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.cc
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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.cc
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e5ae6f82374cb064d39cffffdb650840f734880f |
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14-Dec-2015 |
peah <peah@webrtc.org> |
Correcting the check for the return code produced by AudioProcessing::ProcessReverseStream(). Before the change, only -1 was considered to be an error. Allthough the error code scheme for AudioProcessing definitely could be discussed, the current scheme have many error codes that differ from -1 and thus were not caught by the old code. BUG=webrtc:5237 Review URL: https://codereview.webrtc.org/1515073004 Cr-Commit-Position: refs/heads/master@{#11003}
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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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.cc
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98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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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/voice_engine/output_mixer.cc
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0f4b3731c34e796da92572380855dbc7321c8cfe |
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31-Aug-2015 |
minyuel <minyue@webrtc.org> |
Stylizing AudioConferenceMixer. Cleaning AudioConferenceMixer APIs to match Chromium style guide. Main changes: 1. change all mutable references to pointers 2. add const to all non-mutable references 3. add const to as many methods as possible BUG= R=andrew@webrtc.org Review URL: https://codereview.webrtc.org/1311733003 . Cr-Commit-Position: refs/heads/master@{#9821}
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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dce40cf804019a9898b6ab8d8262466b697c56e0 |
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24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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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.cc
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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.cc
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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.cc
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ddbb8a2c243f9d54cb0ce0092e341dfc6e126bb3 |
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22-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support arbitrary input/output rates and downmixing in AudioProcessing. Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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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.cc
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40ee3d07eda24b8e8214429d9885d9ad9a2c04f7 |
|
03-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Consolidate audio conversion from Channel and TransmitMixer. Replace the two versions with a single DownConvertToCodecFormat. As mentioned in comments, this could be further consolidated with RemixAndResample but we should write a full audio converter class in that case. Along the way: - Fix the bug present in Channel::Demultiplex with mono input and a stereo codec. - Remove the 32 kHz max from the OnDataAvailable path. This avoids a 48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we get a straight pass-through to ACM. The 32 kHz conversion is still needed in the RecordedDataIsAvailable path until APM natively supports 48 kHz. - Merge resampler improvements from ACM1 to ACM2. This allows ACM to handle 44.1 kHz audio passed to VoE and was originally done here: https://webrtc-codereview.appspot.com/1590004 - Reuse the RemixAndResample unit tests for DownConvertToCodecFormat. - Remove unused functions from utility.cc. BUG=3155,3000,b/12867572 TESTED=voe_cmd_test using both the OnDataAvailable and RecordedDataIsAvailable paths, with a captured audio format of all combinations of {44.1,48} kHz and {1,2} channels, running through all codecs, and finally using both ACM1 and ACM2. R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11019005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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944cbeb2926feb86a687e5fda9e2ac88ea8e3001 |
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18-Mar-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Resolves TSan v2 warnings in voe_auto_test. See bug report for details. BUG=1590 R=tommi@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5714 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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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.cc
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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.cc
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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.cc
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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.cc
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ae1a58bba4f926d149a5f39243269c3f6625f494 |
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22-Jan-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace AudioFrame's operator= with CopyFrom(). Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders. Review URL: https://webrtc-codereview.appspot.com/1031007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/output_mixer.cc
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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.cc
|