340 Commits

Author SHA1 Message Date
sakal
ff0e72fd16 Add QP sum stats for received streams.
This is not implemented yet in any of the decoders.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
2017-02-07 15:15:17 +00:00
deadbeef
e702b30fec Adding C++ versions of currently spec'd "RtpParameters" structs.
These structs will be used for ORTC objects (and their WebRTC
equivalents).

This CL also introduces some minor changes to the existing implemented
structs:

- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
  MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
  need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).

BUG=webrtc:7013, webrtc:7112

Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
2017-02-04 20:09:01 +00:00
nisse
7d2542623a Delete unneeded includes of base/common.h.
Bulk of the changes were done using

   git grep -l '#include "webrtc/base/common.h"' | \
     xargs sed -i '\,^#include.*webrtc/base/common\.h,d'

followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
2017-01-25 09:47:24 +00:00
nisse
dc2b3f3b9f Delete unused class CompositeMediaEngineWithFakeVoiceEngine.
BUG=None

Review-Url: https://codereview.webrtc.org/2645333002
Cr-Commit-Position: refs/heads/master@{#16230}
2017-01-24 08:54:59 +00:00
nisse
c23b0b26df Delete unused classes DesktopId and ScreencastEventCatcher.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2650703002
Cr-Commit-Position: refs/heads/master@{#16228}
2017-01-24 08:03:32 +00:00
hbos
50cfe1fda7 RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-framesdropped
Implemented as frames_received - frames_rendered.

Part of this CL is adding frames_rendered to VideoReceiveStream::Stats
and updating it at ReceiveStatisticsProxy::OnRenderedFrame.

BUG=webrtc:6757, chromium:659137, chromium:627816
NOTRY=True

Review-Url: https://codereview.webrtc.org/2607933002
Cr-Commit-Position: refs/heads/master@{#16213}
2017-01-23 15:21:55 +00:00
hbos
42f6d2fb6c RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.
VideoReceiverInfo::frames_received added based on
VideoReceiveStream::Stats::frame_counts (.key_frames + .delta_frames).

BUG=webrtc:6757, chromium:659137, chromium:627816

Review-Url: https://codereview.webrtc.org/2607913002
Cr-Commit-Position: refs/heads/master@{#16185}
2017-01-20 11:56:50 +00:00
perkj
0b2d3e217f Revert of Fix flaky WebRtcVideoChannel2BaseTest.GetStats T (patchset #1 id:1 of https://codereview.webrtc.org/2634273002/ )
Reason for revert:
nisse landed  a change that always disable adaptation in these tests.

Original issue's description:
> Fix flaky WebRtcVideoChannel2BaseTest.GetStats T
> This cl allows width and height of the produced encoded stream to be smaller than the configured camera resolution. This is since quality and cpu adapters may request a scaled input frame.
>
> BUG=webrtc:6990
>
> Review-Url: https://codereview.webrtc.org/2634273002
> Cr-Commit-Position: refs/heads/master@{#16118}
> Committed: 311a64ccf5

TBR=sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6990

Review-Url: https://codereview.webrtc.org/2639573002
Cr-Commit-Position: refs/heads/master@{#16120}
2017-01-17 13:56:53 +00:00
nisse
2013e29df1 Disable automatic scaling in tests.
BUG=webrtc:6990

Review-Url: https://codereview.webrtc.org/2636903004
Cr-Commit-Position: refs/heads/master@{#16119}
2017-01-17 13:45:40 +00:00
perkj
311a64ccf5 Fix flaky WebRtcVideoChannel2BaseTest.GetStats T
This cl allows width and height of the produced encoded stream to be smaller than the configured camera resolution. This is since quality and cpu adapters may request a scaled input frame.

BUG=webrtc:6990

Review-Url: https://codereview.webrtc.org/2634273002
Cr-Commit-Position: refs/heads/master@{#16118}
2017-01-17 12:37:02 +00:00
ivoc
4e477a1d7b Added a new echo likelihood stat that reports the maximum value from a previous time period.
BUG=webrtc:6797

Review-Url: https://codereview.webrtc.org/2629563003
Cr-Commit-Position: refs/heads/master@{#16079}
2017-01-15 16:29:46 +00:00
perkj
d533aec3b8 Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source removal.
BUG=webrtc:6371,webrtc:6983

Review-Url: https://codereview.webrtc.org/2469993003
Cr-Commit-Position: refs/heads/master@{#16048}
2017-01-13 13:57:25 +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
nisse
ede5da4960 Replace ASSERT by RTC_DCHECK in all non-test code.
Bulk of the changes were produced using

  git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
    xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.

Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
2017-01-12 13:15:36 +00:00
deadbeef
293e926362 Reland of: 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-Original-Commit-Position: refs/heads/master@{#15777}
Committed: 7a5fa6cd61
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#16016}
2017-01-11 20:28:30 +00:00
nisse
af916899cc Move VideoFrame and related declarations to webrtc/api/video.
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.

BUG=webrtc:5880

Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
2017-01-10 15:44:26 +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
kthelgason
62ffe9a339 Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
Reason for revert:
Relanding as downstream has been fixed.

Original issue's description:
> 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.
> BUG=webrtc:6906
>
> Review-Url: https://codereview.webrtc.org/2584563004 .
> Cr-Commit-Position: refs/heads/master@{#15660}
> Committed: ffb865f3e0

TBR=perkj@webrtc.org,skvlad@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6906

Review-Url: https://codereview.webrtc.org/2614693002
Cr-Commit-Position: refs/heads/master@{#15956}
2017-01-09 09:22:16 +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
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
pbos
c7c26a0e64 Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
Reason for revert:
Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used.

Original issue's description:
> Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
>
> Reason for revert:
> Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
>
> Original issue's description:
> > Replace basictypes.h with stdint.h for int_t types.
> >
> > Removes basictypes.h for types that only makes use of it for fixed-size-int
> > typedefs and replaces it with stdint.h.
> >
> > BUG=webrtc:6853
> > R=tommi@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2604043002
> > Cr-Commit-Position: refs/heads/master@{#15867}
> > Committed: 7fd1a75300
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6853
>
> Review-Url: https://codereview.webrtc.org/2603203003
> Cr-Commit-Position: refs/heads/master@{#15869}
> Committed: 7eb0e23bcf

BUG=webrtc:6853
TBR=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2609783002
Cr-Commit-Position: refs/heads/master@{#15873}
2017-01-02 16:42:32 +00:00
pbos
7eb0e23bcf Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
Reason for revert:
Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.

Original issue's description:
> Replace basictypes.h with stdint.h for int_t types.
>
> Removes basictypes.h for types that only makes use of it for fixed-size-int
> typedefs and replaces it with stdint.h.
>
> BUG=webrtc:6853
> R=tommi@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2604043002
> Cr-Commit-Position: refs/heads/master@{#15867}
> Committed: 7fd1a75300

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6853

Review-Url: https://codereview.webrtc.org/2603203003
Cr-Commit-Position: refs/heads/master@{#15869}
2017-01-02 15:32:25 +00:00
pbos
7fd1a75300 Replace basictypes.h with stdint.h for int_t types.
Removes basictypes.h for types that only makes use of it for fixed-size-int
typedefs and replaces it with stdint.h.

BUG=webrtc:6853
R=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2604043002
Cr-Commit-Position: refs/heads/master@{#15867}
2017-01-02 14:58:46 +00:00
deadbeef
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
deadbeef
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
skvlad
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
kthelgason
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
nisse
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
nisse
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
gyzhou
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
nisse
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
gyzhou
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
gyzhou
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
kthelgason
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
kthelgason
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
kthelgason
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
ossu
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
zhihuang
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
magjed
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
sergeyu
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
brandtr
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
magjed
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
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
Sergey Ulanov
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
magjed
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
brandtr
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