Starting from Chromium Roll [1], WebRTC should start to use NDK r16
for Android builds. The roll cannot be completed because of three
compilation errors:
../../sdk/android/src/jni/pc/androidnetworkmonitor.cc:15:9: error: 'RTLD_NOLOAD' macro redefined [-Werror,-Wmacro-redefined]
^
../../third_party/android_tools/ndk/sysroot/usr/include/dlfcn.h:62:9: note: previous definition is here
../../modules/audio_device/android/audio_record_jni.cc:251:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
../../modules/audio_device/android/audio_track_jni.cc:229:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
This CL forward fixes these errors in order to fix the Chromium Roll
into WebRTC.
[1] - https://webrtc-review.googlesource.com/c/src/+/37540
Bug: webrtc:8710
Change-Id: I5bc64e73919eee7c9e965a442a386b5e1897b56a
Reviewed-on: https://webrtc-review.googlesource.com/37640
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21510}
This CL is part of merging the helper functions for audio and non-audio JNI code.
The GetThreadInfo() function is unrelated to JNI and I would prefer not to keep
it in a JNI helper file. Also, GetThreadInfo() is a very small function and inlining
it makes it simpler and more transparent IMO, as well as removing a lot of unnecessary
std::string creations.
Bug: webrtc:8689
Change-Id: I7d238fee826d310c0f5343d18b92d0dff864fd6a
Reviewed-on: https://webrtc-review.googlesource.com/36302
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21466}
In order to create a clean CL to switch to RTC_ prefixed LOG macros
this CL runs `git cl format --full` on the files with LOG macros in
the following directories:
- modules/audio_device
- modules/media_file
- modules/video_capture
This CL has been automatically generated with:
for m in PLOG \
LOG_TAG \
LOG_GLEM \
LOG_GLE_EX \
LOG_GLE \
LAST_SYSTEM_ERROR \
LOG_ERRNO_EX \
LOG_ERRNO \
LOG_ERR_EX \
LOG_ERR \
LOG_V \
LOG_F \
LOG_T_F \
LOG_E \
LOG_T \
LOG_CHECK_LEVEL_V \
LOG_CHECK_LEVEL \
LOG
do
for d in media_file video_capture audio_device; do
cd modules/$d
git grep -l $m | grep -E "\.(cc|h|m|mm)$" | xargs sed -i "1 s/$/ /"
cd ../..
done
done
git cl format --full
Bug: webrtc:8452
Change-Id: I2858b6928e6bd79957f2e5e0b07028eb68a304b2
Reviewed-on: https://webrtc-review.googlesource.com/21322
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20613}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_device/android/audio_track_jni.cc (Browse further)
