ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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788acd17adf6b3d605b5ea66cf394eb81fc086a9 |
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15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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12acd6ea8c5dcec953061144b82002d984967807 |
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11-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reorder includes in audio_processing_impl_unittest. TBR=aluebs Review URL: https://webrtc-codereview.appspot.com/9779005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5680 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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a8b97373d5d3154357cc6589ff949ee9f6f99d8d |
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10-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add tests and modify tools for new float deinterleaved interface. - Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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e84978f3d8612e7e482791552b94e0847967d3ba |
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25-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a Config parameter to AudioProcessing::Create(). Also add a parameter-less version; the (int) version is deprecated and should be removed. TBR=aluebs,bjornv BUG=2844 Review URL: https://webrtc-codereview.appspot.com/7609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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60730cfe3ce80e4023cd678373456cb703f000a4 |
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07-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the requirement to call set_sample_rate_hz and friends. Instead have ProcessStream transparently handle changes to the stream audio parameters (sample rate and channels). This removes two locks per 10 ms ProcessStream call taken by VoiceEngine (four total with the audio level indicator.) Also, prepare future improvements by having the splitting filter take a length parameter. This will allow it to work at different sample rates. Remove the useless splitting_filter wrapper. TESTED=voe_cmd_test with audio processing enabled and switching between codecs; unit tests. R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
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