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/voice_engine/transmit_mixer.h
|
ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
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/transmit_mixer.h
|
cdfe20bfc1146030aa59eb37635fd2fbcecd6cdb |
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23-Sep-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Fix the maximum native sample rate in AudioProcessing BUG=webrtc:4983 R=andrew@webrtc.org, henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1338833002 . Cr-Commit-Position: refs/heads/master@{#10037}
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
|
dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
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/transmit_mixer.h
|
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/voice_engine/transmit_mixer.h
|
6b02eea6acb571175ed220137ec44c841df6f535 |
|
12-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6103 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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f5a33f145b74d9c6058c670baf7b6201b78f6e48 |
|
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/transmit_mixer.h
|
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/transmit_mixer.h
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75dd2885c5e8fe1db6ea4384fe744e0bdecdcaeb |
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11-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add an interface for accepting keypress signals to AudioProcessing. R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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c693704cc209083ae439c312ff89bec5a2cf23d0 |
|
30-Jan-2014 |
henrikg@webrtc.org <henrikg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move out typing detection to its own class. This will allow an embedder to use it directly. Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.) R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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023cc5abc7d25fb3133b4d0206b67dcc6204b6e8 |
|
11-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Minor voice engine improvements around AGC. - Remove one unneeded lock in CaptureLevel(), as the call to this method should always come on the same thread as PrepareDemux(). - Remove check on analog AGC before doing volume calculations. Saves a bit of code. Instead check if the incoming volume is set to zero, which is a potentially common occurrence as it indicates no volume is available. R=aluebs@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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0062a6d0995c24f423504aba83b3f086cabec924 |
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24-Dec-2013 |
braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the include guard in transmit_mixer.h R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5334 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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a7cfa6704af8ac20acec4d195088053f7043b66f |
|
24-Dec-2013 |
braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the include guard in transmit_mixer.h R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5333 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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bf00740c92839865f3656fb4ee02b144f26b2012 |
|
17-Sep-2013 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a new voice engine warning for the typing noise off state. The old VE_TYPING_NOISE_WARNING is unchanged and fired whenever typing noise is detected. The new VE_TYPING_NOISE_OFF_WARNING is fired when typing noise was detected and is gone now. This is necessary for converting the typing state to a PeerConnection stats. R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4770 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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8fff1f065ea9d25970c3839294acdd606a5ddf22 |
|
31-Jul-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge r4394 from stable to trunk. r4326 was mistakenly committed to stable, so this is to re-merge back to trunk. Fixed the AGC and interface problems on the new path. In order to make the AGC work properly, we need to cache the volume value passed by the callback, compare it with the value returned by shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to return 0 to indicate no volume needs changing, otherwise return the new volume. By doing this, we avoid setting the volume all the same, which allows the users to change the volume manually. This patch also fixes some minor issues with the interfaces too: make the int channel[] const, and correct the order of the input params in channel::Demultiplex. R=tommi@webrtc.org BUG=[2134] TEST=compile && manual AGC test Review URL: https://webrtc-codereview.appspot.com/1921004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
|
2f84afad30b088ddebb4063bc47ac9a79d735a2b |
|
31-Jul-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge r4326 from stable to trunk. r4326 was mistakenly committed to stable, so this is to re-merge back to trunk. Add new interface to support multiple sources in webrtc. CaptureData() will be called by chrome with a flag |need_audio_processing| to indicate if the data needs to be processed by APM or not. Different from the old interface that will send the data to all voe channels, the new interface will specify a list of voe channels that the data is demultiplexing to. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
|
d900e8bea84c474696bf0219aed1353ce65ffd8e |
|
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/transmit_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/transmit_mixer.h
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3be565b502850f073fbfba2137a3d798464634b9 |
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07-May-2013 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring for typing detection R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1370004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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28e82bfec6db40aa9bb9aeeb3248f5d6d72e0b84 |
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02-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace Resampler with PushResampler in transmit_mixer. * 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
/external/webrtc/webrtc/voice_engine/transmit_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/transmit_mixer.h
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2412085bc1d95bcae2ee6987206b73bd1c2f5080 |
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02-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Don't upsample the capture signal early. * Remove the unneeded _mixingFrequency. * Rename CheckForSendCodecChanges to better elucidate its function. * Remove an unnecessary memcpy. Upsampling should be done late in the chain. This is practically relevant on mobile, where the capture rate is fixed at 16 kHz. When using Opus, the signal was upsampled to 32 kHz and was no longer compatible with AECM, which only supports up to 16 kHz. NEEDS_QA=true TEST=run calls with a variety of capture device rates and codecs BUG=chromium:178040,webrtc:1446 Review URL: https://webrtc-codereview.appspot.com/1146004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3594 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_mixer.h
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6be1e934ad48ccdac734b5cbd50528235ec93816 |
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01-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly error check calls to AudioProcessing. Checks must be made with "!= 0", not "== -1". Additionally: * Clean up the function calling into AudioProcessing. * Remove the unused _noiseWarning. * Make the other warnings bool. BUG=chromium:178040 Review URL: https://webrtc-codereview.appspot.com/1147004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/transmit_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/transmit_mixer.h
|