| 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/webrtc/voice_engine/voe_base_impl.cc
|
| 2515af28e97213b4a4b89269f6b855378d31e153 |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1493663002
Cr-Commit-Position: refs/heads/master@{#10868}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| ad856229a796a8efa1126ef8aa8d238f2b0a2b21 |
|
27-Nov-2015 |
pbos <pbos@webrtc.org> |
Use webrtc/base/logging.h for voice_engine.
BUG=webrtc:5118
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1474363002
Cr-Commit-Position: refs/heads/master@{#10827}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 3e6db2321ccdc8738c9cecbe9bdab13d4f0f658d |
|
26-Nov-2015 |
kjellander <kjellander@webrtc.org> |
audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 13725089ef91f932b37b2447c3f05d9cd9f89984 |
|
25-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 74640895fafbb90a6630a6a91b80da0a7cff229c |
|
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417173004 .
Cr-Commit-Position: refs/heads/master@{#10444}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 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/webrtc/voice_engine/voe_base_impl.cc
|
| e313e0278315c918d1ae810f79e4b3f176d58659 |
|
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove unnecessary fields from VoE SharedData.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1304933008
Cr-Commit-Position: refs/heads/master@{#9882}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 664cdafb8ad7ccef531cb6bf7bd42752841f220f |
|
20-May-2015 |
André Susano Pinto <andresp@google.com> |
Replace assert() with static_assert() if the condition is evaluatable at
compile time.
The condition of static_assert() is evaluated at compile time which is safer and
more efficient.
Note that static_assert() requires C++11.
The changes were generated by the misc-static-assert ClangTidy check by alexfh@google.com
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51019004
Cr-Commit-Position: refs/heads/master@{#9231}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 931e6583b21d2d3d1ee8fd240f63708dc56d1a19 |
|
20-May-2015 |
Tommi <tommi@webrtc.org> |
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0.
In particular and practical terms, this avoids pulling in AudioDeviceModuleImpl and associated classes, in Chrome.
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49999004
Cr-Commit-Position: refs/heads/master@{#9229}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 2013aeced2b7821a407f302802c4a16fd02728b1 |
|
13-May-2015 |
Minyue <minyue@webrtc.org> |
Propagating RTT from send-only channel to receive-only channel.
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.
BUG=3978
TEST=chromium with hangout calls
R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29989004
Cr-Commit-Position: refs/heads/master@{#9186}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| f353dd59b59aea3c3a1c64aa20a66c2a8c32225e |
|
06-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: cleanup VoENetwork implementation
Changes:
1. Documented return values of VoENetwork methods.
2. In VoENetworkImpl: replaced calls to SetLastError() with LOG_F(). SetLastError() is not used anymore because no one is calling LastError() to check for last error. Also, its usage is being removed in Video Engine and we want to be consistent.
3. In VoENetworkImpl: removed WEBRTC_TRACE() usage.
4. In VoENetworkImpl: replaced some defensive code with assert(). We are now assuming that the caller has called VoEBase::Init() where needed. We are also assuming that it is invalid to pass nullptr where data is expected.
5. Updated unit tests accordingly.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53369004
Cr-Commit-Position: refs/heads/master@{#9145}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 2dd6a270c0eb9f540427537b03330d0ed6824f9d |
|
14-Apr-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: format VoEBase according to new style guide
Purely cosmetic changes:
1. virtual => override
2. NULL => nullptr
3. data member name: underscore prefix => suffix
4. clang format
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49669004
Cr-Commit-Position: refs/heads/master@{#8997}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 6fc2d2f487b3e6223dff518e04fb301ba6d2cf43 |
|
13-Apr-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: revert CHECKs into asserts
Including check.h causes build failure in Chrome due to LOG macros redefinition.
Review URL: https://webrtc-codereview.appspot.com/51629004
Cr-Commit-Position: refs/heads/master@{#8984}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 9e5e421b7d1db698c8b68291355445ee3f7fe9b9 |
|
13-Apr-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: cleanup VoEBaseImpl
Changes:
1. Removed _voiceEngineObserver boolean flag, because its value is equal to (_voiceEngineObserverPtr != NULL).
2. Removed WEBRTC_TRACE macro usage wherever it was unnecessary to log. Replaced its usage with LOG_F (new and preferred way to log messages) wherever it is useful to log.
3. Replaced asserts with CHECKs.
Discussion:
To make it easier to review the changes, I didn't reformat the code to make it compliant to the new coding standards. It is up for debate how much reformatting to do: the whole file/class or just the methods that I have touched. My vote - go for the whole class.
R=henrika@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51579004
Cr-Commit-Position: refs/heads/master@{#8983}
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 3985f0151aff9b91418733795a98140079c19a73 |
|
27-Feb-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
ProcessThread improvements.
* Added a way to notify a Module that it's been attached to a ProcessThread.
The benefit of this is to give the module a way to wake up the thread
when it needs work to happen on the worker thread, immediately.
Today, module instances are typically registered with a process thread
outside the control of the modules themselves. I.e. they typically
don't know about the process thread they're attached to.
* Improve ProcessThread's WakeUp algorithm to not call TimeUntilNextProcess
when a WakeUp call is requested. This is an optimization for the above
case which avoids the module having to acquire a lock or do an interlocked
operation before calling WakeUp(), which would ensure the module's
TimeUntilNextProcess() implementation would return 0.
BUG=2822
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39239004
Cr-Commit-Position: refs/heads/master@{#8527}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8527 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 46323b378660849e0fe210e78b6f47ec552d5c5a |
|
13-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Remove useless AudioProcessing::Create() overload.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8046 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 6a364fe11b9af8fe55a64f18efeb8ef7e415dc6b |
|
05-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove uses of build date/time.
Uses of __DATE__ and __TIME__ are blocking deterministic Chromium
builds. We're not really making use of these, and if anything they're
likely to be misleading as it's impossible to distinguish between a new
revision and a freshly-built old branch.
R=mflodman@webrtc.org, tnakamura@webrtc.org
BUG=3983
Review URL: https://webrtc-codereview.appspot.com/27039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 94454b71adc37e15fd3f5a5fc432063f05caabcb |
|
05-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| cb711f77d2ff9ebd42678869a73353809b3af66e |
|
19-May-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add interface to propagate audio capture timestamp to the renderer.
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 1cec3957b88cbab345535137329bd8f3f2a6b39e |
|
12-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| ddbb8a2c243f9d54cb0ce0092e341dfc6e126bb3 |
|
22-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 66803489f9694cb7c7c0dd3ba07b63e2b6b71779 |
|
17-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 5692531f18cae04d8a8107793dc74ae932bdf219 |
|
14-Apr-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added a new OnMoreData() interface which will not feed the playout data to APM.
BUG=3147
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 40ee3d07eda24b8e8214429d9885d9ad9a2c04f7 |
|
03-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Consolidate audio conversion from Channel and TransmitMixer.
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| c7c432aa9b8c9f9ba6d41554917784a27b21426a |
|
02-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
This was only used for logging, except on Mac, where the methods are
now private.
BUG=3132
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 27c6980239a0c6bc81121a1aa75c27f9187aacf4 |
|
18-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move the volume quantization workaround from VoE to AGC.
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.
Add a test to verify the behavior.
TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| c1e28038bac58f096bdb06bc36fddd9130c82f27 |
|
02-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 07e51964143cfc5a00192e9ab71d240d0575718d |
|
29-Jan-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc.
The callback has to go through VoEBaseImpl since VoEChannel is internal to voice engine.
TEST=compile
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5458 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 023cc5abc7d25fb3133b4d0206b67dcc6204b6e8 |
|
11-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Minor voice engine improvements around AGC.
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.
R=aluebs@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 3054ba6bb22229074c77bffd15918a3bf3083130 |
|
04-Dec-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the long disabled WEBRTC_SVNREVISION define.
BUG=500
TESTED=git try
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5215 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 03f33709f8be1da10dde6a2c9b2da5fbc3d35099 |
|
13-Nov-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Inject config when creating channels to override the existing one.
BUG=
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 675e260ad1b2074bde22a8929d25d2640bd4e452 |
|
17-Oct-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check the number of playout channels instead of the send channels in StopPlayout()
BUG=2467
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2420004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4989 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 6c264cc92eb554716814db200b84752d4dfb6ba3 |
|
04-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Clean up AudioProcessing defaults and errors.
- Remove unneeded #defines and switch the remainder to consts.
- All AudioProcessing components are disabled by default, so remove
explicit disables.
- AudioProcessing uses a rational 16 kHz mono default, so no need to
explictly initialize.
- Add assert(false) to real-time errors which should not occur.
TESTED=trybots
R=bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2253005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4924 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 676ff1ed893d6ae59a7ce29a4428e0d7c9f855d9 |
|
07-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Ref-counted rewrite of ChannelManager.
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.
ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.
BUG=2081
R=tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1802004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 8fff1f065ea9d25970c3839294acdd606a5ddf22 |
|
31-Jul-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge r4394 from stable to trunk.
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Fixed the AGC and interface problems on the new path.
In order to make the AGC work properly, we need to cache the volume value passed
by the callback, compare it with the value returned by
shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to
return 0 to indicate no volume needs changing, otherwise return the new volume.
By doing this, we avoid setting the volume all the same, which allows the users
to change the volume manually.
This patch also fixes some minor issues with the interfaces too: make the int
channel[] const, and correct the order of the input params in
channel::Demultiplex.
R=tommi@webrtc.org
BUG=[2134]
TEST=compile && manual AGC test
Review URL: https://webrtc-codereview.appspot.com/1921004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 2f84afad30b088ddebb4063bc47ac9a79d735a2b |
|
31-Jul-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge r4326 from stable to trunk.
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.
Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| d900e8bea84c474696bf0219aed1353ce65ffd8e |
|
03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1434005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 9213521ea98b0977c7cdabd2853060835af226f3 |
|
14-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1463004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 3be565b502850f073fbfba2137a3d798464634b9 |
|
07-May-2013 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring for typing detection
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1370004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 6141e13873d0fdea626de08dfec2efa2c9171c76 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in voice_engine/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1305004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 0c45957e3a6963e1460c0b5b62a6adf43cf44314 |
|
03-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove UDP transport API from VoE
Review URL: https://webrtc-codereview.appspot.com/1236004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 684f0577fbe4ea393fef1dddf2ca7d02e3205b49 |
|
14-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 361bac7a4f30a81e58c53ba86c58ffec085306d7 |
|
13-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| a9a1df00351e07e16df7793dc98dbc5d2a5e9bea |
|
06-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the error return on SetAGC failure introduced by r3605.
BUG=webrtc:1464
Review URL: https://webrtc-codereview.appspot.com/1166005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3616 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 24045c5a02873ad98232e97857593abacf4c3a56 |
|
05-Mar-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise.
bug=issue1370
test=trybots
Review URL: https://webrtc-codereview.appspot.com/1121007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| f0a90c37c4b8a2581268f0054cc9d977e7beee8e |
|
05-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Expose the capture-side AudioProcessing object and allow it to be injected.
* Clean up the configuration code, including removing most of the weird defines.
* Add a unit test.
Review URL: https://webrtc-codereview.appspot.com/1152005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3605 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 0989fb7bfa482074e0161ea177653a44174ac492 |
|
15-Feb-2013 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VoiceEngineImpl inherit from VoiceEngine.
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).
Please see more details in the bug for how this is currently causing problems
with security tools.
BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voe_base_impl.cc
|