Commit graph

14 commits

Author SHA1 Message Date
Ivo Creusen
f1393e23a2 Add UMA histogram for actual Android buffer size
Previously a histogram was added to track the requested buffer size,
this CL adds a histogram for the actually used buffer size.

Bug: b/157429867
Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31385}
2020-05-29 11:14:55 +00:00
Ivo Creusen
bdb5830d69 Add UMA histogram for native audio buffer size in ms
The Android native audio code asks the OS to provide an appropriate
buffer size for real-time audio playout. We should add logging for this
value so we can see what values are used in practice.

Bug: b/157429867
Change-Id: I111a74faefc0e77b5c98921804d6625cba1b84af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176126
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@chromium.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31368}
2020-05-27 14:33:50 +00:00
Alex Narest
44dc241ae8 Allows configuration of playout audio buffer
Playout audio buffer length in Java audio device configuration with fieldtrial.

Bug: webrtc:10928
Change-Id: I79286f09591f4b2c6a6146f23d3dce92a29f6b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150657
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#29005}
2019-08-29 12:57:14 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Sebastian Jansson
c01367db40 Deprecating ThreadChecker specific interface.
All changes outside thread_checker.h are by:
s/CalledOnValidThread/IsCurrent/
s/DetachFromThread/Detach/

Bug: webrtc:9883
Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27494}
2019-04-08 16:58:07 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Yura Yaroshevich
278d03a42c Force alignment of JVM called functions.
Bug: webrtc:9050
Change-Id: I5a064769dac857d2a6afb5f28c556bbcca21f8c6
Reviewed-on: https://webrtc-review.googlesource.com/64160
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22578}
2018-03-23 10:20:55 +00:00
henrika
cb87efd7d3 Avoids issues with start of audio when audio was not initialized on Android
Bug: b/72444507
Change-Id: I44d6e03c13a49033682f8f0bdc10256f724068d3
Reviewed-on: https://webrtc-review.googlesource.com/48020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21959}
2018-02-08 15:04:39 +00:00
henrika
c77b528a20 Adds usage of RTC_LOG macros in JNI audio code on Android.
Based on discussion in https://webrtc-review.googlesource.com/c/src/+/37640

Bug: webrtc:8710
Change-Id: I645b6e08b0a97aac3fe31547cf42fc4ddc25bbf6
Reviewed-on: https://webrtc-review.googlesource.com/37980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21573}
2018-01-11 11:42:31 +00:00
Mirko Bonadei
f641687a80 Forward fixing WebRTC to compile with Android NDK r16.
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}
2018-01-08 07:27:32 +00:00
Magnus Jedvert
9185bde7ee Android: Remove GetThreadInfo()
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}
2018-01-02 10:32:21 +00:00
Mirko Bonadei
72c4250cab Formatting some files with LOG macros usage.
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}
2017-11-09 09:49:12 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
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)