| fcfc804e436502d49b2176fec1f40dce3585527f |
|
14-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Eliminate defines in talk/
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).
When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1588453005
Cr-Commit-Position: refs/heads/master@{#11254}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2d110be77f14cab0bb51efe8b61d9c7a967d04cb |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| e591f9377f33f3f725a30faecd1bef1a71fa6b99 |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Storing raw audio sink for default audio track.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11230}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a4df27b6713583045e51e20c4eb93718d15ca33e |
|
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| f4f5cb09277d5ef6aeac8341e5f54a055867803a |
|
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/
The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
NOTRY=true
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1541633002
Cr-Commit-Position: refs/heads/master@{#11093}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 36d4c545007129446e551c45c17b25377dce89a4 |
|
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ae2c5ad12afc8cc29fe9c59dea432b697b871a87 |
|
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Added option to specify a maximum file size when recording an AEC dump.
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 32d989b3f2d168327ed43d0e4c493550ccee4179 |
|
15-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Disable transport sequence numbers for audio.
Since this isn't fully wired up yet it shouldn't be part of the
SendSideBwe experiment yet.
BUG=webrtc:5263
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1523283002 .
Cr-Commit-Position: refs/heads/master@{#11029}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| f888bb58da04c5095759b5ec7ce2e1fa2cd414fd |
|
12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1505253004 .
Cr-Commit-Position: refs/heads/master@{#10995}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 246b8171a6fbb4e37a5491679bc595238f81e490 |
|
08-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Refactor handling of AudioOptions.
- Remove MediaEngineInterface::GetAudioOptions(), SetAudioOptions() and SetSoundDevices().
- Remove the WebRtcVoiceEngine infrastructure for those calls.
BUG=webrtc:4690
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1500633002
Cr-Commit-Position: refs/heads/master@{#10938}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 9d69c3f4d99240c27d997c37950b561605d403bd |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Return a copy of the supported RTP header extensions instead of a reference.
This also renames the method to better reflect what it does.
BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1486123002 .
Cr-Commit-Position: refs/heads/master@{#10910}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.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/talk/media/webrtc/webrtcvoiceengine.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/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1d63dd0eaa44d13c5ae083200937b18bce2132ae |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
- Remove the DF_PLAY/DF_SEND flags, only allow sending.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1487393002
Cr-Commit-Position: refs/heads/master@{#10872}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7e4e01a4413fa98644b94ab9d8a9dccc664f39f2 |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
Rework filtering functionality to be reused for both Audio+Video.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1481963002
Cr-Commit-Position: refs/heads/master@{#10869}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2515af28e97213b4a4b89269f6b855378d31e153 |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1493663002
Cr-Commit-Position: refs/heads/master@{#10868}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 26c8c91de2db5da06ff337aae48e1d725aa91ab7 |
|
27-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Using Rent-A-Codec for static Codec access in WVoE/MC.
Mostly moved code around in WebRtcVoiceEngine:
- Added new internal class WebRtcVoiceCodecs for static codec functions and the CodecPrefs.
- ConstructCodecs() -> WebRtcVoiceCodecs::SupportedCodecs().
- FindWebRtcCodec -> WebRtcVoiceCodecs::ToCodecInst().
- WebRtcVoiceMediaChannel::SetRecvCodecsInternal() folded into WebRtcVoiceMediaChannel::SetRecvCodecs() (slight logic change).
- Change to how SetRecPayloadType() is implemented in fakewebrtcvoiceengine.h (lines 460-470).
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1461333002
Cr-Commit-Position: refs/heads/master@{#10819}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| bd13838ccc87f94d1e951bcf780979622b020359 |
|
21-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1457653003
Cr-Commit-Position: refs/heads/master@{#10734}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7add0584390dcfb236165a6472ede6c2a94eaeed |
|
20-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7e63ef0e8f3baf832005e2e378b6834c0d005f12 |
|
20-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Allow default audio receive channel to receive on any unsignalled SSRC.
BUG=webrtc:5208
Review URL: https://codereview.webrtc.org/1455923003
Cr-Commit-Position: refs/heads/master@{#10723}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.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/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5237aaf243d29732f59557361b7a993c0a18cf0e |
|
11-Nov-2015 |
tfarina <tfarina@chromium.org> |
Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| be57983f4bd875c39a229bab5112b32dad004057 |
|
10-Nov-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Rename Maybe to Optional
And add examples of good and bad usage to the documentation.
R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1432553007 .
Cr-Commit-Position: refs/heads/master@{#10588}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.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/talk/media/webrtc/webrtcvoiceengine.cc
|
| 102c6a61bc0b42dc0956d013530fc0213b7e881b |
|
30-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Replace rtc::cricket::Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.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/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4f4ec0a9270a8cefadfa12e9fa3b979b58b15392 |
|
22-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c96df779b0c9255f25dc78c20a4cd4dff1776384 |
|
21-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel.
- Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
- Create webrtc::AudioSendStreams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1415563003
Cr-Commit-Position: refs/heads/master@{#10361}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0a617e22a46d476abcaaa081cc90300d335da9f9 |
|
21-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove the default send channel in WVoE.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364643003
Cr-Commit-Position: refs/heads/master@{#10344}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 43e83d44f01683fbd304e37d47d2f6db0d52660d |
|
20-Oct-2015 |
solenberg <solenberg@webrtc.org> |
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: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
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}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a457752f4afc496ed7f4d6b584b08d8635f18cc0 |
|
20-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 112a3d81db02d349af0ce6c0827da6d8fbc421a8 |
|
16-Oct-2015 |
ivoc <ivoc@webrtc.org> |
Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1374253002
Cr-Commit-Position: refs/heads/master@{#10297}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1ac561447e3e1d81a1e390f95a385b5ed8fe0932 |
|
13-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 3.
Get rid of default receive channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1385893002
Cr-Commit-Position: refs/heads/master@{#10262}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 8fb30c328b7b5e1ad33e970d1dabca55fdc18926 |
|
13-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 2.
Rename voe_channel_ to default_send_channel_id_.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1388733002
Cr-Commit-Position: refs/heads/master@{#10261}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| d4cec0d8fa7913bc9dfa9137e44cca9098e16698 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove MediaChannel::SetRemoteRenderer().
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1398823003
Cr-Commit-Position: refs/heads/master@{#10237}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 98c68865e715f693390209adb454ab3a5b6de332 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
Cr-Commit-Position: refs/heads/master@{#10226}
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10235}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4bac9c53da9988741d59753c2d789adb94de5e68 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397773002
Cr-Commit-Position: refs/heads/master@{#10234}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0b67546d8c080f376565a4c1cedd14947fdbaf2b |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 1.
Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1388723002
Cr-Commit-Position: refs/heads/master@{#10233}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| eefbc3bbd7a6962265b028cf259b5028944561d1 |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )
Reason for revert:
Breaks Chrome since its build files were not updated prior to file removal.
Original issue's description:
> - Remove AudioTrackRenderer.
> - Remove AddChannel/RemoveChannel from AudioRenderer interface.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
> Cr-Commit-Position: refs/heads/master@{#10226}
TBR=tommi@webrtc.org,solenberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1393343003
Cr-Commit-Position: refs/heads/master@{#10228}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1c0bb386b67835feb5934f503dddfe0912bce3ac |
|
08-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10226}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| d97ec30ce4f22ba2d88314d67ff44458144a5096 |
|
07-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1386653002
Cr-Commit-Position: refs/heads/master@{#10194}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5b14b42e93f17d0ea57f1f8b3e8224082c514946 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1362913004
Cr-Commit-Position: refs/heads/master@{#10133}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| dfc8f4ff8731390828884a0a91b99e51f2950275 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1378513003
Cr-Commit-Position: refs/heads/master@{#10130}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 63b345441a995665c1cdd0329b65f895675874ff |
|
29-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Simplify handling of options in WebRtcVoiceMediaEngine.
Also removes unnecessary typedef ChannelList.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364753002
Cr-Commit-Position: refs/heads/master@{#10107}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4a3ccad29e4f14c4a66d10edda0d364ea415e309 |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetAudioDelayOffset() and friends.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364093002
Cr-Commit-Position: refs/heads/master@{#10047}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 61e933eac7673feb2f8663c3e71e503b714b350f |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364083002
Cr-Commit-Position: refs/heads/master@{#10045}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c14f5ff60fb0c42c97702de112a9e8f1eccba574 |
|
23-Sep-2015 |
henrika <henrika@webrtc.org> |
Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.
R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE
Review URL: https://codereview.webrtc.org/1344563002 .
Cr-Commit-Position: refs/heads/master@{#10030}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| d5c75b1a0ba1548d3561109e3e5e63757509e9ae |
|
23-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Reduce LS_INFO spam from voice_engine/.
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).
BUG=
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1347353004 .
Cr-Commit-Position: refs/heads/master@{#10028}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7d173362d01229fe262df37e34ecb061aee8edc3 |
|
23-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove the [Un]RegisterVoiceProcessor() API.
BUG=webrtc:4690
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1361633002 .
Cr-Commit-Position: refs/heads/master@{#10027}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 09677342ae9dce4f4ec9c294342a8b1789dcdba2 |
|
23-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.
BUG=webrtc:4690
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1360773002 .
Cr-Commit-Position: refs/heads/master@{#10026}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c1a1b353ec96a92f8b88dba5a058af8744e81560 |
|
22-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove the SetLocalMonitor() API.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1344083004
Cr-Commit-Position: refs/heads/master@{#10020}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 22011c1b54021ec9a2b4885519e5ce995b1300a2 |
|
22-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
BUG=webrtc:4690
TBR=juberti
Review URL: https://codereview.webrtc.org/1325023005
Cr-Commit-Position: refs/heads/master@{#10011}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| b071a19019a0a2173cc139c960d6ef6946a1c581 |
|
17-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1327933002 .
Cr-Commit-Position: refs/heads/master@{#9973}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 709ed67c38d0a942f3bf3e68e337cc27a27bc353 |
|
15-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1269863005 .
Cr-Commit-Position: refs/heads/master@{#9939}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1dd98f321920c1442dd5b3f791ea0fca133c2756 |
|
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
|
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 86d907cffda803ee34ee68f9833c1980d1b9f7a6 |
|
07-Sep-2015 |
henrika <henrika@webrtc.org> |
Refactor the AudioDevice for iOS and improve the performance and stability
This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| bb741b3afa23ec59c1948841f2de71f422245564 |
|
07-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove GetOutputScaling from VoiceMediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1331443003
Cr-Commit-Position: refs/heads/master@{#9870}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 658910cc3cb54705672c28fffedba4e982fa3989 |
|
03-Sep-2015 |
stefan <stefan@webrtc.org> |
Revert "Speculative revert of "- Move test cases for more natural ordering.""
Did not resolve the build bot issue.
This reverts commit 02d283a6ff5364d94aa88f5f5df4cfd3a5411346.
BUG=
TBR=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1324123002
Cr-Commit-Position: refs/heads/master@{#9849}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 02d283a6ff5364d94aa88f5f5df4cfd3a5411346 |
|
01-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Speculative revert of "- Move test cases for more natural ordering."
This reverts commit c20a5dc9305b988ca173cd63e606124b02e6d54c.
BUG=webrtc:4959
R=solenberg@webrtc.org
TBR=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1309313008 .
Cr-Commit-Position: refs/heads/master@{#9829}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c20a5dc9305b988ca173cd63e606124b02e6d54c |
|
31-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.
BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1319163002 .
Cr-Commit-Position: refs/heads/master@{#9817}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| af9fb218864b8cb4cccd32280b68dd1b34cb2213 |
|
26-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Use C++11 loops in WebRtcVoiceMediaEngine/Channel.
- Pull out part of WebRtcVoiceMediaChannel::SetRecvCodecs() into WebRtcVoiceMediaChannel::SetRecvCodecsInternal().
BUG=webrtc:4690
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1291343002 .
Cr-Commit-Position: refs/heads/master@{#9785}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c2ee2c86f905991a8cd05ee1f35bea105b41e4e0 |
|
08-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229283003 .
Cr-Commit-Position: refs/heads/master@{#9690}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 6bb1b6e7fe5631e9f218b80292df5b64623c5616 |
|
24-Jul-2015 |
pbos <pbos@webrtc.org> |
Control combined_audio_video_bwe with config bool.
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".
BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1256803004
Cr-Commit-Position: refs/heads/master@{#9633}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 8fc7fa798f7a36955f1b933980401afad2aff592 |
|
15-Jul-2015 |
pbos <pbos@webrtc.org> |
Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.
BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1181653002
Cr-Commit-Position: refs/heads/master@{#9586}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0f133b99c655cbdb347b4a71ac872c071532189f |
|
02-Jul-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Rename APM Config ReportedDelay to DelayAgnostic
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 441f6347311bcf2079435c3888d67e1fb321f9f8 |
|
09-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec."
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 3fbf3f8841b5460503fb646eaedcb063620434a8 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.
BUG=4696
TBR=bjornv@webrtc.org
Review URL: https://codereview.webrtc.org/1166463006
Cr-Commit-Position: refs/heads/master@{#9380}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 73f72105c4b671624613cc132bfa86cfc956318b |
|
03-Jun-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Actively turns off platform-AEC when DA-AEC is used
When initiating a call default audio options are applied, which turns on platform-AEC if such exists. Then, if delay agnostic AEC (DA-AEC) is enabled through a media constraint no action with respect to platform-AEC is taken (a bug) and turning on SW AEC. Hence, we run both AECs.
This CL makes sure the platform-AEC is disabled if we want to run DA-AEC.
BUG=
TESTED=locally on Nexus 4 and Nexus 6.
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52049004
Cr-Commit-Position: refs/heads/master@{#9361}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 8e6fd46cc324f8946e68396edcb252ffaf2d4579 |
|
02-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50179004
Cr-Commit-Position: refs/heads/master@{#9350}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5263b3c1ddb10ecca58d9f08364aad2d6ba1d95d |
|
01-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.
When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.
BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55479004
Cr-Commit-Position: refs/heads/master@{#9344}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c28a896a7bbd8a1ffef44a1f66ac67c43b4eeada |
|
29-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690
Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code
R=solenberg@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56499004
Cr-Commit-Position: refs/heads/master@{#9330}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ccb49e79fd4c439a30b9a999eab4ef329ba8425c |
|
19-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51009004
Cr-Commit-Position: refs/heads/master@{#9216}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2013aeced2b7821a407f302802c4a16fd02728b1 |
|
13-May-2015 |
Minyue <minyue@webrtc.org> |
Propagating RTT from send-only channel to receive-only channel.
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.
BUG=3978
TEST=chromium with hangout calls
R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29989004
Cr-Commit-Position: refs/heads/master@{#9186}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 300eeb68f55c5091c7045e377578586733cddf16 |
|
12-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interfaces.
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.
Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).
BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55379004
Cr-Commit-Position: refs/heads/master@{#9179}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 64dad838e61e92e4a72437b153c5eba7a200fb4a |
|
11-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49329004
Cr-Commit-Position: refs/heads/master@{#9169}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1f629232d5f852452499104c28e7d61c7b0b8c77 |
|
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55369004
Cr-Commit-Position: refs/heads/master@{#9165}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| fd32f35aff8fc28ec084bddc274de284e0422a57 |
|
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.
Contains a tentative fix to the chrome build breakage caused by the
original change.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47139004
Cr-Commit-Position: refs/heads/master@{#9164}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| cdb47a4533b7b1e29e803ed6591a68bb1a4f1692 |
|
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53399004
Cr-Commit-Position: refs/heads/master@{#9161}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 208a2294cde839025318f1b3d57559cb0611a4e7 |
|
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.
R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50869004
Cr-Commit-Position: refs/heads/master@{#9159}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| d3ddc1b69e9cdfd7c6d38ab02b8d8ab891d30fd1 |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
BUG=
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49929004
Cr-Commit-Position: refs/heads/master@{#9156}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| e444a3dcd317ff81b344a89625376e2afcffb1e2 |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
WebRtcVoiceEngine: Get rid of unnecessary template base class.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46219004
Cr-Commit-Position: refs/heads/master@{#9155}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| aaf8ff2e45ece09028b8064eec6234260d9cc081 |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
WebRtcVoiceEngine: virtual to override + git cl format.
BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54369004
Cr-Commit-Position: refs/heads/master@{#9154}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 6179b89e53eda4db57baf2efb8d85779defb410c |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove unused API on WebRtcVoiceEngine.
BUG=1695
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46209004
Cr-Commit-Position: refs/heads/master@{#9153}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4b60c73e74d62beff484b7f54d8f3267cb66274f |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ccfc93913ce015309429ea07ddf24808f111efb9 |
|
07-May-2015 |
Bjorn Volcker <bjornv@chromium.org> |
Reinterpret AudioOption delay_agnostic_aec to override HW-AEC
This CL will change the behavior when enabling Delay Agnostic AEC through the media constraint (and AudioOption delay_agnostic_aec)
FROM
Use DA-AEC instead of AECM if there is no HW-AEC
TO
Use DA-AEC even if there is a HW-AEC
Before this change the user will not really know if the Delay Agnostic AEC is running or not, so it is more intuitive if the option overrides the built-in one if the user has asked for it.
BUG=4472
TESTED=locally with a modified AppRTCDemo app
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49859004
Cr-Commit-Position: refs/heads/master@{#9147}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7c027b64ae53a29bc528b4241cc540694c239304 |
|
22-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46999004
Cr-Commit-Position: refs/heads/master@{#9053}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1d83f1e89f3e54b38d49ff877c763d0ac52fdb8b |
|
07-Apr-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC
In https://webrtc-codereview.appspot.com/48699004/ I made the audio option delay_agnostic_aec override HW-AEC if such exists. That is not an expected behavior and is fixed in this CL.
In addition we now check if EnableBuiltInAEC() was successful before disabling the SW-AEC. This revealed a bug in that return value, also fixed here.
BUG=4472
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47969004
Cr-Commit-Position: refs/heads/master@{#8936}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ef88309a6e2b3193cf1658bf245de295900ba4fe |
|
06-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Forward declare AudioFrame type in voiceprocess.h
No need to include this header since the API is just taking a pointer to
it.
BUG=1092
TEST=./webrtc/build/gyp_webrtc && ninja -C out/Debug
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44059004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8928}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7100dcd3176f6522ee96be797f73a1f50da0f5d1 |
|
27-Mar-2015 |
Minyue Li <minyue@webrtc.org> |
Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
Specifically,
usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.
BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48499004
Cr-Commit-Position: refs/heads/master@{#8872}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5225dd818047a06fe2f2a246db0fd18bb4deef5b |
|
26-Mar-2015 |
Brave Yao <braveyao@webrtc.org> |
If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
BUG=4289
TEST=Manual/Auto Test
R=juberti@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44629004
Cr-Commit-Position: refs/heads/master@{#8863}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| bf395c1fc0a29b54fac4b6f6e9f6c117762faa15 |
|
25-Mar-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.
This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.
The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/
BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699004
Cr-Commit-Position: refs/heads/master@{#8861}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| eebcab5ce99d3e8641dd92a569916b0d24e29fca |
|
24-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 3f11823a1a802d6073c416d32c347e7fb6b236f7 |
|
16-Mar-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Disables SW AEC when built-in AEC is enabled
As of r7849 the built-in AEC on devicing supporting it is enabled by default.
Unfortunately, the SW AEC (AECM) was not disabled, hence running on top of the built-in one. This is not necessary. In fact it reduce double talk performance significantly.
BUG=4431
TESTED=manually
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49419004
Cr-Commit-Position: refs/heads/master@{#8735}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8735 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
|
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| c0bd7be0df67735d63f5cdd302a3b85f88239874 |
|
18-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding two new stats to VoiceReceiverInfo
There have been requests of two new stats namely
speech_expand_rate and secondary_decoded_rate.
BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40789004
Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| f9b5c1b3d009887df02505d12ece2f80b2a90d44 |
|
17-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Removing CELT.
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.
BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36099004
Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
|
12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 62f6e756730325ee7b20cf5f81e82b0a70283a05 |
|
11-Feb-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Refactoring WebRTC Java/JNI audio recording in C++ and Java.
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33969004
Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 8cf9bdb3fad92fd783b32152e912859d8b399c97 |
|
09-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove USE_WEBRTC_DEV_BRANCH.
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.
R=bjornv@webrtc.org, juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/39849004
Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| cc64a9cc4fcc7df95cee0fc069b8924c3fb196ce |
|
05-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
As of r8230 (https://webrtc-codereview.appspot.com/39739004/) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.
This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine
BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41749004
Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a954c07ee1c93175e6ebbeb20517b347474362ae |
|
09-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4ef22d1d293fe7b2398e4cd90a0eb2e8fb02b6ea |
|
17-Nov-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Setting Opus FEC as default
BUG=3986
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7710 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 8038d42749e9edd52487baea050acda6f604bf91 |
|
11-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Follow-up fixes for G722
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.
BUG=3951
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| f85dbce041a9c49252b5c27364ce70300b652d78 |
|
07-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.
BUG=3951
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| dced5d7835ec8ada6242c2086af7899f068e96ed |
|
06-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Advertise G722 as 8 kHz rather than 16 kHz"
This reverts r7645.
TBR=pthatcher@webrtc.org
BUG=3951
Review URL: https://webrtc-codereview.appspot.com/24199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 1dcca4028fe06735819ec1ba89e5814d53767a4b |
|
06-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Advertise G722 as 8 kHz rather than 16 kHz
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.
R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.
Review URL: https://webrtc-codereview.appspot.com/27879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2dc6f3154dd233b221c53272a7f64aa20ef2e95e |
|
31-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Adapting bitrate according to maxplaybackrate for Opus.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 8219529b98238244ed4b57acaff4e0b9bf9ddca4 |
|
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Cleaning up r7562-7567.
Wrongly used git svn dcommit for committing a CL.
Then two reverts were applied.
Still something needs to be cleaned.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 879fac81d15cca19f1c9edf48833ac27637fe536 |
|
30-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78822708-> 78823675
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 5f73a375973a8917f6d417aa7d2d2fe80856b6b0 |
|
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Revert 7563 "before rebase" due to wrong submission
> before rebase
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| c673bb9f29fb0c80c112b91942682475560f821d |
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30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
before rebase
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 0b626725761cd89d4422f4538939613cbe5d1f27 |
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30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
adding default rates
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 2623695dfb48ebd745d0d578f5720e8d5160f4f3 |
|
29-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Renaming bandwidth to bitrate in webrtcvoiceengine.
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.
This is to remove the confusion inside webrtcvoiceengine
BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 9d446f2e167d0697364a118a3217ddaa47a3ce4d |
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23-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
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/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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09-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
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/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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(Auto)update libjingle 73399579-> 73626167
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/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| d1591409658e3b35f734dd1b0026661d01c796b5 |
|
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69260070-> 69276003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6439 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 7e71b77f8aab5b7a6f2b669c16f90ec9a4b4609c |
|
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69102234-> 69116997
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 18dfa8d5741443bc0a8a3e99b821516aa28ced01 |
|
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69069003-> 69082899
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| b90619c07fb9b9723ad5160651ab416724d3fa61 |
|
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69049090-> 69054765
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| b525a9d790b3fd5ec63aed92395623c3acdfd5b6 |
|
03-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68379861-> 68445177
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 150835ea34e1ee42d7af993fdcb82d98ff110d78 |
|
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66236292-> 66294299
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 13d6776c46642e708b9a7e8e72c7457b8316d5e2 |
|
02-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66098243-> 66100938
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0d34f1446a93f964cf6e221ca0ebd63935950b14 |
|
02-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66033941-> 66098243
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| af6640fce73fe0945b749ae8db3ddf6fc3d599a5 |
|
28-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65729829-> 65752960
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| f875f15afb5013e45b1af295b15ef4853c46a53b |
|
14-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64709629-> 64813990
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 15192f909e5a7e43287d2ec6cbb567c59afba7ce |
|
10-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64594651-> 64630087
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 05e7b44b83f9f12a827646c496f5d6ae796b4b99 |
|
01-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63948945-> 64147530
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5825 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 79047f99c1d39c6d3c16bd9bf0db3fb2eb1741bc |
|
07-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62691533-> 62713454
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 704bf9ebec9c9425e1898f6c3f15eff685175b23 |
|
27-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62063505-> 62278774
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a7b981843f35bb6c26cf3bc95b5a00a0b9f50a93 |
|
21-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702).
BUG=N/A
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| ef2215110c00ee1d8225b08815bfdcee918767f9 |
|
21-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5590 "description"
> description
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2643805a2057b92e916bcf4f71668bc80766625e |
|
20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
description
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| b8c254abd6fa784294277e2baa8298c3352faf78 |
|
15-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 61549749-> 61608469
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 9cf037b83184374230c6825e4aa407cdafaba434 |
|
07-Feb-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61168196
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 67ee6b9a6260fa80b83326c4b4fec8857c0e578c |
|
03-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60923971
Review URL: https://webrtc-codereview.appspot.com/7909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a8910d2f882730cbd0487946ce5aeda28759751c |
|
23-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60094938.
Review URL: https://webrtc-codereview.appspot.com/7489005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
|
16-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 59676287
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| aebb1ade9d760841f243e380fa22b7ecff2d3ecc |
|
14-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pRevert 5371 "Revert 5367 "Update talk to 59410372.""
> Revert 5367 "Update talk to 59410372."
>
> > Update talk to 59410372.
> >
> > R=jiayl@webrtc.org, wu@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/6929004
>
> TBR=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6999004
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 44461fa5cbecd556691b0ba963f95973f6abece1 |
|
13-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5367 "Update talk to 59410372."
> Update talk to 59410372.
>
> R=jiayl@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6929004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 0f3356e20b70416f13e12ef596da66f6c347eea7 |
|
11-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 59410372.
R=jiayl@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a9890800e078105f21f0a21358ee59a0b3736af6 |
|
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58127566 together with
https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 2018269dc3a1c1bb01c946583ca0750ae0db68e3 |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5274 "Update talk to 58113193 together with https://webrt..."
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
>
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a129b6cd132788a931b47da3370ae473673f320d |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 9caf2765b285f7511d8355177c2d55209d7573e4 |
|
11-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58037405.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/5579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 5bc25c41fc7880545052770dbcfe67f233c9b0c0 |
|
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 57692857
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a23f0ca4ba5105eb76b6fa30447c806812a8f3c2 |
|
13-Nov-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56619788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| de305014c62832a382d38144a9dc518cf1d02f88 |
|
31-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55906045.
Review URL: https://webrtc-codereview.appspot.com/3159005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| cecfd1832dc375225da3f5f18ecac63006ed06bf |
|
30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55821645.
TEST=try bots
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 97077a3ab27259164eb121034b6e0ebe9ba592df |
|
25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 1d1ffc9ad267d7e6e9ec9001052fd4abf29d7622 |
|
16-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54898858.
TEST=try bots
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/2414004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 4551b793dea4b5451cbfa13b206b6d11a25081d0 |
|
09-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53920541.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2371004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 78187525665490922748d79377bcb351579e03c0 |
|
08-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53856368.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2366004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4941 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a27be8e4a1f59a51ecafba71ba30ddd0bcc9f1f1 |
|
28-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| 967bfff54d00f176a554bf9f955f14dde99f7bb9 |
|
19-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 52534915.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2251004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|
| a59696b2a5f0c138d4176249bac223ad6c4316d5 |
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14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 52300956
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2213004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
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30-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51664136.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2148004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 9dba52562725dbaced0d671982201ede753d72e8 |
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05-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Update libjingle to 50389769.
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004
RISK=P1
TESTED=try bots
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1967004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
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26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
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22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1848004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 9de257d00f1f805af28f15fd814a8a84460028e5 |
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17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1824004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
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| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.cc
|