ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames 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}
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
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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/webrtc/modules/audio_device/android/audio_track_jni.h
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dce40cf804019a9898b6ab8d8262466b697c56e0 |
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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/modules/audio_device/android/audio_track_jni.h
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ee369e4277e48624bb557f0264644ed19a40dd67 |
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25-May-2015 |
henrika <henrika@chromium.org> |
Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes BUG=NONE TEST=./webrtc/build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AudioDevice* R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51079004 Cr-Commit-Position: refs/heads/master@{#9271}
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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b26198972c1fcb4aa7abaf3895b007e301e7d5dc |
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18-May-2015 |
henrika <henrika@chromium.org> |
Adding support for OpenSL ES output in native WebRTC 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}
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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8324b525dce2c502bbd24b3946bbae207645cde9 |
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27-Mar-2015 |
henrika <henrika@chromium.org> |
Adding playout volume control to WebRtcAudioTrack.java. Also adds a framework for an AudioManager to be used by both sides (playout and recording). This initial implementation only does very simple tasks like setting up the correct audio mode (needed for correct volume behavior). Note that this CL is mainly about modifying the volume. The added AudioManager is only a place holder for future work. I could have done the same parts in the WebRtcAudioTrack class but feel that it is better to move these parts to an AudioManager already at this stage. The AudioManager supports Init() where actual audio changes are done (set audio mode etc.) but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the case here. Hence, the AM now serves as the center for getting audio parameters and then inject these into playout and recording sides. Previously, both sides acquired their own parameters and that is more error prone. BUG=NONE TEST=AudioDeviceTest R=perkj@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45829004 Cr-Commit-Position: refs/heads/master@{#8875}
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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474d1eb22376898b36bcd04b0ce3860fa12fd984 |
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09-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Adds C++/JNI/Java unit test for audio device module on Android. 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
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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962c62475e31ccb5b1315bf646138652e273d0f5 |
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23-Feb-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Refactoring WebRTC Java/JNI audio track in C++ and Java. This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I. - Removes unused code and old WEBRTC logging macros - Now uses optimal sample rate and buffer size in Java AudioTrack (used hard-coded sample rate before) - Makes code more inline with the implementation in Chrome - Adds helper methods for JNI handling to improve readability - Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy) - Simplified the delay estimate - Adds basic thread checks - Removes all locks in C++ land - Removes all locks in Java - Improves construction/destruction - Additional cleanup Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate). BUG=NONE R=magjed@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39169004 Cr-Commit-Position: refs/heads/master@{#8460} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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62f6e756730325ee7b20cf5f81e82b0a70283a05 |
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11-Feb-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Refactoring WebRTC Java/JNI audio recording in C++ and Java. This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC: - Removes unused code and old WEBRTC logging macros - Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before) - Makes code more inline with the implementation in Chrome - Adds helper methods for JNI handling to improve readability - Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy) - Adds basic thread checks - Removes all locks in C++ land - Removes all locks in Java - Improves construction/destruction - Additional cleanup Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate). BUG=NONE R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33969004 Cr-Commit-Position: refs/heads/master@{#8325} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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c7c432aa9b8c9f9ba6d41554917784a27b21426a |
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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/modules/audio_device/android/audio_track_jni.h
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573a1b45b5b7638605d9727be57c73e838d6ee45 |
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10-Jan-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android: Fixes crash when exiting WebRTCDemo. BUG=2738 R=fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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f6acf98a46ec62ee97a51a1549933c0783ea4355 |
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20-Dec-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the android clang bot for compiling with thread annotations. TBR=niklas.enbom@webrtc.org R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6279005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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179908c81c48076b680559aa1810a593c3383acd |
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18-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
JNI Audio: remove dead members. BUG=2735 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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9ee75e9c77b467e74e470905822d0279b0e8a639 |
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11-Dec-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). BUG=N/A R=fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_track_jni.h
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