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29 Commits

Author SHA1 Message Date
henrik.lundin
f00082da37 Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h 
BUG=webrtc:6506
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device;master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng

Review-Url: https://codereview.webrtc.org/2544123003
Cr-Commit-Position: refs/heads/master@{#15414}
 2016-12-05 10:22:18 +00:00
henrik.lundin
e066b302ab Remove API-related #defines from voice_engine_configurations.h 
BUG=webrtc:6506
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device;master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng

Review-Url: https://codereview.webrtc.org/2549443002
Cr-Commit-Position: refs/heads/master@{#15379}
 2016-12-02 10:30:23 +00:00
henrik.lundin
06c1d6484e Prep to remove API-related #defines from voice_engine_configurations.h 
The follwing #defines are marked as deprecated:
- WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- WEBRTC_VOICE_ENGINE_HARDWARE_API
- WEBRTC_VOICE_ENGINE_FILE_API
- WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- WEBRTC_VOICE_ENGINE_CODEC_API
- WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2538093003
Cr-Commit-Position: refs/heads/master@{#15341}
 2016-11-30 16:21:54 +00:00
henrik.lundin
097529f34c Remove 3 defines in voice_engine_configurations.h 
WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are now gone.

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2530373002
Cr-Commit-Position: refs/heads/master@{#15309}
 2016-11-30 08:12:57 +00:00
henrik.lundin
ae0b3338e3 Prep to remove APM-related #defines from voice_engine_configurations.h 
Make sure that WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are always defined.

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2401393002
Cr-Commit-Position: refs/heads/master@{#14587}
 2016-10-10 14:24:58 +00:00
solenberg
11ace15c19 The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it. 
The following APIs are removed from VoEAudioProcessing:

  virtual int SetRxNsStatus(int channel,
                            bool enable,
                            NsModes mode = kNsUnchanged) = 0;
  virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) = 0;
  virtual int SetRxAgcStatus(int channel,
                             bool enable,
                             AgcModes mode = kAgcUnchanged) = 0;
  virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) = 0;
  virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
  virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
  virtual int RegisterRxVadObserver(int channel,
                                    VoERxVadCallback& observer) = 0;
  virtual int DeRegisterRxVadObserver(int channel) = 0;

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2295113002
Cr-Commit-Position: refs/heads/master@{#14227}
 2016-09-15 11:29:21 +00:00
tommi
31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ 
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.

BUG=

Review URL: https://codereview.webrtc.org/1607353002

Cr-Commit-Position: refs/heads/master@{#11346}
 2016-01-21 18:37:44 +00:00
ivoc
d66b44d565 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.

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}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
 2016-01-15 11:06:41 +00:00
ivoc
a4df27b671 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}
 2015-12-19 18:14:18 +00:00
ivoc
f4f5cb0927 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}
 2015-12-19 18:02:39 +00:00
ivoc
36d4c54500 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}
 2015-12-18 16:05:21 +00:00
ivoc
ae2c5ad12a 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}
 2015-12-18 11:53:42 +00:00
pbos
ad856229a7 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}
 2015-11-27 17:48:40 +00:00
Henrik Kjellander
98f53510b2 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}
 2015-10-28 17:17:50 +00:00
Peter Boström
d5c75b1a0b 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}
 2015-09-23 11:24:43 +00:00
Jelena Marusic
0d266054ac VoE: apply new style guide on VoE interfaces and their implementations 
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49239004

Cr-Commit-Position: refs/heads/master@{#9130}
 2015-05-04 12:15:41 +00:00
bjornv@webrtc.org
cc64a9cc4f 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
 2015-02-05 12:53:24 +00:00
henrikg@webrtc.org
863b536100 Allow opening an AEC dump from an existing file handle. 
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.

This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.

BUG=2567
R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 16:05:17 +00:00
pbos@webrtc.org
676ff1ed89 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
 2013-08-07 17:57:36 +00:00
pbos@webrtc.org
d900e8bea8 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
 2013-07-03 15:12:26 +00:00
andrew@webrtc.org
0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. 
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.

R=niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1683004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 17:03:47 +00:00
pbos@webrtc.org
956aa7e087 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
 2013-05-21 13:52:32 +00:00
pbos@webrtc.org
9213521ea9 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
 2013-05-14 08:31:39 +00:00
sjlee@webrtc.org
b4c441a785 Enable the below APIs for iOS. 
class VoEAudioProcessing
  int RegisterRxVadObserver();
  int DeRegisterRxVadObserver();
  int SetEcMetricsStatus();
  int GetEcMetricsStatus()
  int GetEchoMetrics();
  int GetEcDelayMetrics();

class VoENetEqStats
  int GetNetworkStatistics();

class VoEVolumeControl
  int SetChannelOutputVolumeScaling();
  int GetChannelOutputVolumeScaling();
Review URL: https://webrtc-codereview.appspot.com/1159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-25 11:12:20 +00:00
andrew@webrtc.org
f0a90c37c4 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
 2013-03-05 01:12:49 +00:00
tommi@webrtc.org
0989fb7bfa 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
 2013-02-15 15:07:32 +00:00
andrew@webrtc.org
50419b0777 Add libjingle-style stream-style logging. 
Add a highly stripped-down version of libjingle's base/logging.h. It is
a thin wrapper around WEBRTC_TRACE, maintaining the libjingle log
semantics to ease a transition to that format.

Also add some helper macros for easy API and function failure logging.

Review URL: https://webrtc-codereview.appspot.com/931010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 19:07:54 +00:00
andrew@webrtc.org
ddcc9429e7 Check the channels in receive-side processing frames. 
The number of channels must be set correctly before calling ProcessStream. This
was preventing stereo frames from being processed.

Also fix voe_cmd_test, which wasn't enabling rx NS properly.

BUG=issue713, 7375579

Review URL: https://webrtc-codereview.appspot.com/929013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 18:39:40 +00:00
andrew@webrtc.org
14b43beb7c 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
 2012-10-22 18:19:23 +00:00