6955870806624479723addfae6dcf5d13968796c |
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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/voice_engine/output_mixer.h
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66085beef83c790a69666b9be8a74bb2eee44fab |
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16-Dec-2015 |
peah <peah@webrtc.org> |
Bugfix that fixes the error where the audio processing module is called using the wrong sample rate for the render signal. The CL is basically a partial revert of the related changes done on output_mixer.cc in the CL https://codereview.webrtc.org/1234463003. The CL also reverts the removal of the input_sample_rate_hz() method that was removed as part of the CL https://codereview.webrtc.org/1379123002 (as it was at that point no longer used). It should be noted that this CL turns off the effect of the IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are used. While it may be possible to solve that by adding upsampling after the API call, that approach was discarded due to that: -That would add extra processing in the echo path, leading to possible AEC performance reduction. -That would add extra complexity for the mobile case. -That would only patch the intelligibility enhancer operation as the proper way to do such an operation is within APM. -The intelligibility enhancer is not active by default anywhere. BUG=webrtc:5237 Review URL: https://codereview.webrtc.org/1525173002 Cr-Commit-Position: refs/heads/master@{#11045}
/external/webrtc/webrtc/voice_engine/output_mixer.h
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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/voice_engine/output_mixer.h
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60d9b332a5391045439bfb6a3a5447973e3d5603 |
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14-Aug-2015 |
ekmeyerson <ekmeyerson@webrtc.org> |
Integrate Intelligibility with APM - Integrates intelligibility into audio_processing. - Allows modification of reverse stream if intelligibility enabled. - Makes intelligibility available in audioproc_float test. - Adds reverse stream processing to audioproc_float. - (removed) Makes intelligibility toggleable in real time in voe_cmd_test. - Cleans up intelligibility construction, parameters, constants and dead code. TBR=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1234463003 Cr-Commit-Position: refs/heads/master@{#9713}
/external/webrtc/webrtc/voice_engine/output_mixer.h
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4540ffacc774b2c9a54155c4312dfe8263fd4de3 |
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28-Jul-2015 |
Minyue <minyue@webrtc.org> |
Removing AudioMixerStatusReceiver and ParticipantStatistics. BUG=webrtc:497 R=ajm@chromium.org, andrew@webrtc.org, henrikg@webrtc.org Review URL: https://codereview.webrtc.org/1216133004 . Cr-Commit-Position: refs/heads/master@{#9647}
/external/webrtc/webrtc/voice_engine/output_mixer.h
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9f277350f8341ee6813032ed4251e4b905e55e06 |
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14-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12299005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6146 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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f5a33f145b74d9c6058c670baf7b6201b78f6e48 |
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19-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Resampler modifications in preparation for arbitrary audioproc rates. - Templatize PushResampler to support int16 and float. - Add a helper method to PushSincResampler to compute the algorithmic delay. This is a prerequisite of: http://review.webrtc.org/9919004/ BUG=2894 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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5692531f18cae04d8a8107793dc74ae932bdf219 |
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14-Apr-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added a new OnMoreData() interface which will not feed the playout data to APM. BUG=3147 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11059005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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d900e8bea84c474696bf0219aed1353ce65ffd8e |
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03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments. BUG= R=mflodman@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1760006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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9213521ea98b0977c7cdabd2853060835af226f3 |
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14-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove const for plain data types in voice_engine/ BUG=1644 R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1463004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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50b2efef6ecb51a9d5818345c58533c5d236ec29 |
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29-Apr-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a wrapper around PushSincResampler and the old Resampler. The old resampler is used whenever it supports the requested rates. Otherwise the sinc resampler is enabled. Integrated with output_mixer in order to test the change through output_mixer_unittest. The sinc resampler will not yet be used, since we don't feed VoE with any rates that trigger it. BUG=webrtc:1395 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1355004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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6141e13873d0fdea626de08dfec2efa2c9171c76 |
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09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in voice_engine/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1305004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.h
|