Having this script being located in the WebRTC repo doesn't make sense
since it has no connection to the source code.
Updating this script should apply to all build configurations and since
this script will be used for Chromium builders, we'll end up with having
to wait for a new WebRTC revision to be rolled in DEPS before it's updated.
TEST=none
BUG=none
TBR=phoglund
Review URL: https://webrtc-codereview.appspot.com/1444006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3992 4adac7df-926f-26a2-2b94-8c16560cd09d
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:
error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~
static_cast
This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).
Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
AutoTestSleep(std::numeric_limits<long>::max());
~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This fixes the errors and is required before stable can be rolled in Chromium.
TBR=mflodman,andresp
Review URL: https://webrtc-codereview.appspot.com/1450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.
TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1438004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.
Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling.
This also removes WEBRTC_PA_GTALK which was not defined anywhere.
BUG=webrtc:1395
TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3955 4adac7df-926f-26a2-2b94-8c16560cd09d
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.
BUG=webrtc:1395
TESTED=Set capture device to 44.1 and render device to 48 and vice versa and observed good AEC. The quality is considerably worse before this change. Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3954 4adac7df-926f-26a2-2b94-8c16560cd09d
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/
Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).
BUG=issue1024
Review URL: https://webrtc-codereview.appspot.com/1342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.
BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
