.

An alternative would have tests pass a mocked audio mixer. We
chose not to do that, because we believe that tests should use
the real thing unless there are reasons against it. Construction
time is not an issue, because the real mixer is relatively
lightweight.

We couldn't find a way to test any mixer-related changes in AudioState
before the mixes is connected. The next dependent CL
https://codereview.webrtc.org/2436033002/ contains unit tests for
mixer usage.

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2469743002
Cr-Commit-Position: refs/heads/master@{#15134}
2016-11-17 14:48:56 +00:00
magjed
614d5b78d6 Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/
The class VideoEncoderSoftwareFallbackWrapper is an implementation
detail of webrtc/media/engine/webrtcvideoengine2.cc and should not be
directly under webrtc/video_encoder.h. The main purpose is to improve
the dependency graph in WebRTC so that VideoEncoderSoftwareFallbackWrapper
can depend on cricket::VideoCodec.

The test for VideoEncoderSoftwareFallbackWrapper is also moved from
webrtc/video/video_encoder_unittest.cc to
webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc.

BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2484863009
Cr-Commit-Position: refs/heads/master@{#15085}
2016-11-15 14:31:01 +00:00
aleloi
5de52fd38e Created a mocked AudioTransport.
There are currently two nearly identical classes called
MockAudioTransport defined in two unit tests:
android/audio_transport_unittest.cc and
/ios/audio_transport_unittest_ios.cc

This change defines a common mocked AudioTransport. The two current
mocks are rewritten to use the common one. A GN target is created for
this mock and MockAudioDevice.

This change will allow to provide a mocked AudioTransport to
AudioState in a dependent CL https://codereview.webrtc.org/2454373002/

BUG=webrtc:6346
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2493483002
Cr-Commit-Position: refs/heads/master@{#15010}
2016-11-10 09:05:39 +00:00
kjellander
71a1b61c4f WebRTC: Fix and enable -Woverloaded-virtual warnings.
Essentially applying the same change as in
https://codereview.webrtc.org/2023413002 in more locations.

There's only one change affecting production code: enabling the warning
for webrtc/media:rtc_media. The rest are test changes.

With these changes, the only place the warning is disabled is in the Windows
implementation of webrtc/modules/video_capture:video_capture_internal_impl,
which is harder to fix, since it relies on sample code from the Windows SDK.

BUG=webrtc:6653
NOTRY=True

Review-Url: https://codereview.webrtc.org/2468093004
Cr-Commit-Position: refs/heads/master@{#14938}
2016-11-07 09:18:14 +00:00
nisse
a144be384f Delete videorendererfactory.h and cricket::GdiVideoRenderer.
Ultimately, all of webrtc/media/devices should be deleted, since it is
unused in webrtc and has no unit tests. But for the time being we
need to keep cricket::GtkVideoRenderer since there are a few
applications depending on it.

BUG=webrtc:5924

Review-Url: https://codereview.webrtc.org/2460793002
Cr-Commit-Position: refs/heads/master@{#14835}
2016-10-31 07:59:54 +00:00
magjed
1e45cc6ee0 Replace WebRtcVideoEncoderFactory::VideoCodec with cricket::VideoCodec
This CL introduces two new functions to the WebRtcVideoEncoderFactory
interface based on cricket::VideoFormat instead of
WebRtcVideoEncoderFactory::VideoCodec. The functions are:
WebRtcVideoEncoderFactory::CreateVideoEncoder() and
WebRtcVideoEncoderFactory::supported_codecs(). In order to make a smooth
transition to the new interface, the old functions are kept, and default
implementations are provided for both the old and new functions so that
external clients can switch from the old to the new functions in peace.
The default implementations will just convert between
cricket::VideoFormat and WebRtcVideoEncoderFactory::VideoCodec. Once all
external clients have updated their code, the plan is to remove the old
functions and all default implementations to make
WebRtcVideoEncoderFactory a pure interface again.

BUG=webrtc:6402,webrtc:6337

Review-Url: https://codereview.webrtc.org/2449993003
Cr-Commit-Position: refs/heads/master@{#14826}
2016-10-28 14:43:52 +00:00
kjellander
af1ae310ef Remove dead dependencies on xmllite and xmpp.
media/ and p2p/ doesn't actually depend on these anymore.

BUG=webrtc:5539
NOTRY=True

Review-Url: https://codereview.webrtc.org/2447533003
Cr-Commit-Position: refs/heads/master@{#14761}
2016-10-25 08:27:17 +00:00
nisse
74097fd3f5 Delete unused file screencastid.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2433913003
Cr-Commit-Position: refs/heads/master@{#14757}
2016-10-25 07:17:52 +00:00
nisse
dda1e60153 Reland of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2441493003/ )
Reason for revert:
#include of mediacommon.h now deleted in the offending downstream app.

Original issue's description:
> Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ )
>
> Reason for revert:
> Turned out this broke a downstream app. Reverting while investigating.
>
> Original issue's description:
> > Delete unused file mediacommon.h.
> >
> > BUG=None
> >
> > Committed: https://crrev.com/a34e796a1b59a1068c77a3e36d2a24fd37c08afc
> > Cr-Commit-Position: refs/heads/master@{#14689}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Committed: https://crrev.com/ca20e7cfed301982986a4da23c819b4e50b5a6e9
> Cr-Commit-Position: refs/heads/master@{#14697}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2426393006
Cr-Commit-Position: refs/heads/master@{#14718}
2016-10-21 06:57:18 +00:00
nisse
ca20e7cfed Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ )
Reason for revert:
Turned out this broke a downstream app. Reverting while investigating.

Original issue's description:
> Delete unused file mediacommon.h.
>
> BUG=None
>
> Committed: https://crrev.com/a34e796a1b59a1068c77a3e36d2a24fd37c08afc
> Cr-Commit-Position: refs/heads/master@{#14689}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2441493003
Cr-Commit-Position: refs/heads/master@{#14697}
2016-10-20 09:53:06 +00:00
nisse
a34e796a1b Delete unused file mediacommon.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2437703002
Cr-Commit-Position: refs/heads/master@{#14689}
2016-10-20 07:45:02 +00:00
nisse
09347858f7 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ )
This cl now makes cricket::VideoFrame and cricket::WebRtcVideoFrame aliases for webrtc::VideoFrame.

Reason for revert:
Fixing backwards compatibility issues.

Original issue's description:
> Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
>
> Reason for revert:
> Breaks compile for Chromium builds:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142
>
> FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
> ../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
>                  weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
>                                              ~~~~~ ^
> 1 error generated.
>
> Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.
>
> Original issue's description:
> > Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> > all methods but a few constructors. And similarly for the
> > subclass cricket::WebRtcVideoFrame.
> >
> > TBR=tkchin@webrtc.org  # Added an include line
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> > Cr-Commit-Position: refs/heads/master@{#14576}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d36dd499c8f253cbcf37364c2a070c2e8c7100e9
> Cr-Commit-Position: refs/heads/master@{#14583}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2411953002
Cr-Commit-Position: refs/heads/master@{#14678}
2016-10-19 07:30:35 +00:00
kjellander
e40a7ee007 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
These suppressions are causing GN errors when Chromium targets are depending
directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004)

BUG=webrtc:4256
NOTRY=True

Review-Url: https://codereview.webrtc.org/2408133008
Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-17 06:56:20 +00:00
kjellander
d36dd499c8 Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
Reason for revert:
Breaks compile for Chromium builds:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142

FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
                 weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
                                             ~~~~~ ^
1 error generated.

Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.

Original issue's description:
> Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> all methods but a few constructors. And similarly for the
> subclass cricket::WebRtcVideoFrame.
>
> TBR=tkchin@webrtc.org  # Added an include line
> BUG=webrtc:5682
>
> Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> Cr-Commit-Position: refs/heads/master@{#14576}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
NOTRY=True
NOPRESUBMIT=True
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2402853002
Cr-Commit-Position: refs/heads/master@{#14583}
2016-10-09 05:21:40 +00:00
nisse
dda6ec008a Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
all methods but a few constructors. And similarly for the
subclass cricket::WebRtcVideoFrame.

TBR=tkchin@webrtc.org  # Added an include line
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2315663002
Cr-Commit-Position: refs/heads/master@{#14576}
2016-10-07 15:16:59 +00:00
nisse
4e8baf46a7 Delete unused code, cpuinfo.{h,cc}.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2385053004
Cr-Commit-Position: refs/heads/master@{#14538}
2016-10-06 06:44:45 +00:00
nisse
f5297a019e Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ )
Reason for revert:
Downstream code is being fixed.

Original issue's description:
> Revert of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #9 id:160001 of https://codereview.webrtc.org/2262443003/ )
>
> Reason for revert:
> Breaks downstream testcode, still using CapturedFrame.
>
> Original issue's description:
> > Delete VideoFrameFactory, CapturedFrame, and related code.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/66ac50e58c790624d51ede10ae438cbadbca9d2e
> > Cr-Commit-Position: refs/heads/master@{#14315}
>
> TBR=pthatcher@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/66492210e57ee8efce2ad4d45a8781df1fcaa5e3
> Cr-Commit-Position: refs/heads/master@{#14320}

TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2370993003
Cr-Commit-Position: refs/heads/master@{#14453}
2016-09-30 08:34:35 +00:00
kjellander
660312b0c6 Enable //build/config/clang:extra_warnings for rtc_media
The code compiles fine with this config removed.

BUG=webrtc:6323
NOTRY=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2369893002
Cr-Commit-Position: refs/heads/master@{#14381}
2016-09-26 13:11:57 +00:00
nisse
2a27b0a478 Delete unused class FakeScreenCapturerFactory.
Last use of the header file probably deleted in cl
https://codereview.webrtc.org/1903393004

BUG=None

Review-Url: https://codereview.webrtc.org/2353043003
Cr-Commit-Position: refs/heads/master@{#14370}
2016-09-23 11:42:49 +00:00
kjellander
b62dbbe985 GN: Change rtc_source_set targets --> rtc_static_library
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).

After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()

See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.

NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.

BUG=webrtc:6410, chromium:630755

Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 07:38:58 +00:00
charujain
09baefeaf9 GN: Remove non-existing header file: media/base/videorenderer.h
This header was removed in https://codereview.webrtc.org/1819103003
but the BUILD.gn file was not updated.

BUG=webrtc:5949
NOTRY=True
TBR=nisse@webrtc.org

Review-Url: https://codereview.webrtc.org/2359113002
Cr-Commit-Position: refs/heads/master@{#14349}
2016-09-22 11:51:57 +00:00