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7 commits

Author SHA1 Message Date
Alex Narest
bbeb10925e Reporting audio device underrun counter 
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
 2019-08-16 11:49:55 +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
henrika
32026c3078 Removes Set/GetLoudspeakerStatus APIs from the ADM. 
int32_t SetLoudspeakerStatus(bool enable)
int32_t GetLoudspeakerStatus(bool* enabled) const

These APIs are only implemented on iOS and they do not belong in the
native audio layer since the client can achieve the same functionality
by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h.
It also gives the client a better flexibility in how the audio routing is done.

Bug: webrtc:7306
Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b
Reviewed-on: https://webrtc-review.googlesource.com/23740
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20721}
 2017-11-16 19:44:24 +00:00
henrika
c97cf03ede Removes unused sample-rate APIs from the ADM. 
The following four methods are removed:

SetRecordingSampleRate(const uint32_t samplesPerSec)
RecordingSampleRate(uint32_t* samplesPerSec) const
SetPlayoutSampleRate(const uint32_t samplesPerSec)
PlayoutSampleRate(uint32_t* samplesPerSec) const

Bug: webrtc:7306
Change-Id: I2c3c2e7bd3fb1264da197699fd5de15ab6c35c1b
Reviewed-on: https://webrtc-review.googlesource.com/22001
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20703}
 2017-11-16 08:59:53 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros. 
This CL has been generated with the following script:

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
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
 2017-11-09 11:56:32 +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/audio_device_generic.cc (Browse further)