| 3842c5c7f73639527e653f41c65334245d2317a1 |
|
12-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 25702cb1628941427fa55e528f53483f239ae011 |
|
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 7623ce4aeb9130c937ba5836490cbb3a35679e79 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.
Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}
BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1510183002 .
Cr-Commit-Position: refs/heads/master@{#10948}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| d3c944755ec546f46d5bdd84bff359fe6c4639e9 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Nuke TickTime::UseFakeClock.
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.
BUG=webrtc:5318
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512853002 .
Cr-Commit-Position: refs/heads/master@{#10947}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 8237abf563bf4782ee104408b53cc8e55ce44518 |
|
08-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}
TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695
Review URL: https://codereview.webrtc.org/1507903005
Cr-Commit-Position: refs/heads/master@{#10937}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 03ef053202bc5d5ab43460eebf5403232f157646 |
|
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1506773002 .
Cr-Commit-Position: refs/heads/master@{#10926}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| b572768efbc1e52b97a5ad98932c667956aba4b8 |
|
04-Dec-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Remove calls to VoEDtmf from WVoE/MC.
- Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent().
- Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
BUG=webrtc:4690
R=pthatcher@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1491743004 .
Cr-Commit-Position: refs/heads/master@{#10895}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| ea07373a2eb46f2732a8b5acef06a9b5078f37f8 |
|
01-Dec-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 358057b945725390bcecc330513160aa823f651e |
|
27-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 13725089ef91f932b37b2447c3f05d9cd9f89984 |
|
25-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 8b85de2ba1a8885b70bf9fe8beadc54c5c405335 |
|
16-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 3a94154035fa16e4efd91125311f076b547c38b9 |
|
16-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 566ef247b9779f6c9d0e7ec9eea6b037f4682c53 |
|
07-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 0ccae135562ac180da053fcecda91a0365621f14 |
|
03-Nov-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| 85a0496b8c4ac01da7c716ea7950093659864c8e |
|
27-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Implement AudioSendStream::GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1414743004
Cr-Commit-Position: refs/heads/master@{#10424}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|
| c7a8b08a7cd8d8f37d7f5fb9930d0cdc74baba35 |
|
16-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
/external/webrtc/webrtc/audio/audio_send_stream_unittest.cc
|