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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
712074270169d51b7d9af11c8b5b6f1cff76e2b4
/
modules
/
audio_coding
/
test
/
RTPFile.cc
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/modules/audio_coding/test/RTPFile.cc]
a9a6d4b
Delete voice_engine_configurations.h
by henrik.lundin
· 8 years ago
7056be9
Delete old video defines in engine config.
by mflodman
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
f98dc10
GN: Add target for modules_tests.
by ehmaldonado
· 8 years ago
3e6db23
audio_coding: remove "main" directory
by kjellander
· 9 years ago
[Renamed from webrtc/modules/audio_coding/main/test/RTPFile.cc]
98f5351
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
d324546
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
by pkasting@chromium.org
· 10 years ago
86e1e48
Move system_wrappers.gyp files to the proper directory.
by andresp@webrtc.org
· 10 years ago
4591fbd
Use size_t more consistently for packet/payload lengths.
by pkasting@chromium.org
· 10 years ago
741711a
Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.
by henrik.lundin@webrtc.org
· 10 years ago
a941970
Change explicit static cast from int to uint16_t to implicit cast of 0u.
by fbarchard@google.com
· 10 years ago
3c0aae1
Change gflags and gmock includes to be full paths.
by kjellander@webrtc.org
· 10 years ago
9328f39
cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error
by fbarchard@google.com
· 10 years ago
6ac22e6
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
by henrike@webrtc.org
· 10 years ago
d5726a1
Formatting ACM tests
by tina.legrand@webrtc.org
· 11 years ago
0946a56
WebRtc_Word32 => int32_t etc. in audio_coding/
by pbos@webrtc.org
· 12 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago
[Renamed from src/modules/audio_coding/main/test/RTPFile.cc]
354b0ed
Check return result of fwrite [Audio Module]
by leozwang@webrtc.org
· 12 years ago
16b6b90
Split of stereo packets moved
by tina.legrand@webrtc.org
· 13 years ago
975e4a3
Fix gcc warnings triggered by -Wextra.
by andrew@webrtc.org
· 13 years ago
7f3c724
Renaming 47 files from .cpp to .cc
by kjellander@webrtc.org
· 13 years ago
[Renamed from src/modules/audio_coding/main/test/RTPFile.cpp]
5490c71
Converted to gtest, writing output files properly and no longer uses exceptions.
by kjellander@webrtc.org
· 13 years ago
9775a30
Added variable to catch return value.
by tina.legrand@webrtc.org
· 13 years ago
554ae1a
Changes to solve warnings on Mac, issue #178.
by tina.legrand@webrtc.org
· 13 years ago
543c3ea
Fixing Release compilation errors
by kjellander@webrtc.org
· 13 years ago
470e71d
by niklase@google.com
· 13 years ago