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webrtc_m130/media/base/media_constants.h

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Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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/*
Change to WebRTC license in webrtc/media This was decided to be done in a separate CL from the move that took place in https://codereview.webrtc.org/1587193006/ BUG=webrtc:5420 NOTRY=True TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1676923002 Cr-Commit-Position: refs/heads/master@{#11520}
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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*
Change to WebRTC license in webrtc/media This was decided to be done in a separate CL from the move that took place in https://codereview.webrtc.org/1587193006/ BUG=webrtc:5420 NOTRY=True TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1676923002 Cr-Commit-Position: refs/heads/master@{#11520}
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* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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*/
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
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#ifndef MEDIA_BASE_MEDIA_CONSTANTS_H_
#define MEDIA_BASE_MEDIA_CONSTANTS_H_
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
#include <stddef.h>
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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Export symbols needed by the Chromium component build (part 4). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: I12ef6f85ccef7dae3afe9ecff99725af13d551e2 Reviewed-on: https://webrtc-review.googlesource.com/c/106684 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25246}
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#include "rtc_base/system/rtc_export.h"
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// This file contains constants related to media.
namespace cricket {
extern const int kVideoCodecClockrate;
build: merge media_constants and engine_constants no functional changes BUG=None Change-Id: I994cf7de6fdbf5505ed3359e08700cac5ea9fe3f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202022 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33246}
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extern const int kVideoMtu;
extern const int kVideoRtpSendBufferSize;
extern const int kVideoRtpRecvBufferSize;
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// Default CPU thresholds.
extern const float kHighSystemCpuThreshold;
extern const float kLowSystemCpuThreshold;
extern const float kProcessCpuThreshold;
Revert r6110 and r6109. Effectively re-landing r6104 as well as r6108 which includes a fix to a compile error that caused r6104 to be reverted in r6110. BUG= TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kRedCodecName[];
extern const char kUlpfecCodecName[];
Add new codec for FlexFEC. This CL does nothing except adding new strings and enums corresponding to the new codec. R=stefan@webrtc.org BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2470103002 Cr-Commit-Position: refs/heads/master@{#14943}
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extern const char kFlexfecCodecName[];
Reland "Rename stereo video codec to multiplex" This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174. Original change's description: > Rename stereo video codec to multiplex > > This CL only does the rename from"stereo" to multiplex". With this we have a > better name that doesn't clash with audio's usage of stereo. > > Bug: webrtc:7671 > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652 > Reviewed-on: https://webrtc-review.googlesource.com/43242 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21769} TBR=niklas.enbom@webrtc.org Bug: webrtc:7671 Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511 Reviewed-on: https://webrtc-review.googlesource.com/45820 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21794}
2018-01-25 13:01:09 -08:00
extern const char kMultiplexCodecName[];
Add new codec for FlexFEC. This CL does nothing except adding new strings and enums corresponding to the new codec. R=stefan@webrtc.org BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2470103002 Cr-Commit-Position: refs/heads/master@{#14943}
2016-11-07 03:03:41 -08:00
extern const char kFlexfecFmtpRepairWindow[];
Revert r6110 and r6109. Effectively re-landing r6104 as well as r6108 which includes a fix to a compile error that caused r6104 to be reverted in r6110. BUG= TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:07:01 +00:00
rtx-time implementation provides an implementation of the rtx-time parameter from https://tools.ietf.org/html/rfc4588#section-8 that determines the maximum time a receiver waits for a frame before sending a PLI. BUG=webrtc:12420 Change-Id: Iff20d92c806989cd4d56fe330d105b3dd127ed24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201400 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33627}
2021-03-24 17:49:02 +01:00
extern const char kRtxCodecName[];
extern const char kCodecParamRtxTime[];
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kCodecParamAssociatedPayloadType[];
rtx-time implementation provides an implementation of the rtx-time parameter from https://tools.ietf.org/html/rfc4588#section-8 that determines the maximum time a receiver waits for a frame before sending a PLI. BUG=webrtc:12420 Change-Id: Iff20d92c806989cd4d56fe330d105b3dd127ed24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201400 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33627}
2021-03-24 17:49:02 +01:00
Add optional stereo codec to SDP negotiation - Defines stereo codec case, similar to RTX, that adds stereo codec to the SDP negotiation. The underlying codec's payload type is similarly defined by "apt". - If this negotiation is successful, codec name is included in sdp line via "acn". - Adds codec setting initializers for these specific stereo cases. - Introduces new Stereo*Factory classes as optional convenience wrappers that inserts stereo codec to the existing set of supported codecs on demand. This CL is the step 5 for adding alpha channel support over the wire in webrtc. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: Ie12c56c8fcf7934e216135d73af33adec5248f76 Reviewed-on: https://webrtc-review.googlesource.com/22901 Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21210}
2017-12-11 12:21:02 +05:30
extern const char kCodecParamAssociatedCodecName[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kOpusCodecName[];
If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size. BUG=4289 TEST=Manual/Auto Test R=juberti@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44629004 Cr-Commit-Position: refs/heads/master@{#8863}
2015-03-26 07:39:19 +08:00
extern const char kIsacCodecName[];
extern const char kL16CodecName[];
extern const char kG722CodecName[];
extern const char kIlbcCodecName[];
extern const char kPcmuCodecName[];
extern const char kPcmaCodecName[];
extern const char kCnCodecName[];
extern const char kDtmfCodecName[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// Attribute parameters
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kCodecParamPTime[];
extern const char kCodecParamMaxPTime[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// fmtp parameters
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kCodecParamMinPTime[];
extern const char kCodecParamSPropStereo[];
extern const char kCodecParamStereo[];
extern const char kCodecParamUseInbandFec[];
Adding "usedtx" as Opus codec parameter. This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 Specifically, usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0. BUG=1014 R=juberti@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48499004 Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 05:05:59 +01:00
extern const char kCodecParamUseDtx[];
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kCodecParamMaxAverageBitrate[];
(Auto)update libjingle 75141932-> 75179475 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 07:57:12 +00:00
extern const char kCodecParamMaxPlaybackRate[];
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kParamValueTrue[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// Parameters are stored as parameter/value pairs. For parameters who do not
Use backticks not vertical bars to denote variables in comments for /media Bug: webrtc:12338 Change-Id: Ia800a4017ede1f647b36f809ef3c5b37a2616fdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226949 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34567}
2021-07-26 13:24:57 +02:00
// have a value, `kParamValueEmpty` should be used as value.
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kParamValueEmpty[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// opus parameters.
// Default value for maxptime according to
// http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
extern const int kOpusDefaultMaxPTime;
extern const int kOpusDefaultPTime;
extern const int kOpusDefaultMinPTime;
extern const int kOpusDefaultSPropStereo;
extern const int kOpusDefaultStereo;
extern const int kOpusDefaultUseInbandFec;
Adding "usedtx" as Opus codec parameter. This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 Specifically, usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0. BUG=1014 R=juberti@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48499004 Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 05:05:59 +01:00
extern const int kOpusDefaultUseDtx;
(Auto)update libjingle 75141932-> 75179475 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const int kOpusDefaultMaxPlaybackRate;
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// Prefered values in this code base. Note that they may differ from the default
// values in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
// Only frames larger or equal to 10 ms are currently supported in this code
// base.
extern const int kPreferredMaxPTime;
extern const int kPreferredMinPTime;
extern const int kPreferredSPropStereo;
extern const int kPreferredStereo;
extern const int kPreferredUseInbandFec;
Add support for enabling and negotiating raw RTP packetization. Raw RTP packetization is done using the existing RtpPacketizerGeneric without adding the generic payload header. It is intended to be used together with generic frame descriptor RTP header extension. Bug: webrtc:10625 Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061 Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28154}
2019-06-04 15:38:50 +02:00
extern const char kPacketizationParamRaw[];
Negotiate use of RTCP loss notification feedback (LNTF) When the LossNotifications field trial is in effect, LNTF should be offered/accepted in the SDP message, not assumed to be configured on both sides equally. Bug: webrtc:10662 Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28056}
2019-05-24 13:40:02 +02:00
// rtcp-fb message in its first experimental stages. Documentation pending.
extern const char kRtcpFbParamLntf[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// rtcp-fb messages according to RFC 4585
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
extern const char kRtcpFbParamNack[];
pRevert 5371 "Revert 5367 "Update talk to 59410372."" > Revert 5367 "Update talk to 59410372." > > > Update talk to 59410372. > > > > R=jiayl@webrtc.org, wu@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/6929004 > > TBR=mallinath@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/6999004 TBR=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
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extern const char kRtcpFbNackParamPli[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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// rtcp-fb messages according to
// http://tools.ietf.org/html/draft-alvestrand-rmcat-remb-00
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
extern const char kRtcpFbParamRemb[];
Require negotiation to send transport cc feedback over RTCP. BUG=4312 Review URL: https://codereview.webrtc.org/1452883002 Cr-Commit-Position: refs/heads/master@{#10735}
2015-11-20 18:05:48 -08:00
// rtcp-fb messages according to
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
extern const char kRtcpFbParamTransportCc[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
// ccm submessages according to RFC 5104
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
extern const char kRtcpFbParamCcm[];
extern const char kRtcpFbCcmParamFir[];
Add RRTR parameter to media engine and pass it to video receive stream This allows clients to enable Receiver reference time reports via PeerConnection. RRTR is not enabled by default but can be added to SDP string. Bug: webrtc:9108 Change-Id: I851f0d65152875bf115553a851b839f83e3d241e Reviewed-on: https://webrtc-review.googlesource.com/66861 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22767}
2018-04-04 16:33:49 +02:00
// Receiver reference time report
// https://tools.ietf.org/html/rfc3611 section 4.4
extern const char kRtcpFbParamRrtr[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
// Google specific parameters
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
extern const char kCodecParamMaxBitrate[];
extern const char kCodecParamMinBitrate[];
(Auto)update libjingle 66340694-> 66388864 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 11:15:20 +00:00
extern const char kCodecParamStartBitrate[];
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 22:36:21 +00:00
extern const char kCodecParamMaxQuantization[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
extern const char kComfortNoiseCodecName[];
Export symbols needed by the Chromium component build (part 8). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 11:33:59 +02:00
RTC_EXPORT extern const char kVp8CodecName[];
RTC_EXPORT extern const char kVp9CodecName[];
Add Av1 Decoder wrapper behind a build flag Bug: webrtc:11404 Change-Id: I090ffd173d667e8845de1b986af462516b7c76e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169452 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30757}
2020-03-11 10:45:57 +01:00
RTC_EXPORT extern const char kAv1CodecName[];
Export symbols needed by the Chromium component build (part 4). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: I12ef6f85ccef7dae3afe9ecff99725af13d551e2 Reviewed-on: https://webrtc-review.googlesource.com/c/106684 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25246}
2018-10-17 16:50:07 +02:00
RTC_EXPORT extern const char kH264CodecName[];
Support handling multiple RTX but only generate SDP with RTX associated with VP8. This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 20:24:50 +08:00
Generate FMTP parameters for the H.264 codec. This CL generates FMTP parameters that allow H.264 interoperation with Firefox for the default codec list. BUG=chromium:591971 Review URL: https://codereview.webrtc.org/1880963002 Cr-Commit-Position: refs/heads/master@{#12333}
2016-04-12 10:29:17 -07:00
// RFC 6184 RTP Payload Format for H.264 video
Export symbols needed by the Chromium component build (part 4). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: I12ef6f85ccef7dae3afe9ecff99725af13d551e2 Reviewed-on: https://webrtc-review.googlesource.com/c/106684 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25246}
2018-10-17 16:50:07 +02:00
RTC_EXPORT extern const char kH264FmtpProfileLevelId[];
RTC_EXPORT extern const char kH264FmtpLevelAsymmetryAllowed[];
RTC_EXPORT extern const char kH264FmtpPacketizationMode[];
Keep all codec parameters in VideoReceiveStream::Decoder It will be necessary to keep the H264 profile information in VideoReceiveStream::Decoder. I think it will be easier now and for the future to just store all of the codec parameters unmodified in VideoReceiveStream::Decoder instead of extracting a subset of them to an ad hoc class. BUG=webrtc:6743,webrtc:5948 Review-Url: https://codereview.webrtc.org/2523773003 Cr-Commit-Position: refs/heads/master@{#15239}
2016-11-25 03:56:37 -08:00
extern const char kH264FmtpSpropParameterSets[];
Add new fmtp parameter for H.264 Bug: webrtc:11769, webrtc:8423, webrtc:11376 Change-Id: Ia8f22ff90f817ba46ca03de1e43d3088c05023cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178904 Commit-Queue: Eldar Rello <elrello@microsoft.com> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31878}
2020-08-07 12:08:14 +03:00
extern const char kH264FmtpSpsPpsIdrInKeyframe[];
Generate FMTP parameters for the H.264 codec. This CL generates FMTP parameters that allow H.264 interoperation with Firefox for the default codec list. BUG=chromium:591971 Review URL: https://codereview.webrtc.org/1880963002 Cr-Commit-Position: refs/heads/master@{#12333}
2016-04-12 10:29:17 -07:00
extern const char kH264ProfileLevelConstrainedBaseline[];
Add constrained high profile level for h264 codec to media_constants Bug: None Change-Id: I7b21d21744c9e12e38fde884b409a5c88d0802a4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179369 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31738}
2020-07-15 14:29:06 +00:00
extern const char kH264ProfileLevelConstrainedHigh[];
Generate FMTP parameters for the H.264 codec. This CL generates FMTP parameters that allow H.264 interoperation with Firefox for the default codec list. BUG=chromium:591971 Review URL: https://codereview.webrtc.org/1880963002 Cr-Commit-Position: refs/heads/master@{#12333}
2016-04-12 10:29:17 -07:00
Support handling multiple RTX but only generate SDP with RTX associated with VP8. This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 20:24:50 +08:00
extern const int kDefaultVideoMaxFramerate;
Enable SVC based on number of SSRCs. Number of spatial layers is set equal to number of SSRCs. The maximum value is limited to 3. If spatial layering is enabled, i.e. number of spatial layers is greater than 1, then number of temporal layers is set to 3. Otherwise number of temporal layers is set to 1. Number of spatial and temporal layers can be overwritten through field trial. Bug: webrtc:8931 Change-Id: I37bd7fe053529683dc3e91b4e544fbdb44429340 Reviewed-on: https://webrtc-review.googlesource.com/59440 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22511}
2018-03-14 10:35:35 +01:00
extern const size_t kConferenceMaxNumSpatialLayers;
extern const size_t kConferenceMaxNumTemporalLayers;
extern const size_t kConferenceDefaultNumTemporalLayers;
Reland "sdp: parse and serialize b=TIAS" This reverts commit 20b701f3d79c499b0981f03fbf3a9b0fe531ac5d. Reason for reland: Reverting did not affect the test regression. Original change's description: > Revert "sdp: parse and serialize b=TIAS" > > This reverts commit c6801d4522ab94f965e258e68259fde312023654. > > Reason for revert: Speculatively reverting since it possibly breaks downstream performance test. > > One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case. > > Original change's description: > > sdp: parse and serialize b=TIAS > > > > BUG=webrtc:5788 > > > > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223 > > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> > > Reviewed-by: Taylor <deadbeef@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31729} > > TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:5788 > Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620 > Reviewed-by: Taylor <deadbeef@webrtc.org> > Commit-Queue: Taylor <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31762} TBR=nisse@webrtc.org Bug: webrtc:5788 Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31816}
2020-07-27 15:52:02 -07:00
extern const char kApplicationSpecificBandwidth[];
extern const char kTransportSpecificBandwidth[];
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
} // namespace cricket
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
#endif // MEDIA_BASE_MEDIA_CONSTANTS_H_
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