20 Commits

Author SHA1 Message Date
Stefan Holmer
789ba92e14 Simplify CongestionController.
- Removes the dependency on CallStats.
- Implements Module interface so that we don't have to register
  each internal component to the process thread separately.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1704983002 .

Cr-Commit-Position: refs/heads/master@{#11655}
2016-02-17 14:52:25 +00:00
Stefan Holmer
58c664c13d Clean up of CongestionController.
Removes unused methods and moves out ViERemb to Call.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1663413003 .

Cr-Commit-Position: refs/heads/master@{#11527}
2016-02-08 13:31:53 +00:00
stefan
ba4c0e45ff Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.

TBR=mflodman@webrtc.org
BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1604563002

Cr-Commit-Position: refs/heads/master@{#11487}
2016-02-04 12:12:31 +00:00
stefan
bba9dec4d5 Use separate rtp module lists for send and receive in PacketRouter.
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.

Also moves sending transport feedback to the pacer thread.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1628683002

Cr-Commit-Position: refs/heads/master@{#11443}
2016-02-01 12:40:04 +00:00
stefan
3313ec901f Enable transport seq num extension on receive channel to suppress log warning.
TBR=pbos@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1608563005

Cr-Commit-Position: refs/heads/master@{#11338}
2016-01-21 14:32:48 +00:00
Stefan Holmer
3842c5c7f7 Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12 12:55:11 +00:00
pkasting
25702cb162 Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 21:50:32 +00:00
Fredrik Solenberg
ea07373a2e Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
BUG=webrtc:5268,webrtc:5273
TESTED=Fixed issues reported by:
find webrtc/audio -type f -name *.cc -o -name *.h | xargs cpplint.py
find webrtc/call -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1483323002 .

Cr-Commit-Position: refs/heads/master@{#10853}
2015-12-01 10:26:46 +00:00
solenberg
358057b945 Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1482703002

Cr-Commit-Position: refs/heads/master@{#10828}
2015-11-27 18:46:47 +00:00
solenberg
13725089ef Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1459083007

Cr-Commit-Position: refs/heads/master@{#10788}
2015-11-25 16:16:57 +00:00
solenberg
7add058439 Move some receive stream configuration into webrtc::AudioReceiveStream.
Simplify creation of VoE channels and Call streams in WVoMC.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1454073002

Cr-Commit-Position: refs/heads/master@{#10731}
2015-11-20 17:59:40 +00:00
solenberg
8b85de2ba1 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1442483003

Cr-Commit-Position: refs/heads/master@{#10654}
2015-11-16 17:48:12 +00:00
solenberg
3a94154035 Move some send stream configuration into webrtc::AudioSendStream.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1418503010

Cr-Commit-Position: refs/heads/master@{#10652}
2015-11-16 15:34:59 +00:00
solenberg
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 23:34:58 +00:00
Fredrik Solenberg
0ccae13556 Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1402403008 .

Cr-Commit-Position: refs/heads/master@{#10491}
2015-11-03 09:15:59 +00:00
solenberg
85a0496b8c Implement AudioSendStream::GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1414743004

Cr-Commit-Position: refs/heads/master@{#10424}
2015-10-27 10:35:30 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
2015-10-22 08:49:39 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
2015-10-20 13:01:55 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00