| 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/fakewebrtcvoiceengine.h
|
| b2328d11dcc86fba1661ee3fa0d51fc126939764 |
|
12-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 66085beef83c790a69666b9be8a74bb2eee44fab |
|
16-Dec-2015 |
peah <peah@webrtc.org> |
Bugfix that fixes the error where the audio processing module is called
using the wrong sample rate for the render signal.
The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.
The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).
It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1525173002
Cr-Commit-Position: refs/heads/master@{#11045}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 0ccae135562ac180da053fcecda91a0365621f14 |
|
03-Nov-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1402403008 .
Cr-Commit-Position: refs/heads/master@{#10491}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 5aaa9b4fe454195df1def4ebd36301a706fdd8d8 |
|
02-Oct-2015 |
peah <peah@webrtc.org> |
Removed unused API functions in AudioProcessing and AudioProcessingModule
BUG=
Review URL: https://codereview.webrtc.org/1379123002
Cr-Commit-Position: refs/heads/master@{#10138}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| b04965ccf83c2bc6e2758abab9bea0c18551a54c |
|
09-Sep-2015 |
ivoc <ivoc@webrtc.org> |
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 60d9b332a5391045439bfb6a3a5447973e3d5603 |
|
14-Aug-2015 |
ekmeyerson <ekmeyerson@webrtc.org> |
Integrate Intelligibility with APM
- Integrates intelligibility into audio_processing.
- Allows modification of reverse stream if intelligibility enabled.
- Makes intelligibility available in audioproc_float test.
- Adds reverse stream processing to audioproc_float.
- (removed) Makes intelligibility toggleable in real time in voe_cmd_test.
- Cleans up intelligibility construction, parameters, constants and dead code.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1234463003
Cr-Commit-Position: refs/heads/master@{#9713}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 86c6d33aec684d08189d498912e47cbc17c4d2db |
|
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
An earlier version of this commit caused a crash on AEC dump.
TBR=aluebs@webrtc.org,pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1248393003 .
Cr-Commit-Position: refs/heads/master@{#9626}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 64e753c3998a17429418180b3a947231a9fd98cd |
|
23-Jul-2015 |
magjed <magjed@webrtc.org> |
Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/c204754b7a0cc801c70e8ce6c689f57f6ce00b3b
TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| c204754b7a0cc801c70e8ce6c689f57f6ce00b3b |
|
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1226093007 .
Cr-Commit-Position: refs/heads/master@{#9619}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 4e7aa43ea0fd7106cd39036798877301398966a6 |
|
07-Jul-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
audio_processing: Adds two UMA histograms logging delay jumps in AEC
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.
This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.
Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.
BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229443003.
Cr-Commit-Position: refs/heads/master@{#9544}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| f09e09c7eef8722cc6902069a1ab7deb8948f98b |
|
26-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: Remove unused interfaces
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| c3f4dbc40b9369a7f8eb9248adb8a018b9d8e439 |
|
20-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove rtp_rtcp/ dump functionality.
Removes RTP dumping from VideoEngine and VoiceEngine.
BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47179004
Cr-Commit-Position: refs/heads/master@{#9234}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| adf89b7e33cc54dab9365dddead687a46a074cf0 |
|
29-Apr-2015 |
Ivo Creusen <ivoc@webrtc.org> |
Added SetBitRate function to VoE API to allow changing the audio bitrate.
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.
BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50789004
Cr-Commit-Position: refs/heads/master@{#9115}
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 9b2e1144df6e3622354caca00baf4a7462a0809c |
|
13-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Supporting Opus DTX in Voice Engine.
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.
BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43709004
Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 8315d7de8551963c53162e320835c158088fcdd6 |
|
14-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove dual stream functionality in VoiceEngine
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.
BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| ece3890d3a40fe911ae895e28c329491e795b14d |
|
14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.
R=stefan@webrtc.org, xians@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/27179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 4cebd84c792309c98aed9ba05524ce051341268b |
|
01-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland "Remove DTMF status methods from Voice Engine" r7276
This reverts r7277.
TBR=henrika@webrtc.org,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 7aad5e5cced724c08b0af4bf85db446c5965ac76 |
|
30-Sep-2014 |
xians@webrtc.org <xians@webrtc.org> |
Revert 7338 "Fixed the android build by making the interface pur..."
> Fixed the android build by making the interface pure virtual.
>
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
>
> Review URL: https://webrtc-codereview.appspot.com/24789004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 90d1979d77ab07f9524e6e7738f135636c45bb74 |
|
30-Sep-2014 |
xians@webrtc.org <xians@webrtc.org> |
Fixed the android build by making the interface pure virtual.
TBR=asapersson@webrtc.org, bjornv@webrtc.org,
Review URL: https://webrtc-codereview.appspot.com/24789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 3987f10c1142ffa07d749ce7b055b8a68892c19d |
|
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Remove DTMF status methods from Voice Engine" r7276
This change caused some trouble.
TBR=henrika@webrtc.org,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| bf7b9e0081233661ac0fe9500c0aa5b2aea70376 |
|
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove DTMF status methods from Voice Engine
These methods are not used.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 64a2f10f4b566a91b358e77c4ecdf09ebb33ac59 |
|
22-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove Get/SetNetEQPlayoutMode APIs
These are not used anymore.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 5d639b3ef36c81a2330e5f0a4f7c119294400515 |
|
10-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75141932-> 75179475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| a8d8ad2be6b7c204bbdc8c20a942e0aefb4fa347 |
|
16-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71240799-> 71250251
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| d8524348bbb9e5b960f670d84cb689c46f49b3de |
|
14-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71107853-> 71115715
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 3ffa1f917ec1a8bd7666669ddb3f8ba0fd26cb4e |
|
02-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70422491-> 70424781
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6586 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 0bb9fac98ca95509e7c07debaee316bdaa2f4eaa |
|
02-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70343444-> 70394475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6581 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| d8a90690809f0fa57e88911fb96848e227947424 |
|
01-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70340027-> 70343444
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6579 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| bfa758a54c066f4d0bb125e102fbb654ee177a88 |
|
27-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70004190-> 70103367
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6555 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| ae740dd94cb4f11271e5dc9b27eee1f2e29a37a8 |
|
17-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69359922-> 69365993
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| d054bff3b9a23ddf1e8c0c844f13bc4b10540689 |
|
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69292418-> 69293749
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6452 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 24301a67c66e6091418e83da49cfb367ef2c6645 |
|
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 364f204d16d1f10cf01b1b5543ce020c3e9961b8 |
|
20-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56698267.
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 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/fakewebrtcvoiceengine.h
|
| 1b15f4226ff417095d2146401ca71cd98ab735b3 |
|
07-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51960985.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2188004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.h
|
| 91053e7c5a743f4a92f5079844b0747c927f3bbd |
|
10-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50654631.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2000006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|
| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
|
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/fakewebrtcvoiceengine.h
|
| 723d683ecbe6a934885a60712c66ca2c01700a51 |
|
12-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1797004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvoiceengine.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/fakewebrtcvoiceengine.h
|