| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 8093d5442e4c365bfebc07abcf5fb653bd7a1d57 |
|
12-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Change default SSRC for RTCP receiver reports to not collide with video.
BUG=chromium:547661
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1438183002
Cr-Commit-Position: refs/heads/master@{#10621}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 797ef123249f793655640e8cb6ff1eb4fe7e3223 |
|
22-Oct-2015 |
ivoc <ivoc@webrtc.org> |
Added StopAecDump function to PeerConnectionFactory.
The function to stop recording an AEC dump was missing from the PeerConnectionFactory interface (only a start function was provided). This CL adds the missing stop function.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1415733005
Cr-Commit-Position: refs/heads/master@{#10372}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 1d8a506405734d0cef9653704b036ca4f1388960 |
|
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
|
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 66f43392a31ac566565e910246ef496fcbbafb04 |
|
09-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1324853003
Cr-Commit-Position: refs/heads/master@{#9904}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 76b62ff1ad4819ab11133d30abafd705e78a387f |
|
20-May-2015 |
Tommi <tommi@webrtc.org> |
Clean up now-unused code that was used for libpeerconnection.[so|dll].
BUG=chromium:463660
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56409004
Cr-Commit-Position: refs/heads/master@{#9240}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 9478437fdea4eb31b92ffe0c10368fe5bc9b9e16 |
|
20-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
int8_t*, and char*. Previously, they accepted void*, meaning that
any kind of pointer was accepted. I think requiring an explicit
cast in cases where the input array isn't already of a byte-sized
type is a better compromise between convenience and safety.
2. data() can now return a uint8_t* instead of a char*, which seems
more appropriate for a byte array, and is harder to mix up with
zero-terminated C strings. data<int8_t>() is also available so
that callers that want that type instead won't have to cast, as
is data<char>() (which remains the default until all existing
callers have been fixed).
3. Constructors, SetData(), and AppendData() now accept arrays
natively, not just decayed to pointers. The advantage of this is
that callers don't have to pass the size separately.
4. There are new constructors that allow setting size and capacity
without initializing the array. Previously, this had to be done
separately after construction.
5. Instead of TransferTo(), Buffer now supports swap(), and move
construction and assignment, and has a Pass() method that works
just like std::move(). (The Pass method is modeled after
scoped_ptr::Pass().)
R=jmarusic@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42989004
Cr-Commit-Position: refs/heads/master@{#9033}
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 8296ec518b2659de922668bfe0db57e71eb17e74 |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44779004
Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 3cefbc99f4cc2db744cb130ca629768401a59eb4 |
|
10-Oct-2014 |
xians@webrtc.org <xians@webrtc.org> |
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 97abeee2825ac93b62397feea74d0ad02d42540d |
|
09-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77263371-> 77296420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 1f8a23757af8ec10ba57fc14be221a5d53e8f2f1 |
|
28-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74235596-> 74297316
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| b4c7b09c1352174ecc1faf8c0cd93c66028a0485 |
|
25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 6b21b710686b017badb7853acf5d20ca92e162cd |
|
31-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72205295-> 72320533
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 0d15159b041f34855a291322d6a785211244e02d |
|
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69634309-> 69640360
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6512 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 8563ef448a9dcf7cd5755da488b29e7a7f9cc5de |
|
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69587333-> 69588608
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvoiceengine.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
|
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.h
|
| 9dba52562725dbaced0d671982201ede753d72e8 |
|
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.h
|
| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
|
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.h
|
| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
|
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.h
|