This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1702983002
Cr-Commit-Position: refs/heads/master@{#11658}
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}
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}
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}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=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 -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
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
