2d110be77f14cab0bb51efe8b61d9c7a967d04cb |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) Reason for revert: tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach. Original issue's description: > Storing raw audio sink for default audio track. > > BUG=webrtc:5250 > > Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99 > Cr-Commit-Position: refs/heads/master@{#11230} TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1588693002 Cr-Commit-Position: refs/heads/master@{#11241}
/external/webrtc/webrtc/voice_engine/channel.h
|
e591f9377f33f3f725a30faecd1bef1a71fa6b99 |
|
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Storing raw audio sink for default audio track. BUG=webrtc:5250 Review URL: https://codereview.webrtc.org/1551813002 Cr-Commit-Position: refs/heads/master@{#11230}
/external/webrtc/webrtc/voice_engine/channel.h
|
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/channel.h
|
f888bb58da04c5095759b5ec7ce2e1fa2cd414fd |
|
12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Support for unmixed remote audio into tracks. BUG=chromium:121673 R=solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1505253004 . Cr-Commit-Position: refs/heads/master@{#10995}
/external/webrtc/webrtc/voice_engine/channel.h
|
b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE. This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
/external/webrtc/webrtc/voice_engine/channel.h
|
358057b945725390bcecc330513160aa823f651e |
|
27-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1482703002 Cr-Commit-Position: refs/heads/master@{#10828}
/external/webrtc/webrtc/voice_engine/channel.h
|
3e6db2321ccdc8738c9cecbe9bdab13d4f0f658d |
|
26-Nov-2015 |
kjellander <kjellander@webrtc.org> |
audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|
74f0f3551ecd596dc0f83146d218887082528fa8 |
|
01-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Delete a chain of methods in ViE, VoE and ACM The end goal is to remove AcmReceiver::SetInitialDelay. This change is in preparation for that goal. It turns out that AcmReceiver::SetInitialDelay was only invoked through the following call chain, where each method in the chain is never referenced from anywhere else (except from tests in some cases): ViEChannel::SetReceiverBufferingMode -> ViESyncModule::SetTargetBufferingDelay -> VoEVideoSync::SetInitialPlayoutDelay -> Channel::SetInitialPlayoutDelay -> AudioCodingModule::SetInitialPlayoutDelay -> AcmReceiver::SetInitialDelay The start of the chain, ViEChannel::SetReceiverBufferingMode was never referenced. This change deletes all the methods above except AcmReceiver::SetInitialDelay itself, which will be handled in a follow-up change. BUG=webrtc:3520 Review URL: https://codereview.webrtc.org/1421013006 Cr-Commit-Position: refs/heads/master@{#10471}
/external/webrtc/webrtc/voice_engine/channel.h
|
74640895fafbb90a6630a6a91b80da0a7cff229c |
|
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
/external/webrtc/webrtc/voice_engine/channel.h
|
1d8a506405734d0cef9653704b036ca4f1388960 |
|
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport. This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/webrtc/voice_engine/channel.h
|
2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
|
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces. BUG=webrtc:1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1369263002 Cr-Commit-Position: refs/heads/master@{#10096}
/external/webrtc/webrtc/voice_engine/channel.h
|
cdfe20bfc1146030aa59eb37635fd2fbcecd6cdb |
|
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/channel.h
|
ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces. Starts by removing channel/engine id from ViEChannel which propagates down to the RTP/RTCP module as well as the transport class. IncomingVideoStream::RenderFrame() is untouched for now but receives a fake id instead of the previous channel id. Added a TODO to remove it later but the RenderFrame call is implemented in a lot of platform-dependent files and should probably remove the "manager" aspect of renderers, so preferring to do it separately BUG=webrtc:1695 R=henrika@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1335353005 . Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/webrtc/voice_engine/channel.h
|
ae856f2c9fc358e5cd68d8a595136dcef017ed96 |
|
17-Sep-2015 |
Ivo Creusen <ivoc@webrtc.org> |
Added support for logging the SSRC corresponding to AudioPlayout events. To do this, the logging of this event was moved from the ACM to VoiceEngine Channel. A new LogAudioPlayoutEvent function was added on the RtcEventLog interface, and the LogDebugEvent function was removed since it is no longer being used. BUG=webrtc:4741 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, kwiberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/1340283002 . Cr-Commit-Position: refs/heads/master@{#9972}
/external/webrtc/webrtc/voice_engine/channel.h
|
b04965ccf83c2bc6e2758abab9bea0c18551a54c |
|
09-Sep-2015 |
ivoc <ivoc@webrtc.org> |
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. An option was added to voe_cmd_test to make a RtcEventLog dump. BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1267683002 Cr-Commit-Position: refs/heads/master@{#9901}
/external/webrtc/webrtc/voice_engine/channel.h
|
0f4b3731c34e796da92572380855dbc7321c8cfe |
|
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/channel.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/channel.h
|
743758816853df2040a21c5652b0d0e238b1512f |
|
13-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding locking to webrtc::voe::Channel to fix race conditions Some members are accessed from the video processing thread for the VoEVideoSync interface, and thus need to be protected. This is a problem that TSan sometimes reports. Also moved UpdatePlayoutTimestamp to private section since it's only needed internally. And renamed least_required_delay_ms to LeastRequiredDelayMs, since it no longer just returns a cached value. BUG=webrtc:4663 Review URL: https://codereview.webrtc.org/1263223002 Cr-Commit-Position: refs/heads/master@{#9706}
/external/webrtc/webrtc/voice_engine/channel.h
|
d436298332c7a7ecb51241f3a66588539c2ece83 |
|
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove ResetStatistics from RTP feedback. BUG= R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1213603002 Cr-Commit-Position: refs/heads/master@{#9548}
/external/webrtc/webrtc/voice_engine/channel.h
|
c3f4dbc40b9369a7f8eb9248adb8a018b9d8e439 |
|
20-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove rtp_rtcp/ dump functionality. Removes RTP dumping from VideoEngine and VoiceEngine. BUG=1695 R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47179004 Cr-Commit-Position: refs/heads/master@{#9234}
/external/webrtc/webrtc/voice_engine/channel.h
|
2013aeced2b7821a407f302802c4a16fd02728b1 |
|
13-May-2015 |
Minyue <minyue@webrtc.org> |
Propagating RTT from send-only channel to receive-only channel. This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly. BUG=3978 TEST=chromium with hangout calls R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29989004 Cr-Commit-Position: refs/heads/master@{#9186}
/external/webrtc/webrtc/voice_engine/channel.h
|
300eeb68f55c5091c7045e377578586733cddf16 |
|
12-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interfaces. Removes ViE interfaces, _impl.cc files, managers (such as ViEChannelManager and ViEInputManager) as well as ViESharedData. Interfaces necessary to implement observers have been moved to a corresponding header (such as vie_channel.h). BUG=1695, 4491 R=mflodman@webrtc.org, solenberg@webrtc.org TBR=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/55379004 Cr-Commit-Position: refs/heads/master@{#9179}
/external/webrtc/webrtc/voice_engine/channel.h
|
adf89b7e33cc54dab9365dddead687a46a074cf0 |
|
29-Apr-2015 |
Ivo Creusen <ivoc@webrtc.org> |
Added SetBitRate function to VoE API to allow changing the audio bitrate. If the requested bitrate is not valid for the codec, the codec will decide on an appropriate value. Updated VoE command line tool to use new SetBitRate function. Includes unittests for SetBitRate function. BUG= R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50789004 Cr-Commit-Position: refs/heads/master@{#9115}
/external/webrtc/webrtc/voice_engine/channel.h
|
9b2e1144df6e3622354caca00baf4a7462a0809c |
|
13-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Supporting Opus DTX in Voice Engine. Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API. BUG=1014 R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43709004 Cr-Commit-Position: refs/heads/master@{#8716} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
e9217b4bdbf9a8fd8b4882b2f995665927f28ad2 |
|
06-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove WebRtcACMEncodingType The parameter was not needed; it was sufficient with a bool indicating speech or not speech. This change propagates to the InFrameType callback function. Some tests are updated too. COAUTHOR=kwiberg@webrtc.org R=minyue@webrtc.org TBR=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42209004 Cr-Commit-Position: refs/heads/master@{#8626} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8626 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|
0a7d4eed98ccec0c2b3e7522e7b2dde1919a4ae3 |
|
17-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove usage of BitrateController in VoiceEngine. Bitrate controller is used in VoiceEngine to smoothen the fraction loss from RTCP report blocks. This CL removes the usage of the BitrateController and calculates its own fraction loss average insted. This introduces some duplicated code between BitrateController and Channel, but removes processing overhead and the incorrect bandwidth estimation numbers reported by the bitrate controller. BUG=4310 TEST=voe_cmd_test with network simulator R=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39999004 Cr-Commit-Position: refs/heads/master@{#8386} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8386 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
456f01441aa4f8c0c8b98aa6d9c2af4a4817e8db |
|
23-Jan-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Re-allowing RED in voice engine. Path of audio RED packets was blocked in r4692 by accident. It ought be enabled again. BUG=3619 R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8137 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
8315d7de8551963c53162e320835c158088fcdd6 |
|
14-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove dual stream functionality in VoiceEngine This is old code that is no longer in use. The clean-up is part of the ACM redesign work. The corresponding code in ACM will be deleted in a follow-up CL. BUG=3520 R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values. Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
d16e839c6d29831e79312180085b6a19149df43f |
|
19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rtp-Rtcp sender cleanup. Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions. Also removed const on non-pointer/reference types for related files. BUG= R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34469004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths. See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
ece3890d3a40fe911ae895e28c329491e795b14d |
|
14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report total bitrate for all streams in GetStats. This regression wasn't caught because I accidentally disabled multiple streams for EndToEndTest.GetStats in a refactoring. R=stefan@webrtc.org, xians@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/27179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
3cefbc99f4cc2db744cb130ca629768401a59eb4 |
|
10-Oct-2014 |
xians@webrtc.org <xians@webrtc.org> |
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. This also marks all virtual overrides of other classes in the same files. This will make a subsequent change I intend to do safer, where I'll change the argument types of the base Transport functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions. I've removed some of these. TBR=mflodman@webrtc.org, pkasting@chromium.org BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/28709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
4cebd84c792309c98aed9ba05524ce051341268b |
|
01-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland "Remove DTMF status methods from Voice Engine" r7276 This reverts r7277. TBR=henrika@webrtc.org,pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
3987f10c1142ffa07d749ce7b055b8a68892c19d |
|
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Remove DTMF status methods from Voice Engine" r7276 This change caused some trouble. TBR=henrika@webrtc.org,pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
bf7b9e0081233661ac0fe9500c0aa5b2aea70376 |
|
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove DTMF status methods from Voice Engine These methods are not used. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
64a2f10f4b566a91b358e77c4ecdf09ebb33ac59 |
|
22-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove Get/SetNetEQPlayoutMode APIs These are not used anymore. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
2b58a4433f16c510d56b1d16d2e3bb882b184861 |
|
11-Sep-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Calculating round-trip-time in send-only channel in VoE. TESTS=built chromium and tested with 1:1 hangout call BUG= R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7147 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
adee8f924224e116f041564ddde83c979880e35f |
|
03-Sep-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 BUG= R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
8e24d8777849951ed86fb01e0bf556d4eda65161 |
|
02-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. BUG=3681 R=pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7030 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
6aac93bd9c3da92e92b016d83c8f84c65aae65b6 |
|
12-Aug-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding SetOpusMaxBandwidth in VoE and ACM This is a step to solve https://code.google.com/p/webrtc/issues/detail?id=1906 In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth. TEST = added a test in voe_cmd_test and listened to the result BUG= R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
74aaf29a0ff1b211dbfdbb6309791111a7871779 |
|
16-Jul-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. The filter is an exponential filter borrowed from video coding module. The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic. BUG= R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
4ef438e2defd6c46404f6b367287364cde66b7fb |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine. These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
94454b71adc37e15fd3f5a5fc432063f05caabcb |
|
05-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the chain that propagates the audio frame's rtp and ntp timestamp including: * In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio. * When there're more than one participant, set AudioFrame's RTP timestamp to 0. * Copy ntp_time_ms_ in AudioFrame::CopyFrom method. * In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame. * Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency. Tweaks on ntp_time_ms_: * Init ntp_time_ms_ to -1 in AudioFrame ctor. * When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome. Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms. BUG=3111 R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org TBR=andrew andrew to take another look on audio_conference_mixer_impl.cc Review URL: https://webrtc-codereview.appspot.com/14559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
c1a40a7b68a8d253b0ba32b89f3126931eeaeab3 |
|
28-May-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. This CL is going to be combined with another CL in ACM, which is to be landed. TEST=passed_try_bots BUG= R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
82c4b8531c2c4c2aaf82ff57ee1805037a43ed50 |
|
21-May-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Calculate capture ntp timestamp in local timebase for decoded audio frame. BUG=3111 R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6205 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
cb711f77d2ff9ebd42678869a73353809b3af66e |
|
19-May-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add interface to propagate audio capture timestamp to the renderer. BUG=3111 R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|
66021e0fa230b56db0963b0f2297de8c4038cb6f |
|
12-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=niklas.enbom@webrtc.org, solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13489005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
382c0c209d323c1e6972d988a7b26f08fc2e8a6b |
|
05-May-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Allow the RTP level indicator computation to work at any sample rate. Break out the computation to a separate class, and call directly into this from channel.cc rather than going through AudioProcessing. This circumvents AudioProcessing's sample rate limitations. We now compute the RMS over all samples rather than downmixing to a single channel. This makes the call point in channel.cc easier, is more "correct" and should have similar (negligible) complexity. This caused slight changes in the RMS output, so the ApmTest.Process reference has been updated. Snippet of the failing output: [ RUN ] ApmTest.Process Running test 4 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 5 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 6 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 10 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 11 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 12 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 BUG=3290 TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc produce reasonable printed out results from RMS(). R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
93fd25c20c688961569d3631b875c8ee0dfc2a80 |
|
24-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. * Cast rtp header extension to int in log in rtp_utility.cc. BUG=3237 TEST=try bots R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
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/channel.h
|
66803489f9694cb7c7c0dd3ba07b63e2b6b71779 |
|
17-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12019005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
0f7375504a98e43101f682143ae8f3866aec3ed3 |
|
17-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
b9309beea40e1fd99297d4658a16864a801329c3 |
|
14-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes VoECallReport 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/11779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|
b1f50100757036cf475072c26f5f374eee9588ca |
|
24-Mar-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VoE changes to allow forwarding of packets from VoE to ViE BWE. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
944cbeb2926feb86a687e5fda9e2ac88ea8e3001 |
|
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/channel.h
|
ebdb0e3ad0a787bee066d12cdcd115a38b0a10d1 |
|
07-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. - Add ability to VoE to send Absolute Sender Time header extension. - Refactor handling of RTP header extensions in VoE to work the same as in ViE. - Add API to enable receiving Absolute Sender Time in VoE. This is part of the work to include audio packets in bandwidth estimation, for better accuracy in estimates. BUG= TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
b7a91fa95a63b82a8a1004988a0c070dcffd17f2 |
|
19-Feb-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes VoERTP_RTCP::InsertExtraRTPPacket. Reasons for removing: - Feels like a complete hack IMHO. - Not used by any client. - Unclear functionality regarding time stamp, marker bit etc. - Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead. - We also see lots of TSan races and memcheck errors related to this API. BUG=2296,2240 R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
a07923339bea76571f2f9ac33316eb56dfb47054 |
|
18-Feb-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove external encryption API for VoE. BUG= R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
60730cfe3ce80e4023cd678373456cb703f000a4 |
|
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/voice_engine/channel.h
|
54ae4ffb9e235a9742e2b11298327e02d870571c |
|
19-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTCP statistics. This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable. The change is primarily targeted at the new video engine API. TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up. BUG=2235 R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
167b6dfc73fc2f4c47713bcbd89b58c52612983b |
|
13-Dec-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix jitter buffer delay estimate. BUG=b/12099925 R=niklas.enbom@webrtc.org, niklase@google.com, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5289 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
24301a67c66e6091418e83da49cfb367ef2c6645 |
|
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58174641 together with http://review.webrtc.org/4319005/. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
48df38114d9502f4b4ad700c011190c608a702d5 |
|
08-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for making sure that the packet in order checks are done prior to updating the last received packet state. Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state in the rtp receiver to never get valid. Also makes sure that only valid timestamps and receive times are used for audio/video sync. BUG=2608 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
63420669746cfca6ed1d902c68c656b79ffa5a1b |
|
17-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix tsan failures in channel.cc regarding to the volume settings. BUG=2461 TEST=try bots R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2377004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4992 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
eb524d997b877ba9b0bcaf8c85eae52bd40c37e3 |
|
24-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove deprecated AudioCodingModule::Destroy. Have Channel hold a pointer rather than reference, and shorten the name. TESTED=trybots R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2256004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
f3930e941c15da48c037c62cdb1eebbcbf89c9c7 |
|
19-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Small refactoring of AudioProcessing use in channel.cc. - Apply consistent naming. - Use a scoped_ptr for rx_audioproc_. - Remove now unnecessary AudioProcessing::Destroy(). R=bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2184007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4784 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
e509f943eded156f7a8365b0b001abe73646acfa |
|
12-Sep-2013 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This issue is related to https://chromereviews.googleplex.com/9908014/ I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL. BUG= R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2171004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
|
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for combining RTX and FEC/RED. This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
286fe0b04d97205ac84688bbe613d5749192b2d1 |
|
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" ...and fixes the RTCP bug. BUG=2277 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
a0218a84d17a727111e2e24cf5af915b1b91c06e |
|
21-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4582 "Reverts a second set of reverts caused by a bug in ..." > Reverts a second set of reverts caused by a bug in a dependency. > > Revert "Revert r4328" > > Revert "Revert r4322 "Support sending multiple report blocks and keeping track > of statistics on" > > BUG=1811 > R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/2072004 TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2087004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
1a65d6c36b6a25f9f734176c697c684c3b43ac4b |
|
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reverts a second set of reverts caused by a bug in a dependency. Revert "Revert r4328" Revert "Revert r4322 "Support sending multiple report blocks and keeping track of statistics on" BUG=1811 R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2072004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584. Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
09e8c47ee5c58e5e86b09dc1950f8d3f9f24cd9f |
|
31-Jul-2013 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Merge r4374 from stable to trunk. r4374 was mistakenly committed to stable, so this is to re-merge back to trunk. Store the sequence number in StopSend() and resume it in StartSend(). When restarting the microphone device, we call StopSend() first, then StartSend() later. Since we reset sequence number in StopSend(), it sometimes causes libSRTP to complain about packets being replayed. Libjingle work around it by caching the sequence number in WebRtcVoiceEngine.cc, and call SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend(). When restarting the microphone device, we call StopSend() first, then StartSend() later. Since we reset sequence number in StopSend(), it sometimes causes libSRTP to complain about packets being replayed. Libjingle work around it by caching the sequence number in WebRtcVoiceEngine.cc, and call SetInitSequenceNumber() to resume the sequence number before StartSend(). This patch fixes this problem by storing the sequence number in StopSend(), and resume it in StartSend(). So that we can remove the workaround in libjingle. BUG=2102 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1922004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
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/channel.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/channel.h
|
aa4d96a134a03f998d52fb9699845d9c644eb24b |
|
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4301 R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
b7eda43810125cd01b29671a6beab61ddb48ebdb |
|
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on several SSRCs" R=pwestin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1774006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
717d147ebb168ed498fa4777ffaf8646a1dc6d7a |
|
10-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support sending multiple report blocks and keeping track of statistics on several SSRCs. BUG=1811 TEST=vie_auto_test --automated, voe_auto_test --automated, trybots R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1768004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
|
05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|
da710448b2f59a11a652988ef40a7d92bf5268ce |
|
07-Jun-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix size_t to int conversion error on Win64. TBR=pwestin Review URL: https://webrtc-codereview.appspot.com/1626005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4192 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
db249956807d916729aacfaf3382923e8239d533 |
|
05-Jun-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up Nack for Voe R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1614004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4184 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
a5cb98cbbd11e93cb6d0a6232387814aac168c7d |
|
29-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out RTP header parsing from the RTP module. This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
e46c8d387587ba148e229a7bb18f1cc0708a2a87 |
|
22-May-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. TEST=unit-test, manual, trybots. R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1384005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in voice_engine/ BUG=1662 R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1434005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
|
9213521ea98b0977c7cdabd2853060835af226f3 |
|
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/channel.h
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1de01354e68da71bc62c81af17afeac8ed374a18 |
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11-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding playout buffer status to the voe video sync Review URL: https://webrtc-codereview.appspot.com/1311004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
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0c45957e3a6963e1460c0b5b62a6adf43cf44314 |
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03-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove UDP transport API from VoE Review URL: https://webrtc-codereview.appspot.com/1236004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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a442d4d98337bc25e4c469e20fde62aab33e2f59 |
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28-Mar-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. Today I had to figure out this code was legacy. Now next person doesn't have to. BUG= Review URL: https://webrtc-codereview.appspot.com/1247004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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684f0577fbe4ea393fef1dddf2ca7d02e3205b49 |
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14-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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361bac7a4f30a81e58c53ba86c58ffec085306d7 |
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13-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. Review URL: https://webrtc-codereview.appspot.com/1029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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b7edd065306329309dac6767fe4914c185f941f8 |
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12-Mar-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove DTMF detection. Talk team has been in the loop and there is no need for DTMF detection at the receiver side. test=voe_auto_test, VoE extended test DTMF Review URL: https://webrtc-codereview.appspot.com/1168004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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24045c5a02873ad98232e97857593abacf4c3a56 |
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05-Mar-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. bug=issue1370 test=trybots Review URL: https://webrtc-codereview.appspot.com/1121007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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6388c3e2fdfc91b3648fb7d408a14ddb25e41cd1 |
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12-Feb-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. TEST=ACM unit test is added, also a manual integration test is writen. Review URL: https://webrtc-codereview.appspot.com/1097009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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1b60ceb499ee35460886f2bfaecee1f47319f925 |
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13-Dec-2012 |
roosa@google.com <roosa@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add GetAudioFrame API to VoiceEngine. Allows the caller to pull frames from a channel instead of sending them to the output mixer. BUG= Review URL: https://webrtc-codereview.appspot.com/973012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3273 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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0870f02cdbcce7de8c6a4dceb6d1678c2c6c518f |
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12-Dec-2012 |
roosa@google.com <roosa@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add API to retreive last received RTP timestamp to VoiceEngine. BUG= Review URL: https://webrtc-codereview.appspot.com/969016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3271 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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42259e7ebc7126f5a7036940fcab65b3f8d2af38 |
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11-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VoE Changes to enable dual_streaming. TEST=added new unit-test This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/ which is under review. Should be committed after issue 933015 is committed. Committed: https://code.google.com/p/webrtc/source/detail?r=3231 Review URL: https://webrtc-codereview.appspot.com/970005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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2cf22a6abce2d38e673505a4cfd5624a3710b5cd |
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04-Dec-2012 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 3231 - VoE Changes to enable dual_streaming. TEST=added new unit-test This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/ which is under review. Should be committed after issue 933015 is committed. Review URL: https://webrtc-codereview.appspot.com/970005 TBR=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/929040 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3236 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.h
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767d87cf24be9a3239e0bc26ad9f3e99604615f8 |
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03-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VoE Changes to enable dual_streaming. TEST=added new unit-test This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/ which is under review. Should be committed after issue 933015 is committed. Review URL: https://webrtc-codereview.appspot.com/970005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3231 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/channel.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/channel.h
|