Commit Graph

267 Commits

Author SHA1 Message Date
1e23461d5e Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
Reason for revert:
Broke chromium FYI bot because the chromium mock PC overrides the method whose signature is changing.

Also broke a downstream internal test, which I need to investigate further.

Original issue's description:
> Adding error output param to SetConfiguration, using new RTCError type.
>
> Most notably, will return "INVALID_MODIFICATION" if a field in the
> configuration was modified and modification of that field isn't supported.
>
> Also changing RTCError to a class that wraps an enum type, because it will
> eventually need to hold other information (like SDP line number), to match
> the RTCError that was recently added to the spec:
> https://github.com/w3c/webrtc-pc/pull/850
>
> BUG=webrtc:6916
>
> Review-Url: https://codereview.webrtc.org/2587133004
> Cr-Commit-Position: refs/heads/master@{#15777}
> Committed: 7a5fa6cd61

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2600813002
Cr-Commit-Position: refs/heads/master@{#15778}
2016-12-24 09:43:32 +00:00
7a5fa6cd61 Adding error output param to SetConfiguration, using new RTCError type.
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.

Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850

BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#15777}
2016-12-24 08:47:59 +00:00
ffb865f3e0 Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ )
Reason for revert:
Breaks downstream build.

Original issue's description:
> Delete unused code from systeminfo.
>
> BUG=webrtc:6906
>
> Review-Url: https://codereview.webrtc.org/2578323005
> Cr-Commit-Position: refs/heads/master@{#15655}
> Committed: 617ca316e9

TBR=perkj@webrtc.org,kthelgason@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6906

Review-Url: https://codereview.webrtc.org/2584563004 .
Cr-Commit-Position: refs/heads/master@{#15660}
2016-12-17 00:48:28 +00:00
617ca316e9 Delete unused code from systeminfo.
BUG=webrtc:6906

Review-Url: https://codereview.webrtc.org/2578323005
Cr-Commit-Position: refs/heads/master@{#15655}
2016-12-16 14:07:03 +00:00
df2ceb88a8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
Reason for revert:
Fixing perf tests.

Original issue's description:
> Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
>
> Reason for revert:
> Crashes perf tests, e.g.,
>
> ./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
>
> dies with an assert related to rtc::Optional.
>
> Original issue's description:
> > Delete VideoFrame default constructor, and IsZeroSize method.
> >
> > This ensures that the video_frame_buffer method never can return a
> > null pointer.
> >
> > BUG=webrtc:6591
> >
> > Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> > Cr-Commit-Position: refs/heads/master@{#15574}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6591
>
> Committed: https://crrev.com/0989fbcad2ca4eb5805a77e8ebfefd3af06ade23
> Cr-Commit-Position: refs/heads/master@{#15597}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574183002
Cr-Commit-Position: refs/heads/master@{#15633}
2016-12-15 14:30:00 +00:00
0989fbcad2 Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
Reason for revert:
Crashes perf tests, e.g.,

./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'

dies with an assert related to rtc::Optional.

Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
2016-12-14 10:06:49 +00:00
95aa96465d Support external audio mixer in webrtc 2.
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.

This CL has rewritten based on reverted CL
https://codereview.chromium.org/2539213003/
The only difference is that
  static MediaEngineInterface* Create(
      webrtc::AudioDeviceModule* adm,
      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
          audio_decoder_factory,
      WebRtcVideoEncoderFactory* video_encoder_factory,
      WebRtcVideoDecoderFactory* video_decoder_factory);
in media/engine/webrtcmediaengine.h is kept in this CL instead of
replaced for backward compatibility.

BUG=webrtc:6457

Review-Url: https://codereview.webrtc.org/2570993002
Cr-Commit-Position: refs/heads/master@{#15580}
2016-12-13 22:06:35 +00:00
bfcf561923 Delete VideoFrame default constructor, and IsZeroSize method.
This ensures that the video_frame_buffer method never can return a
null pointer.

BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
2016-12-13 14:08:39 +00:00
39ce11f7f6 Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ )
Reason for revert:
A interface change broke some downstream code in google3.

Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}

TBR=solenberg@webrtc.org,aleloi@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6457

Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
2016-12-13 01:07:00 +00:00
f6bcac59e8 Support external audio mixer in webrtc.
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.

BUG=webrtc:6457

Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
2016-12-13 00:25:16 +00:00
c8474178d6 Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
Reason for revert:
There was a bug in the implementation where the adapter could get stuck at really low resolutions. That has now been fixed.

Original issue's description:
> Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
>
> Reason for revert:
> Issue discovered with scaling back up.
>
> Original issue's description:
> > Add ability to scale to arbitrary factors
> >
> > This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
> >
> > BUG=webrtc:6837
> >
> > Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> > Cr-Commit-Position: refs/heads/master@{#15469}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6837
>
> Committed: https://crrev.com/7722a4cc8d31e5e924e9e6c5c97412ce8bbbe59d
> Cr-Commit-Position: refs/heads/master@{#15470}

R=magjed@webrtc.org
BUG=webrtc:6837,webrtc:6848

Review-Url: https://codereview.webrtc.org/2558243003
Cr-Commit-Position: refs/heads/master@{#15485}
2016-12-08 16:04:58 +00:00
7722a4cc8d Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
Reason for revert:
Issue discovered with scaling back up.

Original issue's description:
> Add ability to scale to arbitrary factors
>
> This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
>
> BUG=webrtc:6837
>
> Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> Cr-Commit-Position: refs/heads/master@{#15469}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837

Review-Url: https://codereview.webrtc.org/2557323002
Cr-Commit-Position: refs/heads/master@{#15470}
2016-12-08 10:18:31 +00:00
710c335d78 Add ability to scale to arbitrary factors
This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.

BUG=webrtc:6837

Review-Url: https://codereview.webrtc.org/2555483005
Cr-Commit-Position: refs/heads/master@{#15469}
2016-12-08 10:12:37 +00:00
hta
b39db841b6 Refactoring: Declare cricket::Codec constructors protected.
This makes it obvious that cricket::Codec should not be
instantiated; only subclasses should be instantiated.

BUG=none

Review-Url: https://codereview.webrtc.org/2546363002
Cr-Commit-Position: refs/heads/master@{#15468}
2016-12-08 09:50:52 +00:00
f515ab8c3f Moved call.h and most of api/call/* into call/
BUG=webrtc:6716

Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 12:53:04 +00:00
ebbe4f2ed5 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
BUG=b/31996729

Review-Url: https://codereview.webrtc.org/2539813003
Cr-Commit-Position: refs/heads/master@{#15449}
2016-12-06 18:45:47 +00:00
hta
9aa96889a3 Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
Reason for revert:
Fixed timeouts in slow tests

Original issue's description:
> Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
>
> Reason for revert:
> Failures on the Linux Memcheck bot
>
> Original issue's description:
> > This approach passes packetization mode to the encoder as part of
> > a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
> >
> > BUG=600254
> >
> > Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> > Cr-Commit-Position: refs/heads/master@{#15437}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=600254
>
> Committed: https://crrev.com/243a0a7a7fd6b5da1e32df31f1bfbb6a68dc09f3
> Cr-Commit-Position: refs/heads/master@{#15441}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558463002
Cr-Commit-Position: refs/heads/master@{#15445}
2016-12-06 13:36:13 +00:00
hta
243a0a7a7f Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
Reason for revert:
Failures on the Linux Memcheck bot

Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
2016-12-06 12:22:05 +00:00
hta
e59647b991 This approach passes packetization mode to the encoder as part of
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.

BUG=600254

Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
2016-12-06 10:22:54 +00:00
b49fc142e3 RtpDataEngine, FindCodecByName: Don't reassign codecs
BUG=None

Review-Url: https://codereview.webrtc.org/2541583003
Cr-Commit-Position: refs/heads/master@{#15327}
2016-11-30 12:52:10 +00:00
80ed35e21c Implement periodic bandwidth probing in application-limited region.
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.

BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
2016-11-28 21:11:24 +00:00
ffc61181d8 Don't cache video codec list in VideoEngine2.
A WebRtcVideoEngine2 object seems to be reused between PeerConnections,
which means that the field trial added in
https://codereview.webrtc.org/2511703002/ may not activate/deactivate
as intended between calls. This CL removes the caching of video codecs,
which gets rid of this problem.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2521393004
Cr-Commit-Position: refs/heads/master@{#15265}
2016-11-28 14:02:28 +00:00
5dfac56813 Keep all codec parameters in VideoReceiveStream::Decoder
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}
2016-11-25 11:56:41 +00:00
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
e2b1501101 Start probes only after network is connected.
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}
2016-11-23 00:08:37 +00:00
10165ab8e7 Unify VideoCodecType to/from string functionality
BUG=None

Review-Url: https://codereview.webrtc.org/2509273002
Cr-Commit-Position: refs/heads/master@{#15200}
2016-11-22 18:17:04 +00:00
9688e384e6 Add support for FEC-FR semantics in StreamParams.
This allows us to associate a FlexFEC SSRC with a
media SSRC in the SDP.

BUG=webrtc:5654
R=magjed@webrtc.org
CC=stefan@webrtc.org, perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/2503403004
Cr-Commit-Position: refs/heads/master@{#15177}
2016-11-22 08:59:53 +00:00
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
1acfbd22cc Expose RtpCodecParameters to VoiceMediaInfo stats.
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].

Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.

[1] https://w3c.github.io/webrtc-stats/#codec-dict*

BUG=chromium:659117

Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
2016-11-18 07:43:39 +00:00
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
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
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
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
725e484e33 Use different RTX payload types for different H264 profiles
This CL is a quick fix to unblock H264 High Profile. This CL is supposed
to be superseded by a proper fix of
https://bugs.chromium.org/p/webrtc/issues/detail?id=6705 like
https://codereview.webrtc.org/2493133002/.

BUG=webrtc:6677

Review-Url: https://codereview.webrtc.org/2497773003
Cr-Commit-Position: refs/heads/master@{#15099}
2016-11-16 08:48:21 +00:00
906c5dc6b7 Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ )
Reason for revert:
It broke downstream test.

Original issue's description:
> Start probes only after network is connected.
>
> 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
>
> Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
> Cr-Commit-Position: refs/heads/master@{#15094}

TBR=philipel@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2504783002
Cr-Commit-Position: refs/heads/master@{#15098}
2016-11-15 22:39:09 +00:00
5c99c76255 Start probes only after network is connected.
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

Review-Url: https://codereview.webrtc.org/2458863002
Cr-Commit-Position: refs/heads/master@{#15094}
2016-11-15 20:25:37 +00:00
b829d9f2ee Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms.
BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2493753002
Cr-Commit-Position: refs/heads/master@{#15079}
2016-11-15 10:34:54 +00:00
a65704b5c9 Expose RtpCodecParameters to VideoMediaInfo stats.
Payload type -> RtpCodecParameters maps added for sender and receiver
side. It contains information that will be needed for RTCCodecStats[1]
dictionaries.

Video[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VideoMediaInfo.

A similar change should be made for VoiceMediaInfo and
Voice[Sender/Receiver]Info.

[1] https://w3c.github.io/webrtc-stats/#codec-dict*

BUG=chromium:659117

Review-Url: https://codereview.webrtc.org/2484193002
Cr-Commit-Position: refs/heads/master@{#15060}
2016-11-14 10:28:20 +00:00
f823ededce Negotiate H264 profiles in SDP
This CL will start to distinguish H264 profiles during SDP negotiation.
We currently don't look at the H264 profile at all and assume they are
all Constrained Baseline Level 3.1. This CL will start to check profiles
for equality when matching, and will generate the correct answer H264
level.

Each local supported H264 profile needs to be listed explicitly in the
list of local supported codecs, even if they are redundant. For example,
Baseline profile should be listed explicitly even though another profile
that is a superset of Baseline is also listed. The reason for this is to
simplify the code and avoid profile intersection during matching. So
VideoCodec::Matches will check for profile equality, and not check if
one codec is a subset of the other. This also leads to the nice property
that VideoCodec::Matches is symmetric, i.e. iif a.Matches(b) then
b.Matches(a).

BUG=webrtc:6337
TBR=tkchin@webrtc.org

Review-Url: https://codereview.webrtc.org/2483173002
Cr-Commit-Position: refs/heads/master@{#15051}
2016-11-12 17:53:08 +00:00
b05fa2466a Optimize FindCodecById and ReferencedCodecsMatch
These functions currently copy cricket::Codec classes by value which is
expensive since they contain e.g. std::map<std::string, std::string>
containers with parameters. This CL avoids copying them altogether.

BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2493733003
Cr-Commit-Position: refs/heads/master@{#15040}
2016-11-11 12:00:20 +00:00
acd935b540 Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
Reason for revert:
Relanding after known downstream breakages have been fixed.

Original issue's description:
> Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
>
> Reason for revert:
> Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
>
> Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
>
> Original issue's description:
> > Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
> >
> > Replaced with webrtc::VideoFrame.
> >
> > TBR=mflodman@webrtc.org
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> > Cr-Commit-Position: refs/heads/master@{#14885}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@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/7341ab8e2505c9763d208e069bda269018357e7d
> Cr-Commit-Position: refs/heads/master@{#14886}

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

Review-Url: https://codereview.webrtc.org/2487633002
Cr-Commit-Position: refs/heads/master@{#15039}
2016-11-11 11:55:19 +00:00
3cf8ece954 Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
Reason for revert:
This CL probably broke Chromium FYI.

Original issue's description:
> Stop caching supported codecs in WebRtcVideoEngine2
>
> We currently cache the result of GetSupportedCodecs in a member variable
> |video_codecs_| in WebRtcVideoEngine2. This means we need to keep
> |video_codecs_| and the result of GetSupportedCodecs in sync, which is
> error prone. It's simpler to just call GetSupportedCodecs when we need
> it, and we actually end up making fewer calls, so it's faster as well.
> This CL also returns all std::vectors by-value instead of by-ref. Move
> semantic together with in-place filtering of codecs actually end up with
> fewer copies, and it's also simpler to not return references.
>
> BUG=webrtc:6337
>
> Committed: https://crrev.com/9f71ec5a3e3175751f4475b126cfda89767363f2
> Cr-Commit-Position: refs/heads/master@{#15007}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2489173004
Cr-Commit-Position: refs/heads/master@{#15014}
2016-11-10 11:36:57 +00:00
9f71ec5a3e Stop caching supported codecs in WebRtcVideoEngine2
We currently cache the result of GetSupportedCodecs in a member variable
|video_codecs_| in WebRtcVideoEngine2. This means we need to keep
|video_codecs_| and the result of GetSupportedCodecs in sync, which is
error prone. It's simpler to just call GetSupportedCodecs when we need
it, and we actually end up making fewer calls, so it's faster as well.
This CL also returns all std::vectors by-value instead of by-ref. Move
semantic together with in-place filtering of codecs actually end up with
fewer copies, and it's also simpler to not return references.

BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2492473002
Cr-Commit-Position: refs/heads/master@{#15007}
2016-11-10 07:45:20 +00:00
79e05888e8 Set actual transport overhead in rtp_rtcp
BUG=webrtc:6557

Review-Url: https://codereview.webrtc.org/2437503004
Cr-Commit-Position: refs/heads/master@{#14968}
2016-11-08 10:50:16 +00:00
3663c52382 Provide move semantic for cricket::Codec and subclasses
The cricket::Codec class contains std containers like
std::map<std::string, std::string> and is expensive to copy. This CL
adds move constructors and move assignment operators for it and all
subclasses.

This CL also:
 * Implement functions with '= default' instead of doing it manually.
 * Makes codec::Matches symmetric. We currently don't check if the
   payload type of the callee is static, and might incorrectly return
   true in these cases.

BUG=None

Review-Url: https://codereview.webrtc.org/2481193002
Cr-Commit-Position: refs/heads/master@{#14956}
2016-11-07 18:14:44 +00:00
87d7d77700 Add new codec for FlexFEC.
This CL does nothing except adding new strings and enums corresponding to
the new codec.

R=stefan@webrtc.org
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2470103002
Cr-Commit-Position: refs/heads/master@{#14943}
2016-11-07 11:04:03 +00:00
7341ab8e25 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
Reason for revert:
Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio

Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.

Original issue's description:
> Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
>
> Replaced with webrtc::VideoFrame.
>
> TBR=mflodman@webrtc.org
> BUG=webrtc:5682
>
> Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> Cr-Commit-Position: refs/heads/master@{#14885}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2471783002
Cr-Commit-Position: refs/heads/master@{#14886}
2016-11-02 10:40:05 +00:00
45c8b89400 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
Replaced with webrtc::VideoFrame.

TBR=mflodman@webrtc.org
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2383093002
Cr-Commit-Position: refs/heads/master@{#14885}
2016-11-02 10:20:28 +00:00
803d97f159 Let ViEEncoder express resolution requests as Sinkwants.
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.

To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.

BUG=webrtc:5687
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
2016-11-01 18:45:54 +00:00
9c41e47b12 Remove unnecessary test fixture in codec_unittest.cc
BUG=None

Review-Url: https://codereview.webrtc.org/2462053002
Cr-Commit-Position: refs/heads/master@{#14857}
2016-10-31 16:06:07 +00:00