admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

197 lines
8.9 KiB
C
Raw Normal View History

Added API for getting the send-side estimated bandwidth. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/372006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1591 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-02 08:37:48 +00:00
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* 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.
*/
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
Refactoring the receive-side bandwidth estimation into its own module. Each REMB group has one remote_bitrate_estimator object. For now the estimator keeps one estimate for every SSRC. In a later commit this will be unified and one estimate will be used for all SSRC in one group. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/637009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 08:10:14 +00:00
lint build/include errors fixed in rtp_rtcp module BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1505993003 Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 05:05:27 -08:00
#include <set>
Add MID sending to RTPSender This CL adds the ability to configure RTPSender to include the MID header extension when sending packets. The MID will be included on every packet at the start of the stream until an RTCP acknoledgment is received for that SSRC at which point it will stop being included. The MID will be included on regular RTP streams as well as RTX streams. Bug: webrtc:4050 Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f Reviewed-on: https://webrtc-review.googlesource.com/60582 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22574}
2018-03-22 15:17:27 -07:00
#include <string>
lint build/include errors fixed in rtp_rtcp module BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1505993003 Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 05:05:27 -08:00
#include <utility>
#include <vector>
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
#include "absl/types/optional.h"
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
#include "api/video/video_bitrate_allocation.h"
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
#include "modules/include/module.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
namespace webrtc {
Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
class MockRtpData : public RtpData {
public:
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" This reverts commit 01aa210fad68f1006528d32d388b307c22990734. Reason for revert: downstream project adjusted Original change's description: > Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" > > This reverts commit 9486b117daac09c9f7ac8450ccda835938cf3150. > > Reason for revert: Breaks downstream project > > Original change's description: > > Enable and fix chromium clang warnings in rtp_rtcp test targets > > > > Bug: webrtc:163 > > Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c > > Reviewed-on: https://webrtc-review.googlesource.com/60802 > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22357} > > TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org > > Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:163 > Reviewed-on: https://webrtc-review.googlesource.com/61060 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22365} TBR=danilchap@webrtc.org,phoglund@webrtc.org,oprypin@webrtc.org,terelius@webrtc.org Change-Id: I0b4cb6d05b37caeb52cca9abf95417ad3ad6f76b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:163 Reviewed-on: https://webrtc-review.googlesource.com/61080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22368}
2018-03-09 15:37:03 +00:00
MockRtpData();
~MockRtpData();
Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
MOCK_METHOD3(OnReceivedPayloadData,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(const uint8_t* payload_data,
size_t payload_size,
const WebRtcRTPHeader* rtp_header));
Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
};
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
class MockRtpRtcp : public RtpRtcp {
public:
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" This reverts commit 01aa210fad68f1006528d32d388b307c22990734. Reason for revert: downstream project adjusted Original change's description: > Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" > > This reverts commit 9486b117daac09c9f7ac8450ccda835938cf3150. > > Reason for revert: Breaks downstream project > > Original change's description: > > Enable and fix chromium clang warnings in rtp_rtcp test targets > > > > Bug: webrtc:163 > > Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c > > Reviewed-on: https://webrtc-review.googlesource.com/60802 > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22357} > > TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org > > Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:163 > Reviewed-on: https://webrtc-review.googlesource.com/61060 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22365} TBR=danilchap@webrtc.org,phoglund@webrtc.org,oprypin@webrtc.org,terelius@webrtc.org Change-Id: I0b4cb6d05b37caeb52cca9abf95417ad3ad6f76b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:163 Reviewed-on: https://webrtc-review.googlesource.com/61080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22368}
2018-03-09 15:37:03 +00:00
MockRtpRtcp();
~MockRtpRtcp();
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
2013-05-29 12:12:51 +00:00
MOCK_METHOD2(IncomingRtcpPacket,
Drop return value from RtpRtcp::IncomingRtcpPacket. And from its callee RTCPReceiver::IncomingPacket. BUG=webrtc:5565 Review-Url: https://codereview.webrtc.org/3009403002 Cr-Commit-Position: refs/heads/master@{#19823}
2017-09-13 07:53:37 -07:00
void(const uint8_t* incoming_packet, size_t packet_length));
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
Make OverheadObserver::OnOverheadChanged count RTP headers only This lets the RTP code be unaware of lower layers, and the SetTransportOverhead method is deleted from RTPSender and RtpRtcp. Instead, that method is added to CongestionController and TransportFeedbackAdapter, where it is more appropriate. BUG=wertc:6847 Review-Url: https://codereview.webrtc.org/2589743002 Cr-Commit-Position: refs/heads/master@{#15995}
2017-01-10 08:58:32 -08:00
MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size));
MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec));
Move RTP module activation into PayloadRouter. Simplifies PayloadRouter to not accept dynamically-changing modules as well as usage of PayloadRouter inside ViEChannel::SetSendCodec. BUG=webrtc:5494 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1725363003 . Cr-Commit-Position: refs/heads/master@{#11787}
2016-02-26 16:31:37 +01:00
MOCK_METHOD2(RegisterVideoSendPayload,
void(int payload_type, const char* payload_name));
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD2(RegisterSendRtpHeaderExtension,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(RTPExtensionType type, uint8_t id));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(RTPExtensionType type));
Reland of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) Original issue's description: > Enable audio streams to send padding. > > Useful if bitrate probing is to be used with audio streams. > > BUG=webrtc:7043 > > Review-Url: https://codereview.webrtc.org/2652893004 > Cr-Commit-Position: refs/heads/master@{#16404} > Committed: https://chromium.googlesource.com/external/webrtc/+/e35f89a484ca376d5c187d166714eba578dfadc3 BUG=webrtc:7043 Review-Url: https://codereview.webrtc.org/2675703002 Cr-Commit-Position: refs/heads/master@{#16433}
2017-02-03 08:13:57 -08:00
MOCK_CONST_METHOD0(HasBweExtensions, bool());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_CONST_METHOD0(StartTimestamp, uint32_t());
MOCK_METHOD1(SetStartTimestamp, void(uint32_t timestamp));
Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream. The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport. BUG=webrtc:5687 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1864313003 . Cr-Commit-Position: refs/heads/master@{#12377}
2016-04-15 14:59:13 +02:00
MOCK_CONST_METHOD0(SequenceNumber, uint16_t());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetSequenceNumber, void(uint16_t seq));
Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream. The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport. BUG=webrtc:5687 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1864313003 . Cr-Commit-Position: refs/heads/master@{#12377}
2016-04-15 14:59:13 +02:00
MOCK_METHOD1(SetRtpState, void(const RtpState& rtp_state));
MOCK_METHOD1(SetRtxState, void(const RtpState& rtp_state));
MOCK_CONST_METHOD0(GetRtpState, RtpState());
MOCK_CONST_METHOD0(GetRtxState, RtpState());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_CONST_METHOD0(SSRC, uint32_t());
MOCK_METHOD1(SetSSRC, void(uint32_t ssrc));
Add MID sending to RTPSender This CL adds the ability to configure RTPSender to include the MID header extension when sending packets. The MID will be included on every packet at the start of the stream until an RTCP acknoledgment is received for that SSRC at which point it will stop being included. The MID will be included on regular RTP streams as well as RTX streams. Bug: webrtc:4050 Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f Reviewed-on: https://webrtc-review.googlesource.com/60582 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22574}
2018-03-22 15:17:27 -07:00
MOCK_METHOD1(SetMid, void(const std::string& mid));
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t csrcs[kRtpCsrcSize]));
Use vector of CSRCs for DeliverFrame & SetCSRCs. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28029004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 08:25:50 +00:00
MOCK_METHOD1(SetCsrcs, void(const std::vector<uint32_t>& csrcs));
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetCSRCStatus, int32_t(bool include));
Only return Rtx mode in RTXSendStatus(). There is no need to return 'ssrc' and 'payloadtype' inside this function since they are never used. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38569004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 14:15:15 +00:00
MOCK_METHOD1(SetRtxSendStatus, void(int modes));
MOCK_CONST_METHOD0(RtxSendStatus, int());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetRtxSsrc, void(uint32_t));
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
MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
MOCK_CONST_METHOD0(FlexfecSsrc, absl::optional<uint32_t>());
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
MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair<int, int>());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetSendingStatus, int32_t(bool sending));
MOCK_CONST_METHOD0(Sending, bool());
MOCK_METHOD1(SetSendingMediaStatus, void(bool sending));
MOCK_CONST_METHOD0(SendingMedia, bool());
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_CONST_METHOD4(BitrateSent,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nack_rate));
Add APIs for getting receive-side estimated bandwidth and codec target rate. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/391012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1704 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 14:45:37 +00:00
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int(uint32_t* available_bandwidth));
Add EncodedImageCallback::OnEncodedImage(). OnEncodedImage() is going to replace Encoded(), which is deprecated now. The new OnEncodedImage() returns Result struct that contains frame_id, which tells the encoder RTP timestamp for the frame. BUG=chromium:621691 R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2089773002 . Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25 Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795 Cr-Original-Original-Commit-Position: refs/heads/master@{#13613} Cr-Original-Commit-Position: refs/heads/master@{#13615} Cr-Commit-Position: refs/heads/master@{#13617}
2016-08-02 17:46:41 -07:00
MOCK_METHOD9(SendOutgoingData,
bool(FrameType frame_type,
int8_t payload_type,
uint32_t timestamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_header,
uint32_t* frame_id_out));
Revert "Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )" This reverts commit 46948c17fd09e4957bebc8ea61f0a8e77ff84b48. TBR=mflodman@webrtc.org BUG=webrtc:5859 Review-Url: https://codereview.webrtc.org/2032473002 Cr-Commit-Position: refs/heads/master@{#12992}
2016-06-01 06:31:17 -07:00
MOCK_METHOD5(TimeToSendPacket,
bool(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
Propagate packet pacing information to SenTimeHistory. In order to not make this CL too large I have broken it down into at least two steps. In this CL we only propagate the pacing information part of the way: webrtc::PacedSender::Process <--- propagate from here webrtc::PacedSender::SendPacket webrtc::PacketRouter::TimeToSendPacket webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- to here webrtc::RTPSender::TimeToSendPacket webrtc::RTPSender::PrepareAndSendPacket webrtc::RTPSender::AddPacketToTransportFeedback webrtc::TransportFeedbackAdapter::AddPacket webrtc::SendTimeHistory::AddAndRemoveOld <--- goal is to propagte it here BUG=webrtc:6822 Review-Url: https://codereview.webrtc.org/2628563003 Cr-Commit-Position: refs/heads/master@{#16664}
2017-02-17 03:59:43 -08:00
const PacedPacketInfo& pacing_info));
MOCK_METHOD2(TimeToSendPadding,
size_t(size_t bytes, const PacedPacketInfo& pacing_info));
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
2014-11-14 11:52:04 +00:00
MOCK_METHOD2(RegisterRtcpObservers,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback));
Unify newapi::RtcpMode and RTCPMethod. BUG=webrtc:1695 R=solenberg@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1373903003 Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 02:36:56 -07:00
MOCK_CONST_METHOD0(RTCP, RtcpMode());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method));
MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE]));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_CONST_METHOD2(RemoteCNAME,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint32_t remote_ssrc,
char cname[RTCP_CNAME_SIZE]));
Change audio/video sync to be based on mapping RTP timestamps to NTP. Video Engine: - Instead compensate for video capture delay by modifying RTP timestamps. - Calculate the relative offset between audio and video by converting RTP timestamps to NTP and comparing receive time. RTP/RTCP module: - Removes the awkward video modification of NTP to compensate for video capture delay. - Adjust RTCP RTP timestamp generation in rtcp_sender to have the same offset as packets being sent from rtp_sender. BUG= TEST=trybots,steam_synchronization_unittest Review URL: https://webrtc-codereview.appspot.com/669010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2733 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-11 07:00:42 +00:00
MOCK_CONST_METHOD5(RemoteNTP,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint32_t* received_ntp_secs,
uint32_t* received_ntp_frac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD2(AddMixedCNAME,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint32_t ssrc, const char cname[RTCP_CNAME_SIZE]));
MOCK_METHOD1(RemoveMixedCNAME, int32_t(uint32_t ssrc));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_CONST_METHOD5(RTT,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint32_t remote_ssrc,
int64_t* rtt,
int64_t* avg_rtt,
int64_t* min_rtt,
int64_t* max_rtt));
MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packet_type));
Refactor RTCP sender The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but it has quite a few ramifications. Notable changes: * Removed the rtcpPacketTypeFlags bit vector and don't assume RTCPPacketType values have a single unique bit set. This will allow making this an enum class once rtcp_receiver has been overhauled. * Flags are now stored in a map that is a member of the class. This meant we could remove some bool flags (eg send_remb_) which was previously masked into rtcpPacketTypeFlags and then masked out again when testing if a remb packet should be sent. * Make all build methods, eg. BuildREMB(), have the same signature. An RtcpContext struct was introduced for this purpose. This allowed the use of a map from RTCPPacketType to method pointer. Instead of 18 consecutive if-statements, there is now a single loop. The context class also allowed some simplifications in the build methods themselves. * A few minor simplifications and cleanups. The next step is to gradually replace the builder methods with the builders from the new RtcpPacket classes. BUG=2450 R=asapersson@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48329004 Cr-Commit-Position: refs/heads/master@{#9166}
2015-05-11 10:17:43 +02:00
MOCK_METHOD1(SendCompoundRTCP,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(const std::set<RTCPPacketType>& packet_types));
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
2013-08-15 23:38:54 +00:00
MOCK_CONST_METHOD2(DataCountersRTP,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(size_t* bytes_sent, uint32_t* packets_sent));
Add video send bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateSentInKbps") - media bitrate ("WebRTC.Video.MediaBitrateSentInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps") - retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps") Add retransmitted bytes to StreamDataCounters. Change in UpdateRtpStats to also update counters for retransmitted packet. BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 09:47:53 +00:00
MOCK_CONST_METHOD2(GetSendStreamDataCounters,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(StreamDataCounters*, StreamDataCounters*));
Add statistics gathering for packet loss. Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics. BUG= Review URL: https://codereview.webrtc.org/1198853004 Cr-Commit-Position: refs/heads/master@{#9568}
2015-07-10 18:10:05 -07:00
MOCK_CONST_METHOD3(GetRtpPacketLossStats,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(bool, uint32_t, struct RtpPacketLossStats*));
Fix defect http://code.google.com/p/webrtc/issues/detail?id=222 "ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets. BUG=222 TEST= tested in loopback. No new test added yet. Review URL: http://webrtc-codereview.appspot.com/343003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 13:00:08 +00:00
MOCK_CONST_METHOD1(RemoteRTCPStat,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(std::vector<RTCPReportBlock>* receive_blocks));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD4(SetRTCPApplicationSpecificData,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint8_t sub_type,
uint32_t name,
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1513303003 Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 02:54:47 -08:00
const uint8_t* data,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
uint16_t length));
MOCK_METHOD1(SetRTCPVoIPMetrics, int32_t(const RTCPVoIPMetric* voip_metric));
MOCK_METHOD1(SetRtcpXrRrtrStatus, void(bool enable));
MOCK_CONST_METHOD0(RtcpXrRrtrStatus, bool());
Change RtpRtcp::SetRemb signature to match RtcpTransceiver::SetRemb in particular change bitrate type to int64_t to follow style guide. With an extra interface it will allow to add both RtpRtcp module and RtcpTransceiver as feedback sender to PacketRouter Bug: webrtc:8239 Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba Reviewed-on: https://webrtc-review.googlesource.com/32302 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21250}
2017-12-13 13:35:10 +01:00
MOCK_METHOD2(SetRemb, void(int64_t bitrate, std::vector<uint32_t> ssrcs));
Simplify RtpRtcp interface for REMB Remove REMB accessor as used for debug checks only. Merge SetRembData and SetRembStatus(true) eliminating state 'remb can be send, but no data available yet' Bug: None Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf Reviewed-on: https://webrtc-review.googlesource.com/7983 Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20240}
2017-10-10 17:46:26 +02:00
MOCK_METHOD0(UnsetRemb, void());
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_CONST_METHOD0(TMMBR, bool());
MOCK_METHOD1(SetTMMBRStatus, void(bool enable));
MOCK_METHOD1(OnBandwidthEstimateUpdate, void(uint16_t bandwidth_kbit));
MOCK_CONST_METHOD0(SelectiveRetransmissions, int());
MOCK_METHOD1(SetSelectiveRetransmissions, int(uint8_t settings));
MOCK_METHOD2(SendNACK, int32_t(const uint16_t* nack_list, uint16_t size));
Experiment for the nack module. Testing the nack module by implementing it into the current jitter buffer under the experiment WebRTC-NewVideoJitterBuffer. BUG=webrtc:5514 Review URL: https://codereview.webrtc.org/1778503002 Cr-Commit-Position: refs/heads/master@{#11969}
2016-03-12 03:30:23 -08:00
MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD2(SetStorePacketsStatus,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(bool enable, uint16_t number_to_store));
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
2013-08-15 23:38:54 +00:00
MOCK_CONST_METHOD0(StorePackets, bool());
Refactor some receive-side stats. Removes polling of CName as well as receive codec statistics in favor of internal callbacks keeping a statistics struct up to date. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/28259005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:50:16 +00:00
MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
Add RemoteEstimatorProxy for capturing receive times For use when send-side bandwidth estimation is enabled. Receive times need to be captured, buffered and then sent using TransportFeedback RTCP messaged back to the send side. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1290813008 Cr-Commit-Position: refs/heads/master@{#9898}
2015-09-08 13:25:16 -07:00
MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
MOCK_METHOD3(SendTelephoneEventOutband,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
int32_t(uint8_t key, uint16_t time_ms, uint8_t level));
MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov));
MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps));
Simplify {,Set}UlpfecConfig interface. Prior to this change, we signalled that ULPFEC was disabled through a bool, but that RED was disabled by setting its payload type to -1. The latter is consistent with how we disable RED/ULPFEC in the config, so this CL removes the ULPFEC bool from the {,Set}UlpfecConfig chain of member functions. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2460533002 Cr-Commit-Position: refs/heads/master@{#14944}
2016-11-07 03:05:06 -08:00
MOCK_METHOD2(SetUlpfecConfig,
void(int red_payload_type, int fec_payload_type));
Adding classes for handling multi-frame FEC. The FEC behavior is unchanged with this commit, we will still be limited to FEC over one frame for now. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/450006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1915 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-20 22:10:56 +00:00
MOCK_METHOD2(SetFecParameters,
Simplify SetFecParameters signature. - Change const ptr to const ref in parameter list. Using nullptr as argument was invalid, so no need to send pointer instead of reference. - Change return type to void or bool, where appropriate BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2455963003 Cr-Commit-Position: refs/heads/master@{#14945}
2016-11-07 03:36:05 -08:00
bool(const FecProtectionParams& delta_params,
const FecProtectionParams& key_params));
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method));
MOCK_METHOD0(RequestKeyFrame, int32_t());
MOCK_METHOD0(Process, void());
Add callbacks for send channel rtp statistics BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:29:02 +00:00
MOCK_METHOD1(RegisterSendChannelRtpStatisticsCallback,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
void(StreamDataCountersCallback*));
Add callbacks for send channel rtp statistics BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:29:02 +00:00
MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
Style cleanups in RtpSender. - Renamed variables and some function to comply with style guide. - Removed default argument values. - Removed some dead code. - Cleaned up comments formatting in rtp_rtcp.h R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2067673004 . Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 15:19:10 -07:00
StreamDataCountersCallback*(void));
Move BitrateAllocation to api/ and rename it VideoBitrateAllocation Since the webrtc_common build target does not have visibility set, we cannot easily use BitrateAllocation in other parts of Chromium. This is currently blocking parts of chromium:794608, and I know of other usage outside webrtc already, so moving it to api/ should be warranted. Also, since there's some naming confusion and this class is video specific rename it VideoBitrateAllocation. This also fits with the standard interface for producing these: VideoBitrateAllocator. Bug: chromium:794608 Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe Reviewed-on: https://webrtc-review.googlesource.com/70783 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 12:32:22 +02:00
MOCK_METHOD1(SetVideoBitrateAllocation, void(const VideoBitrateAllocation&));
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
// Members.
unsigned int remote_ssrc_;
Provide default impl of TimeUntilNextProcess in MockRtpRtcp instead of mocking. This method isn't currently mocked or required by any test, so the safe thing is to return a reasonably large value from the implementation to avoid busy loops. BUG=webrtc:7187 TBR=mflodman@webrtc.org Review-Url: https://codereview.webrtc.org/2744233002 Cr-Commit-Position: refs/heads/master@{#17284}
2017-03-17 04:01:30 -07:00
private:
// Mocking this method is currently not required and having a default
// implementation like MOCK_METHOD0(TimeUntilNextProcess, int64_t())
// can be dangerous since it can cause a tight loop on a process thread.
virtual int64_t TimeUntilNextProcess() { return 0xffffffff; }
Add REMB functionality to ViE. This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
};
} // namespace webrtc
Refactoring the receive-side bandwidth estimation into its own module. Each REMB group has one remote_bitrate_estimator object. For now the estimator keeps one estimate for every SSRC. In a later commit this will be unified and one estimate will be used for all SSRC in one group. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/637009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2375 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 08:10:14 +00:00
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
#endif // MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
Reference in New Issue Copy Permalink