| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 1323fc39babb2638c5795c1e23e97c977c5c29f3 |
|
11-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Remove webrtc/test/channel_transport/include
Move the header file into webrtc/test/channel_transport instead.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
R=henrika@webrtc.org, henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1431983006 .
Cr-Commit-Position: refs/heads/master@{#10595}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| d094c04baffabfa914682050615b97faea939bd1 |
|
30-Sep-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Remove AgcManager.
It was not used anywhere.
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1299143003 .
Cr-Commit-Position: refs/heads/master@{#10113}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 0b6a204b21b1360950593dcc80dc49682d637109 |
|
23-Jul-2015 |
andrew <andrew@webrtc.org> |
Configure AudioProcessing directly in agc_harness.
This allows us to configure create-time parameters for AudioProcessing
in a voice engine app and avoid the onerous SetExtraOptions.
voe_cmd_test would require significant refactoring to do the same.
Minor cleanups:
- Use agc_manager_direct. This should allow us to remove agc_manager.
- Use CHECKs rather than ASSERTs.
Review URL: https://codereview.webrtc.org/1247033006
Cr-Commit-Position: refs/heads/master@{#9618}
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| a33f05e8d7f293b5984b3cd7695eadefd16dcaba |
|
29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Re-land "Remove <(webrtc_root) from source file entries."
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.org
TBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 1ece0cbbec63d4fc14e3a6121d3d828c250fc20f |
|
29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert "Remove <(webrtc_root) from source file entries."
And the follow-up fix in r8198 that was not sufficient.
Reason: breaks Chromium bots runhooks (GYP).
I will have to try some more to make sure I don't
include test code, since include_tests==0 in Chromium.
TBR=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37039004
Cr-Commit-Position: refs/heads/master@{#8200}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|
| 2d2a1f9f056bb552e725b70b863d31cbee5ef7d8 |
|
29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove <(webrtc_root) from source file entries.
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.
Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).
I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.
BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37859004
Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/agc_harness.cc
|