The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8024 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8023 4adac7df-926f-26a2-2b94-8c16560cd09d
Due to lack of atomics our tracing code is broken and triggering real
errors in ThreadSanitizer.
R=kjellander@webrtc.org
BUG=2497
TEST=out-tsan/out/Debug/libjingle_media_unittest --gtest_filter=WebRtcVideoMediaChannelTest.GetStatsMultipleRecvStreams + verifying that "race:*trace_event_unique_catstatic*" exists in the list of matched suppressions.
Review URL: https://webrtc-codereview.appspot.com/35719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8022 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
This enables OpenSSL by default for Windows, see
8e72e1d..271c6cc/build/common.gypi
which required libjingle_tests.gyp to be updated since the
targets in third_party/nss/nss.gyp was moved into a condition in
https://codereview.chromium.org/694643002.
New Android dependencies are required due to being introduced in
build/android/pylib/remote/device/remote_device_test_run.py
of 5c49978f09
This should also fix Android test execution that started failing after
https://codereview.chromium.org/815213002 was submitted, since
it's based on e2a338fac9
Relevant other changes:
* src/buildtools: 535aff2..23a4e2f
* src/third_party/android_tools: 4f723e2..8fe116f
* src/third_party/boringssl/src: 00505ec..306e520
* src/third_party/icu: 53ecf0f..51c1a4c
* src/third_party/libvpx: 9fbec81..d3f3dce
* src/tools/swarming_client: 1d4965c..119b084
Details: 8e72e1d..271c6cc/DEPS
Clang version updated 218707:223108:
8e72e1d..271c6cc/tools/clang/scripts/update.sh
Due to this, we had to disable deadlock detection for TSan
due to a bug in Clang (see webrtc:
BUG=4106
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8003 4adac7df-926f-26a2-2b94-8c16560cd09d
First unit test will create peer connection client, run
for a few second, close it and verify that there were
no any errors and local video was rendered.
Second unit test will run peer connection in a loopback mode.
To run the test from command line install AppRTCDemoTest.apk
and execute the command:
adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7991 4adac7df-926f-26a2-2b94-8c16560cd09d
On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.
BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d
1) Added SetMode() to SSLAdapter and OpenSSLAdapter so the mode can be set to
SSL_MODE_DTLS
2) OpenSSLAdapter overrides SendTo() and RecvFrom() to handle calls from
TurnPort via AsyncUdpSocket
3) OpenSSLAdapter derives from MessageHandler to implement an internal DTLS
timer
4) Updated SSLAdapter unit tests
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7981 4adac7df-926f-26a2-2b94-8c16560cd09d
Summary:
- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.
TBR=glaznev
BUG=4103,4109
Review URL: https://webrtc-codereview.appspot.com/31239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
