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/webrtc@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/engine_configurations.h b/engine_configurations.h
index 67effa0..f3d2a46 100644
--- a/engine_configurations.h
+++ b/engine_configurations.h
@@ -109,7 +109,6 @@
#define WEBRTC_VIDEO_ENGINE_CODEC_API
#define WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-#define WEBRTC_VIDEO_ENGINE_NETWORK_API
#define WEBRTC_VIDEO_ENGINE_RENDER_API
#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
#define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
diff --git a/modules/modules.gyp b/modules/modules.gyp
index 837d551..1671c13 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -27,7 +27,6 @@
'pacing/pacing.gypi',
'remote_bitrate_estimator/remote_bitrate_estimator.gypi',
'rtp_rtcp/source/rtp_rtcp.gypi',
- 'udp_transport/source/udp_transport.gypi',
'utility/source/utility.gypi',
'video_coding/codecs/i420/main/source/i420.gypi',
'video_coding/main/source/video_coding.gypi',
diff --git a/modules/udp_transport/OWNERS b/modules/udp_transport/OWNERS
deleted file mode 100644
index 3b2a444..0000000
--- a/modules/udp_transport/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-pwestin@webrtc.org
-henrikg@webrtc.org
-mallinath@webrtc.org
-tomasl@webrtc.org
\ No newline at end of file
diff --git a/modules/udp_transport/source/udp_transport.gypi b/modules/udp_transport/source/udp_transport.gypi
deleted file mode 100644
index a73a587..0000000
--- a/modules/udp_transport/source/udp_transport.gypi
+++ /dev/null
@@ -1,114 +0,0 @@
-# Copyright (c) 2011 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.
-
-{
- 'targets': [
- {
- 'target_name': 'udp_transport',
- 'type': 'static_library',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- },
- 'sources': [
- # PLATFORM INDEPENDENT SOURCE FILES
- '../interface/udp_transport.h',
- 'udp_transport_impl.cc',
- 'udp_socket_wrapper.cc',
- 'udp_socket_manager_wrapper.cc',
- 'udp_transport_impl.h',
- 'udp_socket_wrapper.h',
- 'udp_socket_manager_wrapper.h',
- # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
- # Posix (Linux/Mac)
- 'udp_socket_posix.cc',
- 'udp_socket_posix.h',
- 'udp_socket_manager_posix.cc',
- 'udp_socket_manager_posix.h',
- # Windows
- 'udp_socket2_manager_windows.cc',
- 'udp_socket2_manager_windows.h',
- 'udp_socket2_windows.cc',
- 'udp_socket2_windows.h',
- 'traffic_control_windows.cc',
- 'traffic_control_windows.h',
- ], # source
- 'conditions': [
- # DEFINE PLATFORM SPECIFIC SOURCE FILES
- ['os_posix==0', {
- 'sources!': [
- 'udp_socket_posix.cc',
- 'udp_socket_posix.h',
- 'udp_socket_manager_posix.cc',
- 'udp_socket_manager_posix.h',
- ],
- }],
- ['OS!="win"', {
- 'sources!': [
- 'udp_socket2_manager_windows.cc',
- 'udp_socket2_manager_windows.h',
- 'udp_socket2_windows.cc',
- 'udp_socket2_windows.h',
- 'traffic_control_windows.cc',
- 'traffic_control_windows.h',
- ],
- }],
- ['OS=="linux"', {
- 'cflags': [
- '-fno-strict-aliasing',
- ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_CPLUSPLUSFLAGS': [ '-fno-strict-aliasing' ],
- },
- }],
- ] # conditions
- },
- ], # targets
- 'conditions': [
- ['include_tests==1', {
- 'targets': [
- {
- 'target_name': 'udp_transport_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'udp_transport',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- '<(DEPTH)/testing/gmock.gyp:gmock',
- '<(webrtc_root)/test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'udp_transport_unittest.cc',
- 'udp_socket_manager_unittest.cc',
- 'udp_socket_wrapper_unittest.cc',
- ],
- # Disable warnings to enable Win64 build, issue 1323.
- 'msvs_disabled_warnings': [
- 4267, # size_t to int truncation.
- ],
- }, # udp_transport_unittests
- ], # targets
- }], # include_tests
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/modules/udp_transport/test/SocketManagerTest.cc b/modules/udp_transport/test/SocketManagerTest.cc
deleted file mode 100644
index 03119be..0000000
--- a/modules/udp_transport/test/SocketManagerTest.cc
+++ /dev/null
@@ -1,449 +0,0 @@
-/*
- * 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.
- */
-
-#include <cassert>
-#include <iostream>
-
-#ifdef _WIN32
-#include <windows.h>
-#include <tchar.h>
-#else
-#include <stdio.h>
-#define Sleep(x) usleep(x*1000)
-#endif
-
-#include "udp_transport.h"
-#include "common_types.h"
-#include "trace.h"
-
-//#define QOS_TEST
-//#define QOS_TEST_WITH_OVERRIDE // require admin on Win7
-//#define TOS_TEST // require admin on Win7
-//#define TOS_TEST_USING_SETSOCKOPT
-//#define PCP_TEST
-
-class UdpTransportDataA: public UdpTransportData
-{
-public:
- UdpTransportDataA() :
- _counterRTP(0),
- _counterRTCP(0)
- {
- };
- virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTP++;
- };
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTCP++;
- };
- WebRtc_UWord32 _counterRTP;
- WebRtc_UWord32 _counterRTCP;
-};
-
-class UdpTransportDataB: public UdpTransportData
-{
-public:
- UdpTransportDataB() :
- _counterRTP(0),
- _counterRTCP(0)
- {
- };
- virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTP++;
- };
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTCP++;
- };
- WebRtc_UWord32 _counterRTP;
- WebRtc_UWord32 _counterRTCP;
-};
-
-#ifdef _WIN32
-int _tmain(int argc, _TCHAR* argv[])
-#else
-int main(int argc, char* argv[])
-#endif
-{
- Trace::CreateTrace();
- Trace::SetTraceFile("testTrace.txt");
- Trace::SetEncryptedTraceFile("testTraceDebug.txt");
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- printf("Start UdpTransport test\n");
-
- WebRtc_UWord8 numberOfSocketThreads = 5;
- UdpTransport* client1 = UdpTransport::Create(1,numberOfSocketThreads,NULL);
- numberOfSocketThreads = 0;
- UdpTransport* client2 = UdpTransport::Create(2,numberOfSocketThreads,NULL);
- assert(5 == numberOfSocketThreads);
-
- UdpTransportDataA* client1Callback = new UdpTransportDataA();
- UdpTransportDataB* client2Callback = new UdpTransportDataB();
-
- WebRtc_UWord32 localIP = 0;
- char localIPAddr[64];
- assert( 0 == client1->LocalHostAddress(localIP)); // network host order aka big-endian
-
- sprintf(localIPAddr,"%lu.%lu.%lu.%lu",(localIP>>24)& 0x0ff,(localIP>>16)& 0x0ff ,(localIP>>8)& 0x0ff, localIP & 0x0ff);
- printf("\tLocal IP:%s\n", localIPAddr);
-
- char localIPV6[16];
- char localIPAddrV6[128];
- if( 0 == client1->LocalHostAddressIPV6(localIPV6))
- {
- sprintf(localIPAddrV6,"%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x", localIPV6[0],localIPV6[1],localIPV6[2],localIPV6[3],localIPV6[4],localIPV6[5],localIPV6[6],localIPV6[7], localIPV6[8],localIPV6[9],localIPV6[10],localIPV6[11],localIPV6[12],localIPV6[13],localIPV6[14],localIPV6[15]);
- printf("\tLocal IPV6:%s\n", localIPAddrV6);
- }
-
- char test[9] = "testtest";
- assert( 0 == client1->InitializeReceiveSockets(client1Callback,1234, localIPAddr));
-
-#if defined QOS_TEST_WITH_OVERRIDE || defined QOS_TEST || defined TOS_TEST || defined TOS_TEST_USING_SETSOCKOPT
- assert( -1 == client1->SetQoS(true, 3, 1000)); // should fail
- assert( 0 == client1->InitializeSendSockets("192.168.200.1", 1236,1237));
-#else
- assert( 0 == client1->InitializeSendSockets(localIPAddr, 1236,1237));
-#endif
- assert( 0 == client1->StartReceiving(20));
-
- assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
- assert( 0 == client2->InitializeSendSockets(localIPAddr, 1234,1235));
- assert( 0 == client2->StartReceiving(20));
-
- Sleep(10);
-
-#ifdef TOS_TEST
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetToS(2));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(3));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(0));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested TOS \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef TOS_TEST_USING_SETSOCKOPT
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetToS(2, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(3, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(0, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested TOS using setsockopt \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef QOS_TEST
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetQoS(true, 2, 1000)); // SERVICETYPE_CONTROLLEDLOAD 2
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(true, 3, 1000)); // SERVICETYPE_GUARANTEED 3
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(false, 0)); //
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested QOS \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef QOS_TEST_WITH_OVERRIDE
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetQoS(true, 2, 1000, 1)); // SERVICETYPE_CONTROLLEDLOAD 2
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(true, 2, 1000, 2)); // SERVICETYPE_GUARANTEED 3
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(false, 0)); //
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- printf("Tested QOS with override \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef PCP_TEST
- // Note: We currently don't know how to test that the bits are actually set in the frame,
- // this test simply tests the API and that we can send a packet after setting PCP.
- assert( -1 == client1->SetPCP(-1)); // should fail
- assert( -1 == client1->SetPCP(8)); // should fail
- printf("Setting PCP to 7 returned %d \n", client1->SetPCP(7));
- printf("(Failing is normal, requires the CAP_NET_ADMIN capability to succeed.) \n");
- Sleep(10);
- for (int pcp = 6; pcp >= 0; --pcp)
- {
- assert( 0 == client1->SetPCP(pcp));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
- printf("Tested PCP \n");
- Sleep(5000);
- return 0;
-#endif
-
- Sleep(10);
-
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- // test start rec after a socket has revceived data
- // result: packets received before first startReceive is saved by the OS
-/*
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
- Sleep(10);
- assert( 0 == client2->StartReceiving(20));
-
-// assert( 0 == client2->StopReceiving());
-
- Sleep(10);
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
-
- assert( 0 == client2->StartReceiving(20));
-
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
-*/
- Sleep(10);
-
- assert( 0 == client1Callback->_counterRTP);
- assert( 1 == client2Callback->_counterRTP);
- assert( 0 == client1Callback->_counterRTCP);
- assert( 0 == client2Callback->_counterRTCP);
-
- printf("Sent 1 packet on one socket \n");
-
- char ipAddr[64];
- char tempIpAddr[64];
- char ipMulticastAddr[64];
- WebRtc_UWord16 rtpPort = 0;
- WebRtc_UWord16 rtcpPort = 0;
- bool reusableSocket = true;
- assert( 0 == client2->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1234);
- assert( strncmp(ipAddr, localIPAddr, 16) == 0);
-
- assert( 0 == client2->ReceiveSocketInformation(ipAddr, rtpPort, rtcpPort, ipMulticastAddr, reusableSocket));
- assert( rtpPort == 1236);
- assert( rtcpPort == 1237);
- assert( strncmp(ipAddr, "0.0.0.0", 16) == 0);
- assert( ipMulticastAddr[0] == 0);
- assert( reusableSocket == false);
-
- assert( 0 == client2->SendSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1234);
- assert( rtcpPort == 1235);
- assert( strncmp(ipAddr,localIPAddr, 16) == 0);
-
- const int numberOfPackets = 1000;
- int n = 0;
- while(n < numberOfPackets)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client1->SendRTCPPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- n++;
- }
- int loops = 0;
- for(; loops < 100 &&
- !(client1Callback->_counterRTP == numberOfPackets &&
- client1Callback->_counterRTCP == numberOfPackets &&
- client2Callback->_counterRTP == numberOfPackets+1 &&
- client2Callback->_counterRTCP == numberOfPackets);
- loops++)
- {
- Sleep(10);
- }
- printf("\tSent %d packets on 4 sockets in:%d ms\n", numberOfPackets, loops*10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+1 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->StopReceiving());
- assert( 0 == client2->StopReceiving());
-
- printf("Tear down client 2\n");
-
- // configure that fail
- assert( -1 == client2->InitializeReceiveSockets(client2Callback,1234, localIPAddr)); // port in use
- assert( !client2->ReceiveSocketsInitialized());
- assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
- assert( 0 == client2->StartReceiving(20));
-
- printf("Client 2 re-configured\n");
-
- assert( client1->SendSocketsInitialized());
- assert( client1->ReceiveSocketsInitialized());
- assert( client2->SendSocketsInitialized());
- assert( client2->ReceiveSocketsInitialized());
-
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- // this should not be received since we dont receive in client 1
- assert( 9 == client2->SendPacket(-1, test, 9));
-
- Sleep(10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
- printf("\tSent 1 packet on one socket \n");
-
- printf("Start filter test\n");
-
- assert( 0 == client1->StartReceiving(20));
-
- assert( 0 == client1->SetFilterPorts(1234, 1235)); // should filter out what we send
- assert( 0 == client1->SetFilterIP(localIPAddr));
-
- assert( 0 == client1->FilterIP(tempIpAddr));
- assert( strncmp(tempIpAddr, localIPAddr, 16) == 0);
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
-
- Sleep(10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterPorts(1236, 1237)); // should pass through
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+1 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+1 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterIP("127.0.0.2"));
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+1 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+1 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterIP(NULL));
- assert( 0 == client1->SetFilterPorts(0, 0));
-
- printf("Tested filter \n");
-
- assert( 0 == client2->InitializeSourcePorts(1238, 1239));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+2 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+2 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1238);
- assert( rtcpPort == 1239);
- assert( strncmp(ipAddr, localIPAddr, 16) == 0);
-
- printf("Tested source port \n");
-
- assert( 0 == client2->InitializeSourcePorts(1240 ));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1240);
- assert( rtcpPort == 1241);
-
- printf("Tested SetSendPorts source port \n");
-
- UdpTransport::Destroy(client1);
- UdpTransport::Destroy(client2);
-
- printf("\n\nUdpTransport test done\n");
-
- delete client1Callback;
- delete client2Callback;
-
- Sleep(5000);
- Trace::ReturnTrace();
-};
diff --git a/test/udp_transport.gyp b/test/udp_transport.gyp
new file mode 100644
index 0000000..37c9e73
--- /dev/null
+++ b/test/udp_transport.gyp
@@ -0,0 +1,71 @@
+# Copyright (c) 2011 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.
+
+{
+ 'includes': [
+ '../build/common.gypi',
+ ],
+ 'targets': [
+ {
+ 'target_name': 'udp_transport',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'udp_transport/include',
+ ],
+ },
+ 'sources': [
+ # PLATFORM INDEPENDENT SOURCE FILES
+ 'udp_transport/channel_transport.cc',
+ 'udp_transport/include/channel_transport.h',
+ 'udp_transport/udp_transport.h',
+ 'udp_transport/udp_transport_impl.cc',
+ 'udp_transport/udp_socket_wrapper.cc',
+ 'udp_transport/udp_socket_manager_wrapper.cc',
+ 'udp_transport/udp_transport_impl.h',
+ 'udp_transport/udp_socket_wrapper.h',
+ 'udp_transport/udp_socket_manager_wrapper.h',
+ # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
+ # Posix (Linux/Mac)
+ 'udp_transport/udp_socket_posix.cc',
+ 'udp_transport/udp_socket_posix.h',
+ 'udp_transport/udp_socket_manager_posix.cc',
+ 'udp_transport/udp_socket_manager_posix.h',
+ # win
+ 'udp_transport/udp_socket2_manager_win.cc',
+ 'udp_transport/udp_socket2_manager_win.h',
+ 'udp_transport/udp_socket2_win.cc',
+ 'udp_transport/udp_socket2_win.h',
+ 'udp_transport/traffic_control_win.cc',
+ 'udp_transport/traffic_control_win.h',
+ ], # source
+ },
+ {
+ 'target_name': 'udp_transport_unittests',
+ 'type': 'executable',
+ 'dependencies': [
+ 'udp_transport',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(DEPTH)/testing/gmock.gyp:gmock',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'udp_transport/udp_transport_unittest.cc',
+ 'udp_transport/udp_socket_manager_unittest.cc',
+ 'udp_transport/udp_socket_wrapper_unittest.cc',
+ ],
+ # Disable warnings to enable Win64 build, issue 1323.
+ 'msvs_disabled_warnings': [
+ 4267, # size_t to int truncation.
+ ],
+ }, # udp_transport_unittests
+ ], # targets
+}
diff --git a/modules/udp_transport/source/Android.mk b/test/udp_transport/Android.mk
similarity index 100%
rename from modules/udp_transport/source/Android.mk
rename to test/udp_transport/Android.mk
diff --git a/test/udp_transport/channel_transport.cc b/test/udp_transport/channel_transport.cc
new file mode 100644
index 0000000..8be8938
--- /dev/null
+++ b/test/udp_transport/channel_transport.cc
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2013 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.
+ */
+
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+
+#include "webrtc/test/udp_transport/udp_transport.h"
+#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+
+namespace webrtc {
+
+VoiceChannelTransport::VoiceChannelTransport(VoENetwork* voe_network,
+ int channel)
+ : channel_(channel),
+ voe_network_(voe_network) {
+ WebRtc_UWord8 socket_threads = 1;
+ socket_transport_ = UdpTransport::Create(channel, socket_threads);
+ voe_network_->RegisterExternalTransport(channel, *socket_transport_);
+}
+
+VoiceChannelTransport::~VoiceChannelTransport() {
+ voe_network_->DeRegisterExternalTransport(channel_);
+ UdpTransport::Destroy(socket_transport_);
+}
+
+void VoiceChannelTransport::IncomingRTPPacket(
+ const WebRtc_Word8* incoming_rtp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/) {
+ voe_network_->ReceivedRTPPacket(channel_, incoming_rtp_packet, packet_length);
+}
+
+void VoiceChannelTransport::IncomingRTCPPacket(
+ const WebRtc_Word8* incoming_rtcp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/) {
+ voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet,
+ packet_length);
+}
+
+int VoiceChannelTransport::SetLocalReceiver(WebRtc_UWord16 rtp_port) {
+ return socket_transport_->InitializeReceiveSockets(this, rtp_port);
+}
+
+int VoiceChannelTransport::SetSendDestination(const char* ip_address,
+ WebRtc_UWord16 rtp_port) {
+ return socket_transport_->InitializeSendSockets(ip_address, rtp_port);
+}
+
+
+VideoChannelTransport::VideoChannelTransport(ViENetwork* vie_network,
+ int channel)
+ : channel_(channel),
+ vie_network_(vie_network) {
+ WebRtc_UWord8 socket_threads = 1;
+ socket_transport_ = UdpTransport::Create(channel, socket_threads);
+ vie_network_->RegisterSendTransport(channel, *socket_transport_);
+}
+
+VideoChannelTransport::~VideoChannelTransport() {
+ vie_network_->DeregisterSendTransport(channel_);
+ UdpTransport::Destroy(socket_transport_);
+}
+
+void VideoChannelTransport::IncomingRTPPacket(
+ const WebRtc_Word8* incoming_rtp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/) {
+ vie_network_->ReceivedRTPPacket(channel_, incoming_rtp_packet, packet_length);
+}
+
+void VideoChannelTransport::IncomingRTCPPacket(
+ const WebRtc_Word8* incoming_rtcp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/) {
+ vie_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet,
+ packet_length);
+}
+
+int VideoChannelTransport::SetLocalReceiver(WebRtc_UWord16 rtp_port) {
+ return socket_transport_->InitializeReceiveSockets(this, rtp_port);
+}
+
+int VideoChannelTransport::SetSendDestination(const char* ip_address,
+ WebRtc_UWord16 rtp_port) {
+ return socket_transport_->InitializeSendSockets(ip_address, rtp_port);
+}
+
+} // namespace webrtc
diff --git a/test/udp_transport/include/channel_transport.h b/test/udp_transport/include/channel_transport.h
new file mode 100644
index 0000000..e3291ef
--- /dev/null
+++ b/test/udp_transport/include/channel_transport.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2013 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.
+ */
+
+#ifndef WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_CHANNEL_TRANSPORT_H_
+#define WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_CHANNEL_TRANSPORT_H_
+
+#include "webrtc/test/udp_transport/udp_transport.h"
+
+namespace webrtc {
+
+class ViENetwork;
+class VoENetwork;
+
+// Helper class for VoiceEngine tests.
+class VoiceChannelTransport : public UdpTransportData {
+ public:
+ VoiceChannelTransport(VoENetwork* voe_network, int channel);
+
+ virtual ~VoiceChannelTransport();
+
+ // Start implementation of UdpTransportData.
+ void IncomingRTPPacket(const WebRtc_Word8* incoming_rtp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/);
+
+ void IncomingRTCPPacket(const WebRtc_Word8* incoming_rtcp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/);
+ // End implementation of UdpTransportData.
+
+ // Specifies the ports to receive RTP packets on.
+ int SetLocalReceiver(WebRtc_UWord16 rtp_port);
+
+ // Specifies the destination port and IP address for a specified channel.
+ int SetSendDestination(const char* ip_address, WebRtc_UWord16 rtp_port);
+
+ private:
+ int channel_;
+ VoENetwork* voe_network_;
+ UdpTransport* socket_transport_;
+};
+
+// Helper class for VideoEngine tests.
+class VideoChannelTransport : public UdpTransportData {
+ public:
+ VideoChannelTransport(ViENetwork* vie_network, int channel);
+
+ virtual ~VideoChannelTransport();
+
+ // Start implementation of UdpTransportData.
+ void IncomingRTPPacket(const WebRtc_Word8* incoming_rtp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/);
+
+ void IncomingRTCPPacket(const WebRtc_Word8* incoming_rtcp_packet,
+ const WebRtc_Word32 packet_length,
+ const char* /*from_ip*/,
+ const WebRtc_UWord16 /*from_port*/);
+ // End implementation of UdpTransportData.
+
+ // Specifies the ports to receive RTP packets on.
+ int SetLocalReceiver(WebRtc_UWord16 rtp_port);
+
+ // Specifies the destination port and IP address for a specified channel.
+ int SetSendDestination(const char* ip_address, WebRtc_UWord16 rtp_port);
+
+ private:
+ int channel_;
+ ViENetwork* vie_network_;
+ UdpTransport* socket_transport_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_CHANNEL_TRANSPORT_H_
+
diff --git a/modules/udp_transport/source/traffic_control_windows.cc b/test/udp_transport/traffic_control_win.cc
similarity index 98%
rename from modules/udp_transport/source/traffic_control_windows.cc
rename to test/udp_transport/traffic_control_win.cc
index 09038c0..fcafe7c 100644
--- a/modules/udp_transport/source/traffic_control_windows.cc
+++ b/test/udp_transport/traffic_control_win.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "traffic_control_windows.h"
+#include "traffic_control_win.h"
#include <assert.h>
diff --git a/modules/udp_transport/source/traffic_control_windows.h b/test/udp_transport/traffic_control_win.h
similarity index 100%
rename from modules/udp_transport/source/traffic_control_windows.h
rename to test/udp_transport/traffic_control_win.h
diff --git a/modules/udp_transport/source/udp_socket2_manager_windows.cc b/test/udp_transport/udp_socket2_manager_win.cc
similarity index 98%
rename from modules/udp_transport/source/udp_socket2_manager_windows.cc
rename to test/udp_transport/udp_socket2_manager_win.cc
index 32863d3..94dddc1 100644
--- a/modules/udp_transport/source/udp_socket2_manager_windows.cc
+++ b/test/udp_transport/udp_socket2_manager_win.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "udp_socket2_manager_windows.h"
+#include "udp_socket2_manager_win.h"
#include <assert.h>
#include <stdio.h>
#include "aligned_malloc.h"
-#include "udp_socket2_windows.h"
+#include "udp_socket2_win.h"
namespace webrtc {
WebRtc_UWord32 UdpSocket2ManagerWindows::_numOfActiveManagers = 0;
diff --git a/modules/udp_transport/source/udp_socket2_manager_windows.h b/test/udp_transport/udp_socket2_manager_win.h
similarity index 98%
rename from modules/udp_transport/source/udp_socket2_manager_windows.h
rename to test/udp_transport/udp_socket2_manager_win.h
index 782cb41..8cf5ebc 100644
--- a/modules/udp_transport/source/udp_socket2_manager_windows.h
+++ b/test/udp_transport/udp_socket2_manager_win.h
@@ -22,7 +22,7 @@
#include "event_wrapper.h"
#include "list_wrapper.h"
#include "thread_wrapper.h"
-#include "udp_socket2_windows.h"
+#include "udp_socket2_win.h"
#include "udp_socket_manager_wrapper.h"
#define MAX_IO_BUFF_SIZE 1600
diff --git a/modules/udp_transport/source/udp_socket2_windows.cc b/test/udp_transport/udp_socket2_win.cc
similarity index 98%
rename from modules/udp_transport/source/udp_socket2_windows.cc
rename to test/udp_transport/udp_socket2_win.cc
index 2362cdd..0cb0c61 100644
--- a/modules/udp_transport/source/udp_socket2_windows.cc
+++ b/test/udp_transport/udp_socket2_win.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "udp_socket2_windows.h"
+#include "udp_socket2_win.h"
#include <assert.h>
#include <stdlib.h>
#include <winsock2.h>
-#include "traffic_control_windows.h"
-#include "udp_socket2_manager_windows.h"
+#include "traffic_control_win.h"
+#include "udp_socket2_manager_win.h"
#pragma warning(disable : 4311)
diff --git a/modules/udp_transport/source/udp_socket2_windows.h b/test/udp_transport/udp_socket2_win.h
similarity index 98%
rename from modules/udp_transport/source/udp_socket2_windows.h
rename to test/udp_transport/udp_socket2_win.h
index 8cc46ee..e63f277 100644
--- a/modules/udp_transport/source/udp_socket2_windows.h
+++ b/test/udp_transport/udp_socket2_win.h
@@ -27,7 +27,7 @@
#include "rw_lock_wrapper.h"
#include "trace.h"
#include "udp_socket_wrapper.h"
-#include "udp_socket2_manager_windows.h"
+#include "udp_socket2_manager_win.h"
namespace webrtc {
class UdpSocket2ManagerWindows;
diff --git a/modules/udp_transport/source/udp_socket_manager_posix.cc b/test/udp_transport/udp_socket_manager_posix.cc
similarity index 100%
rename from modules/udp_transport/source/udp_socket_manager_posix.cc
rename to test/udp_transport/udp_socket_manager_posix.cc
diff --git a/modules/udp_transport/source/udp_socket_manager_posix.h b/test/udp_transport/udp_socket_manager_posix.h
similarity index 100%
rename from modules/udp_transport/source/udp_socket_manager_posix.h
rename to test/udp_transport/udp_socket_manager_posix.h
diff --git a/modules/udp_transport/source/udp_socket_manager_unittest.cc b/test/udp_transport/udp_socket_manager_unittest.cc
similarity index 100%
rename from modules/udp_transport/source/udp_socket_manager_unittest.cc
rename to test/udp_transport/udp_socket_manager_unittest.cc
diff --git a/modules/udp_transport/source/udp_socket_manager_wrapper.cc b/test/udp_transport/udp_socket_manager_wrapper.cc
similarity index 97%
rename from modules/udp_transport/source/udp_socket_manager_wrapper.cc
rename to test/udp_transport/udp_socket_manager_wrapper.cc
index 37388be..a1fae5a 100644
--- a/modules/udp_transport/source/udp_socket_manager_wrapper.cc
+++ b/test/udp_transport/udp_socket_manager_wrapper.cc
@@ -14,7 +14,7 @@
#ifdef _WIN32
#include "fix_interlocked_exchange_pointer_win.h"
-#include "udp_socket2_manager_windows.h"
+#include "udp_socket2_manager_win.h"
#else
#include "udp_socket_manager_posix.h"
#endif
diff --git a/modules/udp_transport/source/udp_socket_manager_wrapper.h b/test/udp_transport/udp_socket_manager_wrapper.h
similarity index 100%
rename from modules/udp_transport/source/udp_socket_manager_wrapper.h
rename to test/udp_transport/udp_socket_manager_wrapper.h
diff --git a/modules/udp_transport/source/udp_socket_posix.cc b/test/udp_transport/udp_socket_posix.cc
similarity index 100%
rename from modules/udp_transport/source/udp_socket_posix.cc
rename to test/udp_transport/udp_socket_posix.cc
diff --git a/modules/udp_transport/source/udp_socket_posix.h b/test/udp_transport/udp_socket_posix.h
similarity index 100%
rename from modules/udp_transport/source/udp_socket_posix.h
rename to test/udp_transport/udp_socket_posix.h
diff --git a/modules/udp_transport/source/udp_socket_wrapper.cc b/test/udp_transport/udp_socket_wrapper.cc
similarity index 98%
rename from modules/udp_transport/source/udp_socket_wrapper.cc
rename to test/udp_transport/udp_socket_wrapper.cc
index 6445e76..8948987 100644
--- a/modules/udp_transport/source/udp_socket_wrapper.cc
+++ b/test/udp_transport/udp_socket_wrapper.cc
@@ -18,7 +18,7 @@
#include "udp_socket_manager_wrapper.h"
#if defined(_WIN32)
- #include "udp_socket2_windows.h"
+ #include "udp_socket2_win.h"
#else
#include "udp_socket_posix.h"
#endif
diff --git a/modules/udp_transport/source/udp_socket_wrapper.h b/test/udp_transport/udp_socket_wrapper.h
similarity index 100%
rename from modules/udp_transport/source/udp_socket_wrapper.h
rename to test/udp_transport/udp_socket_wrapper.h
diff --git a/modules/udp_transport/source/udp_socket_wrapper_unittest.cc b/test/udp_transport/udp_socket_wrapper_unittest.cc
similarity index 94%
rename from modules/udp_transport/source/udp_socket_wrapper_unittest.cc
rename to test/udp_transport/udp_socket_wrapper_unittest.cc
index 2489604..0d6c37d 100644
--- a/modules/udp_transport/source/udp_socket_wrapper_unittest.cc
+++ b/test/udp_transport/udp_socket_wrapper_unittest.cc
@@ -22,8 +22,8 @@
#include "gtest/gtest.h"
#include "gmock/gmock.h"
-#include "modules/udp_transport/source/udp_socket_wrapper.h"
-#include "modules/udp_transport/source/udp_socket_manager_wrapper.h"
+#include "webrtc/test/udp_transport/udp_socket_wrapper.h"
+#include "webrtc/test/udp_transport/udp_socket_manager_wrapper.h"
using ::testing::_;
using ::testing::Return;
diff --git a/modules/udp_transport/interface/udp_transport.h b/test/udp_transport/udp_transport.h
similarity index 84%
rename from modules/udp_transport/interface/udp_transport.h
rename to test/udp_transport/udp_transport.h
index 2ca4eb8..8427c7e 100644
--- a/modules/udp_transport/interface/udp_transport.h
+++ b/test/udp_transport/udp_transport.h
@@ -8,12 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
+#ifndef WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
+#define WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
-#include "common_types.h"
-#include "module.h"
-#include "typedefs.h"
+#include "webrtc/common_types.h"
+#include "webrtc/typedefs.h"
/*
* WARNING
@@ -22,7 +21,6 @@
*
*/
-
#define SS_MAXSIZE 128
#define SS_ALIGNSIZE (sizeof (WebRtc_UWord64))
#define SS_PAD1SIZE (SS_ALIGNSIZE - sizeof(WebRtc_Word16))
@@ -31,94 +29,84 @@
// BSD requires use of HAVE_STRUCT_SOCKADDR_SA_LEN
namespace webrtc {
-struct SocketAddressIn
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+struct SocketAddressIn {
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
#else
- WebRtc_Word16 sin_family;
+ WebRtc_Word16 sin_family;
#endif
- WebRtc_UWord16 sin_port;
- WebRtc_UWord32 sin_addr;
- WebRtc_Word8 sin_zero[8];
+ WebRtc_UWord16 sin_port;
+ WebRtc_UWord32 sin_addr;
+ WebRtc_Word8 sin_zero[8];
};
-struct Version6InAddress
-{
- union
- {
- WebRtc_UWord8 _s6_u8[16];
- WebRtc_UWord32 _s6_u32[4];
- WebRtc_UWord64 _s6_u64[2];
- } Version6AddressUnion;
+struct Version6InAddress {
+ union {
+ WebRtc_UWord8 _s6_u8[16];
+ WebRtc_UWord32 _s6_u32[4];
+ WebRtc_UWord64 _s6_u64[2];
+ } Version6AddressUnion;
};
-struct SocketAddressInVersion6
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+struct SocketAddressInVersion6 {
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
#else
- WebRtc_Word16 sin_family;
+ WebRtc_Word16 sin_family;
#endif
- // Transport layer port number.
- WebRtc_UWord16 sin6_port;
- // IPv6 traffic class and flow info or ip4 address.
- WebRtc_UWord32 sin6_flowinfo;
- // IPv6 address
- struct Version6InAddress sin6_addr;
- // Set of interfaces for a scope.
- WebRtc_UWord32 sin6_scope_id;
+ // Transport layer port number.
+ WebRtc_UWord16 sin6_port;
+ // IPv6 traffic class and flow info or ip4 address.
+ WebRtc_UWord32 sin6_flowinfo;
+ // IPv6 address
+ struct Version6InAddress sin6_addr;
+ // Set of interfaces for a scope.
+ WebRtc_UWord32 sin6_scope_id;
};
-struct SocketAddressStorage
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
+struct SocketAddressStorage {
+ // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
+ WebRtc_Word8 sin_length;
+ WebRtc_Word8 sin_family;
#else
- WebRtc_Word16 sin_family;
+ WebRtc_Word16 sin_family;
#endif
- WebRtc_Word8 __ss_pad1[SS_PAD1SIZE];
- WebRtc_UWord64 __ss_align;
- WebRtc_Word8 __ss_pad2[SS_PAD2SIZE];
+ WebRtc_Word8 __ss_pad1[SS_PAD1SIZE];
+ WebRtc_UWord64 __ss_align;
+ WebRtc_Word8 __ss_pad2[SS_PAD2SIZE];
};
-struct SocketAddress
-{
- union
- {
- struct SocketAddressIn _sockaddr_in;
- struct SocketAddressInVersion6 _sockaddr_in6;
- struct SocketAddressStorage _sockaddr_storage;
- };
+struct SocketAddress {
+ union {
+ struct SocketAddressIn _sockaddr_in;
+ struct SocketAddressInVersion6 _sockaddr_in6;
+ struct SocketAddressStorage _sockaddr_storage;
+ };
};
// Callback class that receives packets from UdpTransport.
-class UdpTransportData
-{
-public:
- virtual ~UdpTransportData() {};
+class UdpTransportData {
+ public:
+ virtual ~UdpTransportData() {};
- virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort) = 0;
+ virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
+ const WebRtc_Word32 rtpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort) = 0;
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort) = 0;
+ virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
+ const WebRtc_Word32 rtcpPacketLength,
+ const char* fromIP,
+ const WebRtc_UWord16 fromPort) = 0;
};
-
-class UdpTransport : public Module, public Transport
-{
-public:
+class UdpTransport : public Transport {
+ public:
enum
{
kIpAddressVersion6Length = 64,
@@ -389,6 +377,7 @@
// is interptreted as IPv6.
static bool IsIpAddressValid(const char* ipaddr, const bool ipV6);
};
+
} // namespace webrtc
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
+#endif // WEBRTC_TEST_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
diff --git a/modules/udp_transport/source/udp_transport_impl.cc b/test/udp_transport/udp_transport_impl.cc
similarity index 98%
rename from modules/udp_transport/source/udp_transport_impl.cc
rename to test/udp_transport/udp_transport_impl.cc
index b7ded87..875ce93 100644
--- a/modules/udp_transport/source/udp_transport_impl.cc
+++ b/test/udp_transport/udp_transport_impl.cc
@@ -179,44 +179,6 @@
__FUNCTION__);
}
-WebRtc_Word32 UdpTransportImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
-
- CriticalSectionScoped cs(_crit);
- _id = id;
- if(_mgr)
- {
- _mgr->ChangeUniqueId(id);
- }
- if(_ptrRtpSocket)
- {
- _ptrRtpSocket->ChangeUniqueId(id);
- }
- if(_ptrRtcpSocket)
- {
- _ptrRtcpSocket->ChangeUniqueId(id);
- }
- if(_ptrSendRtpSocket)
- {
- _ptrSendRtpSocket->ChangeUniqueId(id);
- }
- if(_ptrSendRtcpSocket)
- {
- _ptrSendRtcpSocket->ChangeUniqueId(id);
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::TimeUntilNextProcess()
-{
- return 100;
-}
-
-WebRtc_Word32 UdpTransportImpl::Process()
-{
- return 0;
-}
-
UdpTransport::ErrorCode UdpTransportImpl::LastError() const
{
return _lastError;
@@ -1398,7 +1360,7 @@
{
SocketAddress stLclName;
#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- stLclName.sin_length = 0;
+ stLclName.sin_lenght = 0;
stLclName.sin_family = PF_INET6;
#else
stLclName._sockaddr_storage.sin_family = PF_INET6;
@@ -2561,7 +2523,7 @@
req.r.ifa_family = AF_INET6;
// Fill up all the attributes for the rtnetlink header.
- // The length is very important. 16 signifies the ipv6 address.
+ // The lenght is very important. 16 signifies the ipv6 address.
rta = (struct rtattr*)(((char*)&req) + NLMSG_ALIGN(req.n.nlmsg_len));
rta->rta_len = RTA_LENGTH(16);
diff --git a/modules/udp_transport/source/udp_transport_impl.h b/test/udp_transport/udp_transport_impl.h
similarity index 97%
rename from modules/udp_transport/source/udp_transport_impl.h
rename to test/udp_transport/udp_transport_impl.h
index 9f4fd9f..e2fa795 100644
--- a/modules/udp_transport/source/udp_transport_impl.h
+++ b/test/udp_transport/udp_transport_impl.h
@@ -42,11 +42,6 @@
UdpSocketManager* socket_manager);
virtual ~UdpTransportImpl();
- // Module functions
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
// UdpTransport functions
virtual WebRtc_Word32 InitializeSendSockets(
const char* ipAddr,
diff --git a/modules/udp_transport/source/udp_transport_unittest.cc b/test/udp_transport/udp_transport_unittest.cc
similarity index 96%
rename from modules/udp_transport/source/udp_transport_unittest.cc
rename to test/udp_transport/udp_transport_unittest.cc
index 3125e2e..243cbda 100644
--- a/modules/udp_transport/source/udp_transport_unittest.cc
+++ b/test/udp_transport/udp_transport_unittest.cc
@@ -10,14 +10,14 @@
#include <vector>
-#include "udp_transport.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
+#include "webrtc/test/udp_transport/udp_transport.h"
// We include the implementation header file to get at the dependency-injecting
// constructor.
-#include "udp_transport_impl.h"
+#include "webrtc/test/udp_transport/udp_transport_impl.h"
// We must mock the socket manager, for which we need its definition.
-#include "udp_socket_manager_wrapper.h"
+#include "webrtc/test/udp_transport/udp_socket_manager_wrapper.h"
using ::testing::_;
using ::testing::Return;
diff --git a/video_engine/include/vie_network.h b/video_engine/include/vie_network.h
index 9752008..d3515db 100644
--- a/video_engine/include/vie_network.h
+++ b/video_engine/include/vie_network.h
@@ -65,34 +65,6 @@
// for all sub-API:s before the VideoEngine object can be safely deleted.
virtual int Release() = 0;
- // Specifies the ports to receive RTP packets on. It is also possible to set
- // port for RTCP and local IP address.
- virtual int SetLocalReceiver(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const char* ip_address = NULL) = 0;
-
- // Gets the local receiver ports and address for a specified channel.
- virtual int GetLocalReceiver(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port, char* ip_address) = 0;
-
- // Specifies the destination port and IP address for a specified channel.
- virtual int SetSendDestination(const int video_channel,
- const char* ip_address,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const unsigned short source_rtp_port = 0,
- const unsigned short source_rtcp_port = 0) = 0;
-
- // Get the destination port and address for a specified channel.
- virtual int GetSendDestination(const int video_channel,
- char* ip_address,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- unsigned short& source_rtp_port,
- unsigned short& source_rtcp_port) = 0;
-
// This function registers a user implementation of Transport to use for
// sending RTP and RTCP packets on this channel.
virtual int RegisterSendTransport(const int video_channel,
@@ -114,63 +86,6 @@
const void* data,
const int length) = 0;
- // Gets the source ports and IP address of the incoming stream for a
- // specified channel.
- virtual int GetSourceInfo(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address,
- unsigned int ip_address_length) = 0;
-
- // Gets the local IP address, in string format.
- virtual int GetLocalIP(char ip_address[64], bool ipv6 = false) = 0;
-
- // Enables IPv6, instead of IPv4, for a specified channel.
- virtual int EnableIPv6(int video_channel) = 0;
-
- // The function returns true if IPv6 is enabled, false otherwise.
- virtual bool IsIPv6Enabled(int video_channel) = 0;
-
- // Enables a port and IP address filtering for incoming packets on a
- // specific channel.
- virtual int SetSourceFilter(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const char* ip_address = NULL) = 0;
-
- // Gets current port and IP address filter for a specified channel.
- virtual int GetSourceFilter(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address) = 0;
-
- // This function sets the six‐bit Differentiated Services Code Point (DSCP)
- // in the IP header of the outgoing stream for a specific channel.
- // Windows and Linux only.
- virtual int SetSendToS(const int video_channel,
- const int DSCP,
- const bool use_set_sockOpt = false) = 0;
-
- // Retrieves the six‐bit Differentiated Services Code Point (DSCP) in the IP
- // header of the outgoing stream for a specific channel.
- virtual int GetSendToS(const int video_channel,
- int& DSCP,
- bool& use_set_sockOpt) = 0;
-
- // This function sets the Generic Quality of Service (GQoS) service level.
- // The Windows operating system then maps to a Differentiated Services Code
- // Point (DSCP) and to an 802.1p setting. Windows only.
- virtual int SetSendGQoS(const int video_channel, const bool enable,
- const int service_type,
- const int overrideDSCP = 0) = 0;
-
- // This function retrieves the currently set GQoS service level for a
- // specific channel.
- virtual int GetSendGQoS(const int video_channel,
- bool& enabled,
- int& service_type,
- int& overrideDSCP) = 0;
-
// This function sets the Maximum Transition Unit (MTU) for a channel. The
// RTP packet will be packetized based on this MTU to optimize performance
// over the network.
@@ -197,14 +112,6 @@
const bool enable,
const unsigned int sample_time_seconds = KDefaultSampleTimeSeconds) = 0;
- // This function handles sending a raw UDP data packet over an existing RTP
- // or RTCP socket.
- virtual int SendUDPPacket(const int video_channel,
- const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket = false) = 0;
-
protected:
ViENetwork() {}
virtual ~ViENetwork() {}
diff --git a/video_engine/test/android/jni/vie_android_java_api.cc b/video_engine/test/android/jni/vie_android_java_api.cc
index 8c8aa0b..d1b6f2b 100644
--- a/video_engine/test/android/jni/vie_android_java_api.cc
+++ b/video_engine/test/android/jni/vie_android_java_api.cc
@@ -34,6 +34,9 @@
#include "common_types.h"
#include "android_media_codec_decoder.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+
#define WEBRTC_LOG_TAG "*WEBRTCN*"
#define VALIDATE_BASE_POINTER \
if (!voeData.base) \
@@ -107,6 +110,8 @@
VoEHardware* hardware;
VoERTP_RTCP* rtp;
JavaVM* jvm;
+ scoped_ptr<VoiceChannelTransport> transport;
+ // TODO
} VoiceEngineData;
class AndroidVideoRenderCallback;
@@ -122,6 +127,7 @@
ViECapture* capture;
ViEExternalCodec* externalCodec;
VideoCallbackAndroid* callback;
+ scoped_ptr<VideoChannelTransport> transport;
} VideoEngineData;
@@ -589,7 +595,7 @@
if (voiceChannel >= 0) {
vieData.base->ConnectAudioChannel(channel, voiceChannel);
}
-
+ vieData.transport.reset(new VideoChannelTransport(vieData.netw, channel));
return channel;
}
else {
@@ -610,13 +616,10 @@
{
__android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
- if (vieData.vie) {
- int ret = vieData.netw->SetLocalReceiver(channel, port);
- return ret;
+ if (vieData.transport.get()) {
+ return vieData.transport->SetLocalReceiver(port);
}
- else {
- return -1;
- }
+ return -1;
}
/*
@@ -646,7 +649,10 @@
"SetSendDestination: channel=%d, port=%d, ip=%s\n",
channel, port, ip);
- return vieData.netw->SetSendDestination(channel, ip, port);
+ if (vieData.transport.get()) {
+ return vieData.transport->SetSendDestination(ip, port);
+ }
+ return -1;
}
@@ -1264,7 +1270,7 @@
}
jint channel = voeData.base->CreateChannel();
-
+ voeData.transport.reset(new VoiceChannelTransport(voeData.netw, channel));
return channel;
}
@@ -1295,7 +1301,10 @@
{
__android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
VALIDATE_BASE_POINTER;
- return voeData.base->SetLocalReceiver(channel, port);
+ if (voeData.transport.get()) {
+ return voeData.transport->SetLocalReceiver(port);
+ }
+ return -1;
}
/*
@@ -1319,9 +1328,13 @@
"Could not get UTF string");
return -1;
}
- jint retVal = voeData.base->SetSendDestination(channel, port, ipaddrNative);
+ if (voeData.transport.get()) {
+ jint retVal = voeData.transport->SetSendDestination(ipaddrNative, port);
+ env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
+ return retVal;
+ }
env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
- return retVal;
+ return -1;
}
/*
diff --git a/video_engine/test/auto_test/automated/vie_api_integration_test.cc b/video_engine/test/auto_test/automated/vie_api_integration_test.cc
index 7b49103..96b401b 100644
--- a/video_engine/test/auto_test/automated/vie_api_integration_test.cc
+++ b/video_engine/test/auto_test/automated/vie_api_integration_test.cc
@@ -48,10 +48,6 @@
tests_->ViEImageProcessAPITest();
}
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest), RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkAPITest();
-}
-
TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest), RunsRenderTestWithoutErrors) {
tests_->ViERenderAPITest();
}
diff --git a/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
index c091475..d224bde 100644
--- a/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ b/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
@@ -51,11 +51,6 @@
}
TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
- RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkExtendedTest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
RunsRenderTestWithoutErrors) {
tests_->ViERenderExtendedTest();
}
diff --git a/video_engine/test/auto_test/automated/vie_standard_integration_test.cc b/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
index 4f90aac..30364e0 100644
--- a/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
+++ b/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
@@ -54,10 +54,6 @@
tests_->ViEImageProcessStandardTest();
}
-TEST_F(ViEStandardIntegrationTest, RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkStandardTest();
-}
-
TEST_F(ViEStandardIntegrationTest, RunsRenderTestWithoutErrors) {
tests_->ViERenderStandardTest();
}
diff --git a/video_engine/test/auto_test/source/vie_autotest.cc b/video_engine/test/auto_test/source/vie_autotest.cc
index 363eccb..200e256 100644
--- a/video_engine/test/auto_test/source/vie_autotest.cc
+++ b/video_engine/test/auto_test/source/vie_autotest.cc
@@ -67,7 +67,6 @@
ViEEncryptionStandardTest();
ViEFileStandardTest();
ViEImageProcessStandardTest();
- ViENetworkStandardTest();
ViERenderStandardTest();
ViERtpRtcpStandardTest();
}
@@ -80,7 +79,6 @@
ViEEncryptionExtendedTest();
ViEFileExtendedTest();
ViEImageProcessExtendedTest();
- ViENetworkExtendedTest();
ViERenderExtendedTest();
ViERtpRtcpExtendedTest();
}
@@ -93,7 +91,6 @@
ViEEncryptionAPITest();
ViEFileAPITest();
ViEImageProcessAPITest();
- ViENetworkAPITest();
ViERenderAPITest();
ViERtpRtcpAPITest();
}
diff --git a/video_engine/test/auto_test/source/vie_autotest_android.cc b/video_engine/test/auto_test/source/vie_autotest_android.cc
index d1622db..80d8d90 100644
--- a/video_engine/test/auto_test/source/vie_autotest_android.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_android.cc
@@ -162,21 +162,14 @@
vieAutoTest.ViEImageProcessExtendedTest();
break;
- case 8: // network
- vieAutoTest.ViENetworkExtendedTest();
- break;
-
- case 9: // Render
+ case 8: // Render
vieAutoTest.ViERenderExtendedTest();
break;
- case 10: // RTP/RTCP
+ case 9: // RTP/RTCP
vieAutoTest.ViERtpRtcpExtendedTest();
break;
- case 11:
- break;
-
default:
break;
}
diff --git a/video_engine/test/auto_test/source/vie_autotest_base.cc b/video_engine/test/auto_test/source/vie_autotest_base.cc
index e9d24f3..8fc7824 100644
--- a/video_engine/test/auto_test/source/vie_autotest_base.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_base.cc
@@ -9,11 +9,12 @@
*/
#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/primitives/base_primitives.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-#include "video_engine/test/libvietest/include/tb_interfaces.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "webrtc/video_engine/test/auto_test/primitives/base_primitives.h"
+#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
void ViEAutoTest::ViEBaseStandardTest() {
// ***************************************************************
@@ -150,20 +151,30 @@
// Create a receive only channel and a send channel. Verify we can't send on
// the receive only channel.
EXPECT_EQ(0, vie_base->CreateReceiveChannel(video_channel2,
- video_channel));
+ video_channel));
EXPECT_EQ(0, vie_base->CreateChannel(video_channel3, video_channel));
const char* ip_address = "127.0.0.1\0";
const int send_port = 1234;
+
EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 1));
- EXPECT_EQ(0, vie_network->SetSendDestination(video_channel, ip_address,
- send_port));
EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 2));
- EXPECT_EQ(0, vie_network->SetSendDestination(video_channel2, ip_address,
- send_port + 2));
EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 3));
- EXPECT_EQ(0, vie_network->SetSendDestination(video_channel3, ip_address,
- send_port + 4));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport_1(
+ new webrtc::VideoChannelTransport(vie_network, video_channel));
+
+ ASSERT_EQ(0, video_channel_transport_1->SetSendDestination(ip_address,
+ send_port));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport_2(
+ new webrtc::VideoChannelTransport(vie_network, video_channel2));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport_3(
+ new webrtc::VideoChannelTransport(vie_network, video_channel3));
+
+ ASSERT_EQ(0, video_channel_transport_3->SetSendDestination(ip_address,
+ send_port + 4));
EXPECT_EQ(0, vie_base->StartSend(video_channel));
EXPECT_EQ(-1, vie_base->StartSend(video_channel2));
diff --git a/video_engine/test/auto_test/source/vie_autotest_codec.cc b/video_engine/test/auto_test/source/vie_autotest_codec.cc
index 87169a6..74ce9d0 100644
--- a/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -8,21 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "common_types.h" // NOLINT
-#include "engine_configurations.h" // NOLINT
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/test/libvietest/include/tb_capture_device.h"
-#include "video_engine/test/libvietest/include/tb_I420_codec.h"
-#include "video_engine/test/libvietest/include/tb_interfaces.h"
-#include "video_engine/test/libvietest/include/tb_video_channel.h"
-#include "video_engine/include/vie_base.h"
-#include "video_engine/include/vie_capture.h"
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/include/vie_network.h"
-#include "video_engine/include/vie_render.h"
-#include "video_engine/include/vie_rtp_rtcp.h"
-#include "voice_engine/include/voe_base.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_I420_codec.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
+#include "webrtc/video_engine/include/vie_base.h"
+#include "webrtc/video_engine/include/vie_capture.h"
+#include "webrtc/video_engine/include/vie_codec.h"
+#include "webrtc/video_engine/include/vie_external_codec.h"
+#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/video_engine/include/vie_render.h"
+#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_base.h"
class TestCodecObserver
: public webrtc::ViEEncoderObserver,
@@ -164,13 +167,16 @@
break;
}
}
-
const char* ip_address = "127.0.0.1";
const uint16_t rtp_port = 6000;
- EXPECT_EQ(0, network->SetLocalReceiver(video_channel, rtp_port));
- EXPECT_EQ(0, base->StartReceive(video_channel));
- EXPECT_EQ(0, network->SetSendDestination(
- video_channel, ip_address, rtp_port));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport(
+ new webrtc::VideoChannelTransport(network, video_channel));
+
+ ASSERT_EQ(0, video_channel_transport->SetSendDestination(ip_address,
+ rtp_port));
+ ASSERT_EQ(0, video_channel_transport->SetLocalReceiver(rtp_port));
+
EXPECT_EQ(0, base->StartSend(video_channel));
// Make sure all codecs runs
@@ -300,10 +306,14 @@
const char* ip_address = "127.0.0.1";
const uint16_t rtp_port = 6000;
- EXPECT_EQ(0, network->SetLocalReceiver(video_channel, rtp_port));
- EXPECT_EQ(0, base->StartReceive(video_channel));
- EXPECT_EQ(0, network->SetSendDestination(
- video_channel, ip_address, rtp_port));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport(
+ new webrtc::VideoChannelTransport(network, video_channel));
+
+ ASSERT_EQ(0, video_channel_transport->SetSendDestination(ip_address,
+ rtp_port));
+ ASSERT_EQ(0, video_channel_transport->SetLocalReceiver(rtp_port));
+
EXPECT_EQ(0, base->StartSend(video_channel));
// Codec specific tests
@@ -329,6 +339,7 @@
// the received streams.
TbInterfaces video_engine("ViECodecExtendedTest2");
TbCaptureDevice tb_capture(video_engine);
+ webrtc::ViENetwork* network = video_engine.network;
// Create channel 1.
int video_channel_1 = -1;
@@ -341,17 +352,25 @@
EXPECT_NE(video_channel_1, video_channel_2)
<< "Channel 2 should be unique.";
+ const char* ip_address = "127.0.0.1";
uint16_t rtp_port_1 = 12000;
uint16_t rtp_port_2 = 13000;
- EXPECT_EQ(0, video_engine.network->SetLocalReceiver(
- video_channel_1, rtp_port_1));
- EXPECT_EQ(0, video_engine.network->SetSendDestination(
- video_channel_1, "127.0.0.1", rtp_port_1));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport_1(
+ new webrtc::VideoChannelTransport(network, video_channel_1));
+
+ ASSERT_EQ(0, video_channel_transport_1->SetSendDestination(ip_address,
+ rtp_port_1));
+ ASSERT_EQ(0, video_channel_transport_1->SetLocalReceiver(rtp_port_1));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport_2(
+ new webrtc::VideoChannelTransport(network, video_channel_2));
+
+ ASSERT_EQ(0, video_channel_transport_2->SetSendDestination(ip_address,
+ rtp_port_2));
+ ASSERT_EQ(0, video_channel_transport_2->SetLocalReceiver(rtp_port_2));
+
EXPECT_EQ(0, video_engine.rtp_rtcp->SetLocalSSRC(video_channel_1, 1));
- EXPECT_EQ(0, video_engine.network->SetLocalReceiver(
- video_channel_2, rtp_port_2));
- EXPECT_EQ(0, video_engine.network->SetSendDestination(
- video_channel_2, "127.0.0.1", rtp_port_2));
EXPECT_EQ(0, video_engine.rtp_rtcp->SetLocalSSRC(video_channel_2, 2));
tb_capture.ConnectTo(video_channel_1);
tb_capture.ConnectTo(video_channel_2);
@@ -483,9 +502,6 @@
EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine));
}
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-#include "video_engine/include/vie_external_codec.h"
-#endif
void ViEAutoTest::ViECodecExternalCodecTest() {
ViETest::Log(" ");
ViETest::Log("========================================");
diff --git a/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
index b862b64..08c4438 100644
--- a/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
@@ -15,12 +15,15 @@
#include <algorithm>
#include "gflags/gflags.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/primitives/choice_helpers.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-#include "video_engine/test/auto_test/primitives/input_helpers.h"
-#include "video_engine/test/libvietest/include/vie_to_file_renderer.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "webrtc/video_engine/test/auto_test/primitives/choice_helpers.h"
+#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
+#include "webrtc/video_engine/test/auto_test/primitives/input_helpers.h"
+#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
+#include "webrtc/voice_engine/include/voe_network.h"
#define VCM_RED_PAYLOAD_TYPE 96
#define VCM_ULPFEC_PAYLOAD_TYPE 97
@@ -202,6 +205,12 @@
"ERROR: %s at line %d", __FUNCTION__,
__LINE__);
+ webrtc::VoENetwork* voe_network=
+ webrtc::VoENetwork::GetInterface(voe);
+ number_of_errors += ViETest::TestError(voe_network != NULL,
+ "ERROR: %s at line %d", __FUNCTION__,
+ __LINE__);
+
webrtc::VoEAudioProcessing* voe_apm =
webrtc::VoEAudioProcessing::GetInterface(voe);
number_of_errors += ViETest::TestError(voe_apm != NULL,
@@ -270,6 +279,8 @@
int buffer_delay_ms = 0;
bool is_image_scale_enabled = false;
bool remb = true;
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport;
+ webrtc::scoped_ptr<webrtc::VoiceChannelTransport> voice_channel_transport;
while (!start_call) {
// Get the IP address to use from call.
@@ -340,13 +351,17 @@
if (start_call == true) {
// Configure audio channel first.
audio_channel = voe_base->CreateChannel();
- error = voe_base->SetSendDestination(audio_channel, audio_tx_port,
- ip_address.c_str());
+
+ voice_channel_transport.reset(
+ new webrtc::VoiceChannelTransport(voe_network, audio_channel));
+
+ error = voice_channel_transport->SetSendDestination(ip_address.c_str(),
+ audio_tx_port);
number_of_errors += ViETest::TestError(error == 0,
"ERROR: %s at line %d",
__FUNCTION__, __LINE__);
- error = voe_base->SetLocalReceiver(audio_channel, audio_rx_port);
+ error = voice_channel_transport->SetLocalReceiver(audio_rx_port);
number_of_errors += ViETest::TestError(error == 0,
"ERROR: %s at line %d",
__FUNCTION__, __LINE__);
@@ -472,13 +487,17 @@
file_renderer.PrepareForRendering(output_path, filename);
RenderToFile(vie_renderer, video_channel, &file_renderer);
}
- error = vie_network->SetSendDestination(video_channel, ip_address.c_str(),
- video_tx_port);
+
+ video_channel_transport.reset(
+ new webrtc::VideoChannelTransport(vie_network, video_channel));
+
+ error = video_channel_transport->SetSendDestination(ip_address.c_str(),
+ video_tx_port);
number_of_errors += ViETest::TestError(error == 0,
"ERROR: %s at line %d",
__FUNCTION__, __LINE__);
- error = vie_network->SetLocalReceiver(video_channel, video_rx_port);
+ error = video_channel_transport->SetLocalReceiver(video_rx_port);
number_of_errors += ViETest::TestError(error == 0,
"ERROR: %s at line %d",
__FUNCTION__, __LINE__);
diff --git a/video_engine/test/auto_test/source/vie_autotest_file.cc b/video_engine/test/auto_test/source/vie_autotest_file.cc
index 8083171..ce2d327 100644
--- a/video_engine/test/auto_test/source/vie_autotest_file.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_file.cc
@@ -18,6 +18,9 @@
#include "voe_codec.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+
class ViEAutotestFileObserver: public webrtc::ViEFileObserver
{
public:
@@ -103,14 +106,18 @@
break;
}
}
-
-
const char* ipAddress = "127.0.0.1";
const unsigned short rtpPort = 6000;
- EXPECT_EQ(0, ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort));
+
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport>
+ video_channel_transport(
+ new webrtc::VideoChannelTransport(ptrViENetwork, videoChannel));
+
+ EXPECT_EQ(0, video_channel_transport->SetSendDestination(ipAddress,
+ rtpPort));
+ EXPECT_EQ(0, video_channel_transport->SetLocalReceiver(rtpPort));
+
EXPECT_EQ(0, ptrViEBase->StartReceive(videoChannel));
- EXPECT_EQ(0, ptrViENetwork->SetSendDestination(
- videoChannel, ipAddress, rtpPort));
EXPECT_EQ(0, ptrViEBase->StartSend(videoChannel));
webrtc::ViEFile* ptrViEFile = webrtc::ViEFile::GetInterface(ptrViE);
EXPECT_TRUE(ptrViEFile != NULL);
diff --git a/video_engine/test/auto_test/source/vie_autotest_loopback.cc b/video_engine/test/auto_test/source/vie_autotest_loopback.cc
index 1f8fcdd..50d4349 100644
--- a/video_engine/test/auto_test/source/vie_autotest_loopback.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_loopback.cc
@@ -32,6 +32,8 @@
#include "vie_network.h"
#include "vie_render.h"
#include "vie_rtp_rtcp.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
#define VCM_RED_PAYLOAD_TYPE 96
#define VCM_ULPFEC_PAYLOAD_TYPE 97
@@ -472,6 +474,9 @@
// Setting External transport
TbExternalTransport extTransport(*(ptrViENetwork), videoChannel, NULL);
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport(
+ new webrtc::VideoChannelTransport(ptrViENetwork, videoChannel));
+
int testMode = 0;
std::cout << std::endl;
std::cout << "Enter 1 for testing packet loss and delay with "
@@ -524,17 +529,17 @@
std::cout << std::endl;
std::cout << "Using rtp port: " << rtpPort << std::endl;
std::cout << std::endl;
- error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
+
+ error = video_channel_transport->SetLocalReceiver(rtpPort);
if (error == -1)
{
- printf("ERROR in ViENetwork::SetLocalReceiver\n");
+ printf("ERROR in SetLocalReceiver\n");
return -1;
}
- error = ptrViENetwork->SetSendDestination(videoChannel,
- ipAddress, rtpPort);
+ error = video_channel_transport->SetSendDestination(ipAddress, rtpPort);
if (error == -1)
{
- printf("ERROR in ViENetwork::SetSendDestination\n");
+ printf("ERROR in SetSendDestination\n");
return -1;
}
}
diff --git a/video_engine/test/auto_test/source/vie_autotest_network.cc b/video_engine/test/auto_test/source/vie_autotest_network.cc
deleted file mode 100644
index 550bd4f..0000000
--- a/video_engine/test/auto_test/source/vie_autotest_network.cc
+++ /dev/null
@@ -1,573 +0,0 @@
-/*
- * 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.
- */
-
-//
-// vie_autotest_network.cc
-//
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "tb_capture_device.h"
-#include "tb_external_transport.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-
-#if defined(_WIN32)
-#include <qos.h>
-#endif
-
-class ViEAutoTestNetworkObserver: public webrtc::ViENetworkObserver
-{
-public:
- ViEAutoTestNetworkObserver()
- {
- }
- virtual ~ViEAutoTestNetworkObserver()
- {
- }
- virtual void OnPeriodicDeadOrAlive(const int videoChannel, const bool alive)
- {
- }
- virtual void PacketTimeout(const int videoChannel,
- const webrtc::ViEPacketTimeout timeout)
- {
- }
-};
-
-void ViEAutoTest::ViENetworkStandardTest()
-{
- TbInterfaces ViE("ViENetworkStandardTest"); // Create VIE
- TbCaptureDevice tbCapture(ViE);
- {
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
-
- //
- // Transport
- //
- TbExternalTransport testTransport(*ViE.network, tbChannel.videoChannel,
- NULL);
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
-
- ViETest::Log("Call started using external transport, video should "
- "see video in both windows\n");
- AutoTestSleep(kAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel));
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- unsigned short rtpPort = 1234;
- memcpy(myIpAddress, "127.0.0.1", sizeof("127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, myIpAddress, rtpPort,
- rtpPort + 1, rtpPort));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- ViETest::Log("Changed to WebRTC SocketTransport, you should still see "
- "video in both windows\n");
- AutoTestSleep(kAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort + 10, rtpPort + 11, myIpAddress));
- ViETest::Log("Added UDP port filter for incorrect ports, you should "
- "not see video in Window2");
- AutoTestSleep(2000);
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, "123.1.1.0"));
- ViETest::Log("Added IP filter for incorrect IP address, you should not "
- "see video in Window2");
- AutoTestSleep(2000);
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- ViETest::Log("Added IP filter for this computer, you should see video "
- "in Window2 again\n");
- AutoTestSleep(kAutoTestSleepTimeMs);
-
- tbCapture.Disconnect(tbChannel.videoChannel);
- }
-}
-
-void ViEAutoTest::ViENetworkExtendedTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- TbInterfaces ViE("ViENetworkExtendedTest"); // Create VIE
- TbCaptureDevice tbCapture(ViE);
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-
- {
- //
- // ToS
- //
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- tbCapture.ConnectTo(tbChannel.videoChannel);
- const char* remoteIp = "192.168.200.1";
- int DSCP = 0;
- bool useSetSockOpt = false;
-
- webrtc::VideoCodec videoCodec;
- EXPECT_EQ(0, ViE.codec->GetSendCodec(
- tbChannel.videoChannel, videoCodec));
- videoCodec.maxFramerate = 5;
- EXPECT_EQ(0, ViE.codec->SetSendCodec(
- tbChannel.videoChannel, videoCodec));
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- unsigned short rtpPort = 9000;
- EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, remoteIp, rtpPort, rtpPort + 1, rtpPort));
-
- // ToS
- int tos_result = ViE.network->SetSendToS(tbChannel.videoChannel, 2);
- EXPECT_EQ(0, tos_result);
- if (tos_result != 0)
- {
- ViETest::Log("ViESetSendToS error!.");
- ViETest::Log("You must be admin to run these tests.");
- ViETest::Log("On Win7 and late Vista, you need to right click the "
- "exe and choose");
- ViETest::Log("\"Run as administrator\"\n");
- getchar();
- }
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
-
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- ViETest::Log("Use Wireshark to capture the outgoing video stream and "
- "verify ToS settings\n");
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0));
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 2, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- tbCapture.Disconnect(tbChannel.videoChannel);
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViENetworkAPITest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- TbInterfaces ViE("ViENetworkAPITest"); // Create VIE
- {
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecI420);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // External transport
- //
- TbExternalTransport testTransport(*ViE.network, tbChannel.videoChannel,
- NULL);
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
- EXPECT_NE(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
-
- // Create a empty RTP packet.
- unsigned char packet[3000];
- memset(packet, 0, sizeof(packet));
- packet[0] = 0x80; // V=2, P=0, X=0, CC=0
- packet[1] = 0x7C; // M=0, PT = 124 (I420)
-
- // Create a empty RTCP app packet.
- unsigned char rtcpacket[3000];
- memset(rtcpacket,0, sizeof(rtcpacket));
- rtcpacket[0] = 0x80; // V=2, P=0, X=0, CC=0
- rtcpacket[1] = 0xCC; // M=0, PT = 204 (RTCP app)
- rtcpacket[2] = 0x0;
- rtcpacket[3] = 0x03; // 3 Octets long.
-
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_NE(0, ViE.network->ReceivedRTCPPacket(
- tbChannel.videoChannel, rtcpacket, 1500));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_EQ(0, ViE.network->ReceivedRTCPPacket(
- tbChannel.videoChannel, rtcpacket, 1500));
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 11));
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 11));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 3000));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 3000));
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel)); // Sending
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel)); // Already deregistered
-
- //
- // Local receiver
- //
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
-
- unsigned short rtpPort = 0;
- unsigned short rtcpPort = 0;
- char ipAddress[64];
- memset(ipAddress, 0, 64);
- EXPECT_EQ(0, ViE.network->GetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->GetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
- //
- // Send destination
- //
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1236, 1237, 1234, 1235));
-
- unsigned short sourceRtpPort = 0;
- unsigned short sourceRtcpPort = 0;
- EXPECT_EQ(0, ViE.network->GetSendDestination(
- tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
- sourceRtpPort, sourceRtcpPort));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- // Not allowed while sending
- EXPECT_NE(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(kViENetworkAlreadySending, ViE.base->LastError());
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->GetSendDestination(
- tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
- sourceRtpPort, sourceRtcpPort));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- //
- // Address information
- //
-
- // GetSourceInfo: Tested in functional test
- EXPECT_EQ(0, ViE.network->GetLocalIP(ipAddress, false));
-
- // TODO(unknown): IPv6
-
- //
- // Filter
- //
- EXPECT_NE(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 1234, 1235, "10.10.10.10"));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 0, 0, NULL));
- EXPECT_NE(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- }
- {
- TbVideoChannel tbChannel(ViE); // Create a video channel
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234));
-
- int DSCP = 0;
- bool useSetSockOpt = false;
- // SetSockOpt should work without a locally bind socket
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- EXPECT_EQ(0, DSCP);
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, -1, true));
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, 64, true));
-
- // Valid
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 20, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
-
- EXPECT_EQ(20, DSCP);
- EXPECT_TRUE(useSetSockOpt);
-
- // Disable
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(0, DSCP);
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- // Get local ip to be able to set ToS withtou setSockOpt
- EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, myIpAddress));
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(
- tbChannel.videoChannel, -1, false));
- EXPECT_NE(0, ViE.network->SetSendToS(
- tbChannel.videoChannel, 64, false)); // Invalid input
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- EXPECT_EQ(0, DSCP);
- int tos_result = ViE.network->SetSendToS(
- tbChannel.videoChannel, 20, false); // Valid
- EXPECT_EQ(0, tos_result);
- if (tos_result != 0)
- {
- ViETest::Log("ViESetSendToS error!.");
- ViETest::Log("You must be admin to run these tests.");
- ViETest::Log("On Win7 and late Vista, you need to right click the "
- "exe and choose");
- ViETest::Log("\"Run as administrator\"\n");
- getchar();
- }
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(20, DSCP);
-#ifdef _WIN32
- EXPECT_FALSE(useSetSockOpt);
-#else // useSetSockOpt is true on Linux and Mac
- EXPECT_TRUE(useSetSockOpt);
-#endif
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, false));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(0, DSCP);
- }
- {
- // From qos.h. (*) -> supported by ViE
- //
- // #define SERVICETYPE_NOTRAFFIC 0x00000000
- // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
- // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
- // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
- // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
- // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
- // #define SERVICETYPE_NOCHANGE 0x00000006
- // #define SERVICETYPE_NONCONFORMING 0x00000009
- // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
- // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
- //
- // #define SERVICE_BESTEFFORT 0x80010000
- // #define SERVICE_CONTROLLEDLOAD 0x80020000
- // #define SERVICE_GUARANTEED 0x80040000
- // #define SERVICE_QUALITATIVE 0x80200000
-
- TbVideoChannel tbChannel(ViE); // Create a video channel
-
-
-#if defined(_WIN32)
- // These tests are disabled since they currently fail on Windows.
- // Exact reason is unkown.
- // See https://code.google.com/p/webrtc/issues/detail?id=1266.
- // TODO(mflodman): remove these APIs?
-
- //// No socket
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
-
- //EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- // tbChannel.videoChannel, 1234));
-
- //// Sender not initialized
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
- //EXPECT_EQ(0, ViE.network->SetSendDestination(
- // tbChannel.videoChannel, "127.0.0.1", 12345));
-
- //// Try to set all non-supported service types
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NETWORK_UNAVAILABLE));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_GENERAL_INFORMATION));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NOCHANGE));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NONCONFORMING));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_NETWORK_CONTROL));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICE_BESTEFFORT));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICE_CONTROLLEDLOAD));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICE_GUARANTEED));
- //EXPECT_NE(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICE_QUALITATIVE));
-
- //// Loop through valid service settings
- //bool enabled = false;
- //int serviceType = 0;
- //int overrideDSCP = 0;
-
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_FALSE(enabled);
- //EXPECT_EQ(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_TRUE(enabled);
- //EXPECT_EQ(SERVICETYPE_BESTEFFORT, serviceType);
- //EXPECT_FALSE(overrideDSCP);
-
- //EXPECT_EQ(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_CONTROLLEDLOAD));
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_TRUE(enabled);
- //EXPECT_EQ(SERVICETYPE_CONTROLLEDLOAD, serviceType);
- //EXPECT_FALSE(overrideDSCP);
-
- //EXPECT_EQ(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_GUARANTEED));
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_TRUE(enabled);
- //EXPECT_EQ(SERVICETYPE_GUARANTEED, serviceType);
- //EXPECT_FALSE(overrideDSCP);
-
- //EXPECT_EQ(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, true, SERVICETYPE_QUALITATIVE));
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_TRUE(enabled);
- //EXPECT_EQ(SERVICETYPE_QUALITATIVE, serviceType);
- //EXPECT_FALSE(overrideDSCP);
-
- //EXPECT_EQ(0, ViE.network->SetSendGQoS(
- // tbChannel.videoChannel, false, SERVICETYPE_QUALITATIVE));
- //EXPECT_EQ(0, ViE.network->GetSendGQoS(
- // tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- //EXPECT_FALSE(enabled);
-#endif
- }
- {
- //
- // MTU and packet burst
- //
- // Create a video channel
- TbVideoChannel tbChannel(ViE);
- // Invalid input
- EXPECT_NE(0, ViE.network->SetMTU(tbChannel.videoChannel, 1600));
- // Valid input
- EXPECT_EQ(0, ViE.network->SetMTU(tbChannel.videoChannel, 800));
-
- //
- // Observer and timeout
- //
- ViEAutoTestNetworkObserver vieTestObserver;
- EXPECT_EQ(0, ViE.network->RegisterObserver(
- tbChannel.videoChannel, vieTestObserver));
- EXPECT_NE(0, ViE.network->RegisterObserver(
- tbChannel.videoChannel, vieTestObserver));
- EXPECT_EQ(0, ViE.network->SetPeriodicDeadOrAliveStatus(
- tbChannel.videoChannel, true)); // No observer
- EXPECT_EQ(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
-
- EXPECT_NE(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->SetPeriodicDeadOrAliveStatus(
- tbChannel.videoChannel, true)); // No observer
-
- // Packet timout notification
- EXPECT_EQ(0, ViE.network->SetPacketTimeoutNotification(
- tbChannel.videoChannel, true, 10));
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
diff --git a/video_engine/test/auto_test/source/vie_autotest_record.cc b/video_engine/test/auto_test/source/vie_autotest_record.cc
index 15b2a83..b7b710e 100644
--- a/video_engine/test/auto_test/source/vie_autotest_record.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_record.cc
@@ -17,20 +17,22 @@
#include <stdio.h>
#include <fstream>
-#include "common_types.h"
-#include "video_engine/test/libvietest/include/tb_external_transport.h"
-#include "voice_engine/include/voe_base.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/include/vie_base.h"
-#include "video_engine/include/vie_capture.h"
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/include/vie_file.h"
-#include "video_engine/include/vie_network.h"
-#include "video_engine/include/vie_render.h"
-#include "video_engine/include/vie_rtp_rtcp.h"
-#include "voice_engine/include/voe_rtp_rtcp.h"
-#include "system_wrappers/interface/tick_util.h"
+#include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
+#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
+#include "webrtc/video_engine/include/vie_base.h"
+#include "webrtc/video_engine/include/vie_capture.h"
+#include "webrtc/video_engine/include/vie_codec.h"
+#include "webrtc/video_engine/include/vie_file.h"
+#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/video_engine/include/vie_render.h"
+#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#define VCM_RED_PAYLOAD_TYPE 96
#define VCM_ULPFEC_PAYLOAD_TYPE 97
@@ -139,6 +141,8 @@
webrtc::VoECodec* voe_codec = webrtc::VoECodec::GetInterface(voe);
webrtc::VoEAudioProcessing* voe_apm =
webrtc::VoEAudioProcessing::GetInterface(voe);
+ webrtc::VoENetwork* voe_network =
+ webrtc::VoENetwork::GetInterface(voe);
// Get the audio device for the call.
memset(audio_capture_device_name, 0, KMaxUniqueIdLength);
@@ -147,20 +151,23 @@
audio_capture_device_index, audio_playbackDeviceName,
audio_playback_device_index);
-
// Get the audio codec for the call.
memset(static_cast<void*>(&audio_codec), 0, sizeof(audio_codec));
GetAudioCodecRecord(voe_codec, audio_codec);
audio_channel = voe_base->CreateChannel();
- error = voe_base->SetSendDestination(audio_channel, audio_tx_port,
- ipAddress);
- error = voe_base->SetLocalReceiver(audio_channel, audio_rx_port);
- error = voe_hardware->SetRecordingDevice(audio_capture_device_index);
- error = voe_hardware->SetPlayoutDevice(audio_playback_device_index);
- error = voe_codec->SetSendCodec(audio_channel, audio_codec);
- error = voe_apm->SetAgcStatus(true, webrtc::kAgcDefault);
- error = voe_apm->SetNsStatus(true, webrtc::kNsHighSuppression);
+
+ webrtc::scoped_ptr<webrtc::VoiceChannelTransport> voice_channel_transport(
+ new webrtc::VoiceChannelTransport(voe_network, audio_channel));
+
+ voice_channel_transport->SetSendDestination(ipAddress, audio_tx_port);
+ voice_channel_transport->SetLocalReceiver(audio_rx_port);
+
+ voe_hardware->SetRecordingDevice(audio_capture_device_index);
+ voe_hardware->SetPlayoutDevice(audio_playback_device_index);
+ voe_codec->SetSendCodec(audio_channel, audio_codec);
+ voe_apm->SetAgcStatus(true, webrtc::kAgcDefault);
+ voe_apm->SetNsStatus(true, webrtc::kNsHighSuppression);
//
// List available capture devices, allocate and connect.
@@ -352,18 +359,17 @@
printf("ERROR in ViENetwork::GetInterface\n");
return -1;
}
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> video_channel_transport(
+ new webrtc::VideoChannelTransport(ptrViENetwork, videoChannel));
- // Setting External transport
- TbExternalTransport extTransport(*(ptrViENetwork), videoChannel, NULL);
- error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
+ error = video_channel_transport->SetSendDestination(ipAddress, rtpPort);
if (error == -1) {
- printf("ERROR in ViENetwork::SetLocalReceiver\n");
+ printf("ERROR in SetSendDestination\n");
return -1;
}
- error = ptrViENetwork->SetSendDestination(videoChannel,
- ipAddress, rtpPort);
+ error = video_channel_transport->SetLocalReceiver(rtpPort);
if (error == -1) {
- printf("ERROR in ViENetwork::SetSendDestination\n");
+ printf("ERROR in SetLocalReceiver\n");
return -1;
}
@@ -532,7 +538,6 @@
printf("ERROR: Could not release all interfaces\n");
return -1;
}
-
bool deleted = webrtc::VideoEngine::Delete(ptrViE);
if (deleted == false) {
printf("ERROR in VideoEngine::Delete\n");
@@ -564,7 +569,7 @@
}
bool GetAudioCodecRecord(webrtc::VoECodec* voe_codec,
- webrtc::CodecInst& audio_codec) {
+ webrtc::CodecInst& audio_codec) {
int error = 0;
int number_of_errors = 0;
memset(&audio_codec, 0, sizeof(webrtc::CodecInst));
diff --git a/video_engine/test/auto_test/vie_auto_test.gypi b/video_engine/test/auto_test/vie_auto_test.gypi
index 984cc75..99f8dda 100644
--- a/video_engine/test/auto_test/vie_auto_test.gypi
+++ b/video_engine/test/auto_test/vie_auto_test.gypi
@@ -20,6 +20,7 @@
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/metrics.gyp:metrics',
'<(webrtc_root)/test/test.gyp:test_support',
+ '<(webrtc_root)/test/udp_transport.gyp:udp_transport',
'<(webrtc_root)/test/libtest/libtest.gyp:libtest',
'video_engine_core',
'libvietest',
@@ -81,7 +82,6 @@
'source/vie_autotest_image_process.cc',
'source/vie_autotest_loopback.cc',
'source/vie_autotest_main.cc',
- 'source/vie_autotest_network.cc',
'source/vie_autotest_render.cc',
'source/vie_autotest_record.cc',
'source/vie_autotest_rtp_rtcp.cc',
diff --git a/video_engine/test/libvietest/include/tb_video_channel.h b/video_engine/test/libvietest/include/tb_video_channel.h
index 5c7e6f8..c3a458a 100644
--- a/video_engine/test/libvietest/include/tb_video_channel.h
+++ b/video_engine/test/libvietest/include/tb_video_channel.h
@@ -11,32 +11,39 @@
#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
-#include "video_engine/test/libvietest/include/tb_interfaces.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-class TbVideoChannel
-{
-public:
- TbVideoChannel(TbInterfaces& Engine,
- webrtc::VideoCodecType sendCodec = webrtc::kVideoCodecVP8,
- int width = 352, int height = 288, int frameRate = 30,
- int startBitrate = 300);
+namespace webrtc {
+class VideoChannelTransport;
+} // namespace webrtc
- ~TbVideoChannel(void);
+class TbVideoChannel {
+ public:
+ TbVideoChannel(TbInterfaces& Engine,
+ webrtc::VideoCodecType sendCodec = webrtc::kVideoCodecVP8,
+ int width = 352, int height = 288, int frameRate = 30,
+ int startBitrate = 300);
- void SetFrameSettings(int width, int height, int frameRate);
+ ~TbVideoChannel(void);
- void StartSend(const unsigned short rtpPort = 11000,
- const char* ipAddress = "127.0.0.1");
+ void SetFrameSettings(int width, int height, int frameRate);
- void StopSend();
+ void StartSend(const unsigned short rtpPort = 11000,
+ const char* ipAddress = "127.0.0.1");
- void StartReceive(const unsigned short rtpPort = 11000);
+ void StopSend();
- void StopReceive();
+ void StartReceive(const unsigned short rtpPort = 11000);
- int videoChannel;
-private:
- TbInterfaces& ViE;
+ void StopReceive();
+
+ int videoChannel;
+
+ private:
+ TbInterfaces& ViE;
+ webrtc::scoped_ptr<webrtc::VideoChannelTransport> channel_transport_;
};
+
#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
diff --git a/video_engine/test/libvietest/testbed/tb_video_channel.cc b/video_engine/test/libvietest/testbed/tb_video_channel.cc
index 3359f18..fe3a438 100644
--- a/video_engine/test/libvietest/testbed/tb_video_channel.cc
+++ b/video_engine/test/libvietest/testbed/tb_video_channel.cc
@@ -11,82 +11,73 @@
#include "video_engine/test/libvietest/include/tb_video_channel.h"
#include "gtest/gtest.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
TbVideoChannel::TbVideoChannel(TbInterfaces& Engine,
webrtc::VideoCodecType sendCodec, int width,
- int height, int frameRate, int startBitrate) :
- videoChannel(-1), ViE(Engine)
-{
- EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel));
+ int height, int frameRate, int startBitrate)
+ : videoChannel(-1),
+ ViE(Engine) {
+ EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel));
+ channel_transport_.reset(new webrtc::VideoChannelTransport(
+ ViE.network, videoChannel));
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- bool sendCodecSet = false;
- for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
- videoCodec.width = width;
- videoCodec.height = height;
- videoCodec.maxFramerate = frameRate;
-
- if (videoCodec.codecType == sendCodec && sendCodecSet == false)
- {
- if(videoCodec.codecType != webrtc::kVideoCodecI420 )
- {
- videoCodec.startBitrate = startBitrate;
- videoCodec.maxBitrate = startBitrate * 3;
- }
- EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
- sendCodecSet = true;
- }
- if (videoCodec.codecType == webrtc::kVideoCodecVP8)
- {
- videoCodec.width = 352;
- videoCodec.height = 288;
- }
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
- }
- EXPECT_TRUE(sendCodecSet);
-}
-
-TbVideoChannel::~TbVideoChannel(void)
-{
- EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel));
-}
-
-void TbVideoChannel::StartSend(const unsigned short rtpPort /*= 11000*/,
- const char* ipAddress /*= "127.0.0.1"*/)
-{
- EXPECT_EQ(0, ViE.network->SetSendDestination(videoChannel, ipAddress,
- rtpPort));
-
- EXPECT_EQ(0, ViE.base->StartSend(videoChannel));
-}
-
-void TbVideoChannel::SetFrameSettings(int width, int height, int frameRate)
-{
- webrtc::VideoCodec videoCodec;
- EXPECT_EQ(0, ViE.codec->GetSendCodec(videoChannel, videoCodec));
+ webrtc::VideoCodec videoCodec;
+ memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
+ bool sendCodecSet = false;
+ for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++) {
+ EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
videoCodec.width = width;
videoCodec.height = height;
videoCodec.maxFramerate = frameRate;
- EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
+ if (videoCodec.codecType == sendCodec && sendCodecSet == false) {
+ if (videoCodec.codecType != webrtc::kVideoCodecI420) {
+ videoCodec.startBitrate = startBitrate;
+ videoCodec.maxBitrate = startBitrate * 3;
+ }
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
+ sendCodecSet = true;
+ }
+ if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
+ videoCodec.width = 352;
+ videoCodec.height = 288;
+ }
EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
+ }
+ EXPECT_TRUE(sendCodecSet);
}
-void TbVideoChannel::StopSend()
-{
- EXPECT_EQ(0, ViE.base->StopSend(videoChannel));
+TbVideoChannel::~TbVideoChannel() {
+ EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel));
}
-void TbVideoChannel::StartReceive(const unsigned short rtpPort /*= 11000*/)
-{
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(videoChannel, rtpPort));
- EXPECT_EQ(0, ViE.base->StartReceive(videoChannel));
+void TbVideoChannel::StartSend(const unsigned short rtp_port,
+ const char* ip_address) {
+ EXPECT_EQ(0, channel_transport_->SetSendDestination(ip_address, rtp_port));
+ EXPECT_EQ(0, ViE.base->StartSend(videoChannel));
}
-void TbVideoChannel::StopReceive()
-{
- EXPECT_EQ(0, ViE.base->StopReceive(videoChannel));
+void TbVideoChannel::SetFrameSettings(int width, int height, int frameRate) {
+ webrtc::VideoCodec videoCodec;
+ EXPECT_EQ(0, ViE.codec->GetSendCodec(videoChannel, videoCodec));
+ videoCodec.width = width;
+ videoCodec.height = height;
+ videoCodec.maxFramerate = frameRate;
+
+ EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
+ EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
+}
+
+void TbVideoChannel::StopSend() {
+ EXPECT_EQ(0, ViE.base->StopSend(videoChannel));
+}
+
+void TbVideoChannel::StartReceive(unsigned short rtp_port) {
+ EXPECT_EQ(0, channel_transport_->SetLocalReceiver(rtp_port));
+ EXPECT_EQ(0, ViE.base->StartReceive(videoChannel));
+}
+
+void TbVideoChannel::StopReceive() {
+ EXPECT_EQ(0, ViE.base->StopReceive(videoChannel));
}
diff --git a/video_engine/video_engine_core.gypi b/video_engine/video_engine_core.gypi
index 8f28ef6..d8e2986 100644
--- a/video_engine/video_engine_core.gypi
+++ b/video_engine/video_engine_core.gypi
@@ -26,7 +26,6 @@
# ModulesShared
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
# ModulesVideo
diff --git a/video_engine/vie_base_impl.cc b/video_engine/vie_base_impl.cc
index 6348bc4..e35a216 100644
--- a/video_engine/vie_base_impl.cc
+++ b/video_engine/vie_base_impl.cc
@@ -325,14 +325,6 @@
shared_data_.SetLastError(kViEBaseInvalidChannelId);
return -1;
}
- if (vie_channel->Receiving()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d already receive.", __FUNCTION__,
- video_channel);
- shared_data_.SetLastError(kViEBaseAlreadyReceiving);
- return -1;
- }
if (vie_channel->StartReceive() != 0) {
shared_data_.SetLastError(kViEBaseUnknownError);
return -1;
diff --git a/video_engine/vie_channel.cc b/video_engine/vie_channel.cc
index 9b505a2..ae3ba9f 100644
--- a/video_engine/vie_channel.cc
+++ b/video_engine/vie_channel.cc
@@ -16,7 +16,6 @@
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/pacing/include/paced_sender.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/udp_transport/interface/udp_transport.h"
#include "modules/utility/interface/process_thread.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "modules/video_processing/main/interface/video_processing.h"
@@ -72,10 +71,6 @@
rtp_rtcp_cs_(CriticalSectionWrapper::CreateCriticalSection()),
default_rtp_rtcp_(default_rtp_rtcp),
rtp_rtcp_(NULL),
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_(*UdpTransport::Create(
- ViEModuleId(engine_id, channel_id), num_socket_threads_)),
-#endif
vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
vie_receiver_(channel_id, &vcm_, remote_bitrate_estimator),
vie_sender_(channel_id),
@@ -212,9 +207,6 @@
channel_id_, engine_id_);
// Make sure we don't get more callbacks from the RTP module.
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_.StopReceiving();
-#endif
module_process_thread_.DeRegisterModule(rtp_rtcp_.get());
module_process_thread_.DeRegisterModule(&vcm_);
module_process_thread_.DeRegisterModule(&vie_sync_);
@@ -233,10 +225,6 @@
if (decode_thread_) {
StopDecodeThread();
}
- // Release modules.
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport::Destroy(&socket_transport_);
-#endif
VideoCodingModule::Destroy(&vcm_);
}
@@ -1288,265 +1276,16 @@
}
}
-WebRtc_Word32 ViEChannel::SetLocalReceiver(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const char* ip_address) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already receiving", __FUNCTION__);
- return -1;
- }
-
- const char* multicast_ip_address = NULL;
- if (socket_transport_.InitializeReceiveSockets(&vie_receiver_, rtp_port,
- ip_address,
- multicast_ip_address,
- rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not initialize receive sockets. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetLocalReceiver(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.ReceiveSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: receive sockets not initialized", __FUNCTION__);
- return -1;
- }
-
- char multicast_ip_address[UdpTransport::kIpAddressVersion6Length];
- if (socket_transport_.ReceiveSocketInformation(ip_address, *rtp_port,
- *rtcp_port,
- multicast_ip_address) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive socket information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSendDestination(
- const char* ip_address,
- const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_UWord16 source_rtp_port,
- const WebRtc_UWord16 source_rtcp_port) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- const bool is_ipv6 = socket_transport_.IpV6Enabled();
- if (UdpTransport::IsIpAddressValid(ip_address, is_ipv6) == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Not a valid IP address: %s", __FUNCTION__, ip_address);
- return -1;
- }
- if (socket_transport_.InitializeSendSockets(ip_address, rtp_port,
- rtcp_port)!= 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not initialize send socket. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
-
- if (source_rtp_port != 0) {
- WebRtc_UWord16 receive_rtp_port = 0;
- WebRtc_UWord16 receive_rtcp_port = 0;
- if (socket_transport_.ReceiveSocketInformation(NULL, receive_rtp_port,
- receive_rtcp_port,
- NULL) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive port information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- // Initialize an extra socket only if send port differs from receive
- // port.
- if (source_rtp_port != receive_rtp_port) {
- if (socket_transport_.InitializeSourcePorts(source_rtp_port,
- source_rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not set source ports. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- }
- }
- vie_sender_.RegisterSendTransport(&socket_transport_);
-
- // Workaround to avoid SSRC colision detection in loppback tests.
- if (!is_ipv6) {
- WebRtc_UWord32 local_host_address = 0;
- const WebRtc_UWord32 current_ip_address =
- UdpTransport::InetAddrIPV4(ip_address);
-
- if ((UdpTransport::LocalHostAddress(local_host_address) == 0 &&
- local_host_address == current_ip_address) ||
- strncmp("127.0.0.1", ip_address, 9) == 0) {
- rtp_rtcp_->SetSSRC(0xFFFFFFFF);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "Running in loopback. Forcing fixed SSRC");
- }
- } else {
- char local_host_address[16];
- char current_ip_address[16];
-
- WebRtc_Word32 conv_result =
- UdpTransport::LocalHostAddressIPV6(local_host_address);
- conv_result += socket_transport_.InetPresentationToNumeric(
- 23, ip_address, current_ip_address);
- if (conv_result == 0) {
- bool local_host = true;
- for (WebRtc_Word32 i = 0; i < 16; i++) {
- if (local_host_address[i] != current_ip_address[i]) {
- local_host = false;
- break;
- }
- }
- if (!local_host) {
- local_host = true;
- for (WebRtc_Word32 i = 0; i < 15; i++) {
- if (current_ip_address[i] != 0) {
- local_host = false;
- break;
- }
- }
- if (local_host == true && current_ip_address[15] != 1) {
- local_host = false;
- }
- }
- if (local_host) {
- rtp_rtcp_->SetSSRC(0xFFFFFFFF);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Running in loopback. Forcing fixed SSRC");
- }
- }
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSendDestination(
- char* ip_address,
- WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- WebRtc_UWord16* source_rtp_port,
- WebRtc_UWord16* source_rtcp_port) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SendSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: send sockets not initialized", __FUNCTION__);
- return -1;
- }
- if (socket_transport_.SendSocketInformation(ip_address, *rtp_port,
- *rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get send socket information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- *source_rtp_port = 0;
- *source_rtcp_port = 0;
- if (socket_transport_.SourcePortsInitialized()) {
- socket_transport_.SourcePorts(*source_rtp_port, *source_rtcp_port);
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
WebRtc_Word32 ViEChannel::StartSend() {
CriticalSectionScoped cs(callback_cs_.get());
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s", __FUNCTION__);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!external_transport_) {
- if (socket_transport_.SendSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: send sockets not initialized", __FUNCTION__);
- return -1;
- }
- }
-#endif
+ if (!external_transport_) {
+ WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
+ "%s: send sockets not initialized", __FUNCTION__);
+ return -1;
+ }
rtp_rtcp_->SetSendingMediaStatus(true);
if (rtp_rtcp_->Sending()) {
@@ -1615,35 +1354,10 @@
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!external_transport_) {
- if (socket_transport_.Receiving()) {
- // Warning, don't return error.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already receiving", __FUNCTION__);
- return 0;
- }
- if (socket_transport_.ReceiveSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: receive sockets not initialized", __FUNCTION__);
- return -1;
- }
- if (socket_transport_.StartReceiving(kViENumReceiveSocketBuffers) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive socket information. Socket error:%d",
- __FUNCTION__, socket_error);
- return -1;
- }
- }
-#endif
if (StartDecodeThread() != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: could not start decoder thread", __FUNCTION__);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_.StopReceiving();
-#endif
vie_receiver_.StopReceive();
return -1;
}
@@ -1659,100 +1373,12 @@
vie_receiver_.StopReceive();
StopDecodeThread();
vcm_.ResetDecoder();
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- return 0;
- }
- }
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving() == false) {
- // Warning, don't return error
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: not receiving",
- __FUNCTION__);
- return 0;
- }
- if (socket_transport_.StopReceiving() != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__, socket_error);
- return -1;
- }
-#endif
-
return 0;
}
-bool ViEChannel::Receiving() {
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- return socket_transport_.Receiving();
-#else
- return false;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSourceInfo(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address,
- WebRtc_UWord32 ip_address_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.IpV6Enabled() &&
- ip_address_length < UdpTransport::kIpAddressVersion6Length) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IP address length is too small for IPv6", __FUNCTION__);
- return -1;
- } else if (ip_address_length < UdpTransport::kIpAddressVersion4Length) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IP address length is too small for IPv4", __FUNCTION__);
- return -1;
- }
-
- if (socket_transport_.RemoteSocketInformation(ip_address, *rtp_port,
- *rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Error getting source ports. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
WebRtc_Word32 ViEChannel::RegisterSendTransport(Transport* transport) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SendSocketsInitialized() ||
- socket_transport_.ReceiveSocketsInitialized()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: socket transport already initialized", __FUNCTION__);
- return -1;
- }
-#endif
- if (rtp_rtcp_->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Sending", __FUNCTION__);
- return -1;
- }
-
CriticalSectionScoped cs(callback_cs_.get());
if (external_transport_) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
@@ -1790,258 +1416,14 @@
WebRtc_Word32 ViEChannel::ReceivedRTPPacket(
const void* rtp_packet, const WebRtc_Word32 rtp_packet_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (!external_transport_) {
- return -1;
- }
- }
return vie_receiver_.ReceivedRTPPacket(rtp_packet, rtp_packet_length);
}
WebRtc_Word32 ViEChannel::ReceivedRTCPPacket(
- const void* rtcp_packet, const WebRtc_Word32 rtcp_packet_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (!external_transport_) {
- return -1;
- }
- }
+ const void* rtcp_packet, const WebRtc_Word32 rtcp_packet_length) {
return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
}
-WebRtc_Word32 ViEChannel::EnableIPv6() {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.IpV6Enabled()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IPv6 already enabled", __FUNCTION__);
- return -1;
- }
-
- if (socket_transport_.EnableIpV6() != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not enable IPv6. Socket error: %d", __FUNCTION__,
- socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-bool ViEChannel::IsIPv6Enabled() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return false;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- return socket_transport_.IpV6Enabled();
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return false;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSourceFilter(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const char* ip_address) {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetFilterIP(ip_address) != 0) {
- // Logging done in module.
- return -1;
- }
- if (socket_transport_.SetFilterPorts(rtp_port, rtcp_port) != 0) {
- // Logging done.
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSourceFilter(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address) const {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.FilterIP(ip_address) != 0) {
- // Logging done in module.
- return -1;
- }
- if (socket_transport_.FilterPorts(*rtp_port, *rtcp_port) != 0) {
- // Logging done in module.
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetToS(const WebRtc_Word32 DSCP,
- const bool use_set_sockOpt) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetToS(DSCP, use_set_sockOpt) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetToS(WebRtc_Word32* DSCP,
- bool* use_set_sockOpt) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.ToS(*DSCP, *use_set_sockOpt) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSendGQoS(const bool enable,
- const WebRtc_Word32 service_type,
- const WebRtc_UWord32 max_bitrate,
- const WebRtc_Word32 overrideDSCP) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetQoS(enable, service_type, max_bitrate, overrideDSCP,
- false) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSendGQoS(bool* enabled,
- WebRtc_Word32* service_type,
- WebRtc_Word32* overrideDSCP) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.QoS(*enabled, *service_type, *overrideDSCP) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
WebRtc_Word32 ViEChannel::SetMTU(WebRtc_UWord16 mtu) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
__FUNCTION__);
@@ -2152,35 +1534,6 @@
return 0;
}
-WebRtc_Word32 ViEChannel::SendUDPPacket(const WebRtc_Word8* data,
- const WebRtc_UWord32 length,
- WebRtc_Word32& transmitted_bytes,
- bool use_rtcp_socket) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- transmitted_bytes = socket_transport_.SendRaw(data, length, use_rtcp_socket);
- if (transmitted_bytes == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
WebRtc_Word32 ViEChannel::EnableColorEnhancement(bool enable) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s(enable: %d)", __FUNCTION__, enable);
@@ -2394,7 +1747,7 @@
return -1;
}
- external_transport_ = NULL;
+ external_encryption_ = NULL;
vie_receiver_.DeregisterExternalDecryption();
vie_sender_.DeregisterExternalEncryption();
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
@@ -2515,11 +1868,7 @@
CriticalSectionScoped cs(callback_cs_.get());
if (networkObserver_) {
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving() || external_transport_) {
-#else
if (external_transport_) {
-#endif
networkObserver_->PacketTimeout(channel_id_, NoPacket);
rtp_packet_timeout_ = true;
}
diff --git a/video_engine/vie_channel.h b/video_engine/vie_channel.h
index fe333aa..631fb37 100644
--- a/video_engine/vie_channel.h
+++ b/video_engine/vie_channel.h
@@ -15,7 +15,6 @@
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "modules/udp_transport/interface/udp_transport.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "system_wrappers/interface/tick_util.h"
@@ -219,27 +218,6 @@
const WebRtc_UWord32 CSRC,
const bool added);
- WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const char* ip_address);
- WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address) const;
- WebRtc_Word32 SetSendDestination(const char* ip_address,
- const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_UWord16 source_rtp_port,
- const WebRtc_UWord16 source_rtcp_port);
- WebRtc_Word32 GetSendDestination(char* ip_address,
- WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- WebRtc_UWord16* source_rtp_port,
- WebRtc_UWord16* source_rtcp_port) const;
- WebRtc_Word32 GetSourceInfo(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address,
- WebRtc_UWord32 ip_address_length);
-
WebRtc_Word32 SetRemoteSSRCType(const StreamType usage,
const uint32_t SSRC) const;
@@ -248,7 +226,6 @@
bool Sending();
WebRtc_Word32 StartReceive();
WebRtc_Word32 StopReceive();
- bool Receiving();
WebRtc_Word32 RegisterSendTransport(Transport* transport);
WebRtc_Word32 DeregisterSendTransport();
@@ -261,25 +238,6 @@
WebRtc_Word32 ReceivedRTCPPacket(const void* rtcp_packet,
const WebRtc_Word32 rtcp_packet_length);
- WebRtc_Word32 EnableIPv6();
- bool IsIPv6Enabled();
- WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const char* ip_address);
- WebRtc_Word32 GetSourceFilter(WebRtc_UWord16* rtp_port,
- WebRtc_UWord16* rtcp_port,
- char* ip_address) const;
-
- WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool use_set_sockOpt);
- WebRtc_Word32 GetToS(WebRtc_Word32* DSCP, bool* use_set_sockOpt) const;
- WebRtc_Word32 SetSendGQoS(const bool enable,
- const WebRtc_Word32 service_type,
- const WebRtc_UWord32 max_bitrate,
- const WebRtc_Word32 overrideDSCP);
- WebRtc_Word32 GetSendGQoS(bool* enabled,
- WebRtc_Word32* service_type,
- WebRtc_Word32* overrideDSCP) const;
-
// Sets the maximum transfer unit size for the network link, i.e. including
// IP, UDP and RTP headers.
WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
@@ -298,11 +256,6 @@
WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
const bool enable, const WebRtc_UWord32 sample_time_seconds);
- WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
- const WebRtc_UWord32 length,
- WebRtc_Word32& transmitted_bytes,
- bool use_rtcp_socket);
-
WebRtc_Word32 EnableColorEnhancement(bool enable);
// Gets the modules used by the channel.
@@ -384,9 +337,6 @@
scoped_ptr<RtpRtcp> rtp_rtcp_;
std::list<RtpRtcp*> simulcast_rtp_rtcp_;
std::list<RtpRtcp*> removed_rtp_rtcp_;
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport& socket_transport_;
-#endif
VideoCodingModule& vcm_;
ViEReceiver vie_receiver_;
ViESender vie_sender_;
diff --git a/video_engine/vie_impl.cc b/video_engine/vie_impl.cc
index b8066d7..6b1eb43 100644
--- a/video_engine/vie_impl.cc
+++ b/video_engine/vie_impl.cc
@@ -127,14 +127,12 @@
return false;
}
#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
ViENetworkImpl* vie_network = vie_impl;
if (vie_network->GetCount() > 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
"ViENetwork ref count: %d", vie_network->GetCount());
return false;
}
-#endif
#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
ViERenderImpl* vie_render = vie_impl;
if (vie_render->GetCount() > 0) {
diff --git a/video_engine/vie_impl.h b/video_engine/vie_impl.h
index ca5d903..b230fba 100644
--- a/video_engine/vie_impl.h
+++ b/video_engine/vie_impl.h
@@ -31,9 +31,7 @@
#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
#include "video_engine/vie_image_process_impl.h"
#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
#include "video_engine/vie_network_impl.h"
-#endif
#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
#include "video_engine/vie_render_impl.h"
#endif
@@ -63,9 +61,7 @@
#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
, public ViEImageProcessImpl
#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
, public ViENetworkImpl
-#endif
#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
, public ViERenderImpl
#endif
@@ -94,9 +90,7 @@
#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
, ViEImageProcessImpl(ViEBaseImpl::shared_data())
#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
, ViENetworkImpl(ViEBaseImpl::shared_data())
-#endif
#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
, ViERenderImpl(ViEBaseImpl::shared_data())
#endif
diff --git a/video_engine/vie_network_impl.cc b/video_engine/vie_network_impl.cc
index 378d08f..96c8be5 100644
--- a/video_engine/vie_network_impl.cc
+++ b/video_engine/vie_network_impl.cc
@@ -10,11 +10,6 @@
#include "video_engine/vie_network_impl.h"
-#include <stdio.h>
-#if (defined(WIN32_) || defined(WIN64_))
-#include <qos.h>
-#endif
-
#include "engine_configurations.h" // NOLINT
#include "system_wrappers/interface/trace.h"
#include "video_engine/include/vie_errors.h"
@@ -28,7 +23,6 @@
namespace webrtc {
ViENetwork* ViENetwork::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
if (!video_engine) {
return NULL;
}
@@ -37,9 +31,6 @@
// Increase ref count.
(*vie_networkImpl)++;
return vie_networkImpl;
-#else
- return NULL;
-#endif
}
int ViENetworkImpl::Release() {
@@ -71,140 +62,6 @@
"ViENetworkImpl::~ViENetworkImpl() Dtor");
}
-int ViENetworkImpl::SetLocalReceiver(const int video_channel,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
- __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- // The channel doesn't exists.
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->Receiving()) {
- shared_data_->SetLastError(kViENetworkAlreadyReceiving);
- return -1;
- }
- if (vie_channel->SetLocalReceiver(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetLocalReceiver(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetLocalReceiver(&rtp_port, &rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkLocalReceiverNotSet);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendDestination(const int video_channel,
- const char* ip_address,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const uint16_t source_rtp_port,
- const uint16_t source_rtcp_port) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, ip_address: %s, rtp_port: %u, rtcp_port: %u, "
- "sourceRtpPort: %u, source_rtcp_port: %u)",
- __FUNCTION__, video_channel, ip_address, rtp_port, rtcp_port,
- source_rtp_port, source_rtcp_port);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel doesn't exist", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel already sending.", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkAlreadySending);
- return -1;
- }
- if (vie_channel->SetSendDestination(ip_address, rtp_port, rtcp_port,
- source_rtp_port,
- source_rtcp_port) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSendDestination(const int video_channel,
- char* ip_address,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- uint16_t& source_rtp_port,
- uint16_t& source_rtcp_port) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSendDestination(ip_address, &rtp_port, &rtcp_port,
- &source_rtp_port,
- &source_rtcp_port) != 0) {
- shared_data_->SetLastError(kViENetworkDestinationNotSet);
- return -1;
- }
- return 0;
-}
-
int ViENetworkImpl::RegisterSendTransport(const int video_channel,
Transport& transport) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
@@ -318,340 +175,6 @@
return vie_channel->ReceivedRTCPPacket(data, length);
}
-int ViENetworkImpl::GetSourceInfo(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port, char* ip_address,
- unsigned int ip_address_length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSourceInfo(&rtp_port, &rtcp_port, ip_address,
- ip_address_length) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetLocalIP(char ip_address[64], bool ipv6) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s( ip_address, ipV6: %d)", __FUNCTION__, ipv6);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- if (!ip_address) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: No argument", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidArgument);
- return -1;
- }
-
- WebRtc_UWord8 num_socket_threads = 1;
- UdpTransport* socket_transport = UdpTransport::Create(
- ViEModuleId(shared_data_->instance_id(), -1), num_socket_threads);
-
- if (!socket_transport) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not create socket module", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
-
- char local_ip_address[64];
- if (ipv6) {
- char local_ip[16];
- if (socket_transport->LocalHostAddressIPV6(local_ip) != 0) {
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not get local IP", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- // Convert 128-bit address to character string (a:b:c:d:e:f:g:h).
- // TODO(mflodman) Change sprintf.
- sprintf(local_ip_address, // NOLINT
- "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:"
- "%.2x%.2x",
- local_ip[0], local_ip[1], local_ip[2], local_ip[3], local_ip[4],
- local_ip[5], local_ip[6], local_ip[7], local_ip[8], local_ip[9],
- local_ip[10], local_ip[11], local_ip[12], local_ip[13],
- local_ip[14], local_ip[15]);
- } else {
- WebRtc_UWord32 local_ip = 0;
- if (socket_transport->LocalHostAddress(local_ip) != 0) {
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not get local IP", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- // Convert 32-bit address to character string (x.y.z.w).
- // TODO(mflodman) Change sprintf.
- sprintf(local_ip_address, "%d.%d.%d.%d", // NOLINT
- static_cast<int>((local_ip >> 24) & 0x0ff),
- static_cast<int>((local_ip >> 16) & 0x0ff),
- static_cast<int>((local_ip >> 8) & 0x0ff),
- static_cast<int>(local_ip & 0x0ff));
- }
- strncpy(ip_address, local_ip_address, sizeof(local_ip_address));
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: local ip = %s", __FUNCTION__, local_ip_address);
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: not available for external transport", __FUNCTION__);
-
- return -1;
-#endif
-}
-
-int ViENetworkImpl::EnableIPv6(int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->EnableIPv6() != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-bool ViENetworkImpl::IsIPv6Enabled(int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return false;
- }
- return vie_channel->IsIPv6Enabled();
-}
-
-int ViENetworkImpl::SetSourceFilter(const int video_channel,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
- __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetSourceFilter(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSourceFilter(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSourceFilter(&rtp_port, &rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendToS(const int video_channel, const int DSCP,
- const bool use_set_sockOpt = false) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, DSCP: %d, use_set_sockOpt: %d)", __FUNCTION__,
- video_channel, DSCP, use_set_sockOpt);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
-
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- " force use_set_sockopt=true since there is no alternative"
- " implementation");
- if (vie_channel->SetToS(DSCP, true) != 0) {
-#else
- if (vie_channel->SetToS(DSCP, use_set_sockOpt) != 0) {
-#endif
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSendToS(const int video_channel,
- int& DSCP, // NOLINT
- bool& use_set_sockOpt) { // NOLINT
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetToS(&DSCP, &use_set_sockOpt) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendGQoS(const int video_channel, const bool enable,
- const int service_type,
- const int overrideDSCP) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, service_type: %d, "
- "overrideDSCP: %d)", __FUNCTION__, video_channel, enable,
- service_type, overrideDSCP);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
-#if (defined(WIN32_) || defined(WIN64_))
- // Sanity check. We might crash if testing and relying on an OS socket error.
- if (enable &&
- (service_type != SERVICETYPE_BESTEFFORT) &&
- (service_type != SERVICETYPE_CONTROLLEDLOAD) &&
- (service_type != SERVICETYPE_GUARANTEED) &&
- (service_type != SERVICETYPE_QUALITATIVE)) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: service type %d not supported", __FUNCTION__,
- video_channel, service_type);
- shared_data_->SetLastError(kViENetworkServiceTypeNotSupported);
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- VideoCodec video_codec;
- if (vie_encoder->GetEncoder(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get max bitrate for the channel",
- __FUNCTION__);
- shared_data_->SetLastError(kViENetworkSendCodecNotSet);
- return -1;
- }
- if (vie_channel->SetSendGQoS(enable, service_type, video_codec.maxBitrate,
- overrideDSCP) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Not supported", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkNotSupported);
- return -1;
-#endif
-}
-
-int ViENetworkImpl::GetSendGQoS(const int video_channel,
- bool& enabled, // NOLINT
- int& service_type, // NOLINT
- int& overrideDSCP) { // NOLINT
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSendGQoS(&enabled, &service_type, &overrideDSCP) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
int ViENetworkImpl::SetMTU(int video_channel, unsigned int mtu) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
@@ -766,38 +289,4 @@
return 0;
}
-int ViENetworkImpl::SendUDPPacket(const int video_channel, const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket = false) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, data: -, length: %d, transmitter_bytes: -, "
- "useRtcpSocket: %d)", __FUNCTION__, video_channel, length,
- use_rtcp_socket);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SendUDPPacket((const WebRtc_Word8*) data, length,
- (WebRtc_Word32&) transmitted_bytes,
- use_rtcp_socket) < 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
} // namespace webrtc
diff --git a/video_engine/vie_network_impl.h b/video_engine/vie_network_impl.h
index 56832df..6e992eb 100644
--- a/video_engine/vie_network_impl.h
+++ b/video_engine/vie_network_impl.h
@@ -25,26 +25,6 @@
public:
// Implements ViENetwork.
virtual int Release();
- virtual int SetLocalReceiver(const int video_channel,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const char* ip_address);
- virtual int GetLocalReceiver(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- char* ip_address);
- virtual int SetSendDestination(const int video_channel,
- const char* ip_address,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const uint16_t source_rtp_port,
- const uint16_t source_rtcp_port);
- virtual int GetSendDestination(const int video_channel,
- char* ip_address,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- uint16_t& source_rtp_port,
- uint16_t& source_rtcp_port);
virtual int RegisterSendTransport(const int video_channel,
Transport& transport);
virtual int DeregisterSendTransport(const int video_channel);
@@ -54,36 +34,6 @@
virtual int ReceivedRTCPPacket(const int video_channel,
const void* data,
const int length);
- virtual int GetSourceInfo(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- char* ip_address,
- unsigned int ip_address_length);
- virtual int GetLocalIP(char ip_address[64], bool ipv6);
- virtual int EnableIPv6(int video_channel);
- virtual bool IsIPv6Enabled(int video_channel);
- virtual int SetSourceFilter(const int video_channel,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const char* ip_address);
- virtual int GetSourceFilter(const int video_channel,
- uint16_t& rtp_port,
- uint16_t& rtcp_port,
- char* ip_address);
- virtual int SetSendToS(const int video_channel,
- const int DSCP,
- const bool use_set_sockOpt);
- virtual int GetSendToS(const int video_channel,
- int& DSCP,
- bool& use_set_sockOpt);
- virtual int SetSendGQoS(const int video_channel,
- const bool enable,
- const int service_type,
- const int overrideDSCP);
- virtual int GetSendGQoS(const int video_channel,
- bool& enabled,
- int& service_type,
- int& overrideDSCP);
virtual int SetMTU(int video_channel, unsigned int mtu);
virtual int SetPacketTimeoutNotification(const int video_channel,
bool enable,
@@ -95,11 +45,6 @@
const int video_channel,
const bool enable,
const unsigned int sample_time_seconds);
- virtual int SendUDPPacket(const int video_channel,
- const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket);
protected:
explicit ViENetworkImpl(ViESharedData* shared_data);
diff --git a/video_engine/vie_receiver.cc b/video_engine/vie_receiver.cc
index 43aa566..57c9d16 100644
--- a/video_engine/vie_receiver.cc
+++ b/video_engine/vie_receiver.cc
@@ -87,20 +87,6 @@
}
}
-void ViEReceiver::IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
- const WebRtc_Word32 rtp_packet_length,
- const char* from_ip,
- const WebRtc_UWord16 from_port) {
- InsertRTPPacket(rtp_packet, rtp_packet_length);
-}
-
-void ViEReceiver::IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
- const WebRtc_Word32 rtcp_packet_length,
- const char* from_ip,
- const WebRtc_UWord16 from_port) {
- InsertRTCPPacket(rtcp_packet, rtcp_packet_length);
-}
-
int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
int rtp_packet_length) {
if (!receiving_) {
diff --git a/video_engine/vie_receiver.h b/video_engine/vie_receiver.h
index 4f43fc9..dda2f29 100644
--- a/video_engine/vie_receiver.h
+++ b/video_engine/vie_receiver.h
@@ -15,7 +15,6 @@
#include "engine_configurations.h" // NOLINT
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "modules/udp_transport/interface/udp_transport.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "typedefs.h" // NOLINT
#include "video_engine/vie_defines.h"
@@ -29,7 +28,7 @@
class RtpRtcp;
class VideoCodingModule;
-class ViEReceiver : public UdpTransportData, public RtpData {
+class ViEReceiver : public RtpData {
public:
ViEReceiver(const int32_t channel_id, VideoCodingModule* module_vcm,
RemoteBitrateEstimator* remote_bitrate_estimator);
@@ -48,16 +47,6 @@
int StartRTPDump(const char file_nameUTF8[1024]);
int StopRTPDump();
- // Implements UdpTransportData.
- virtual void IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
- const WebRtc_Word32 rtp_packet_length,
- const char* from_ip,
- const WebRtc_UWord16 from_port);
- virtual void IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
- const WebRtc_Word32 rtcp_packet_length,
- const char* from_ip,
- const WebRtc_UWord16 from_port);
-
// Receives packets from external transport.
int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length);
int ReceivedRTCPPacket(const void* rtcp_packet, int rtcp_packet_length);
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 1eabee8..7691bf0 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -332,163 +332,6 @@
}
void
-Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IncomingRTPPacket(rtpPacketLength=%d,"
- " fromIP=%s, fromPort=%u)",
- rtpPacketLength, fromIP, fromPort);
-
- // Store playout timestamp for the received RTP packet
- // to be used for upcoming delay estimations
- WebRtc_UWord32 playoutTimestamp(0);
- if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
- {
- _playoutTimeStampRTP = playoutTimestamp;
- }
-
- WebRtc_UWord8* rtpBufferPtr = (WebRtc_UWord8*)incomingRtpPacket;
- WebRtc_Word32 rtpBufferLength = rtpPacketLength;
-
- // SRTP or External decryption
- if (_decrypting)
- {
- CriticalSectionScoped cs(&_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_decryptionRTPBufferPtr)
- {
- // Allocate memory for decryption buffer one time only
- _decryptionRTPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform decryption (SRTP or external)
- WebRtc_Word32 decryptedBufferLength = 0;
- _encryptionPtr->decrypt(_channelId,
- rtpBufferPtr,
- _decryptionRTPBufferPtr,
- rtpBufferLength,
- (int*)&decryptedBufferLength);
- if (decryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_DECRYPTION_FAILED, kTraceError,
- "Channel::IncomingRTPPacket() decryption failed");
- return;
- }
-
- // Replace default data buffer with decrypted buffer
- rtpBufferPtr = _decryptionRTPBufferPtr;
- rtpBufferLength = decryptedBufferLength;
- }
- }
-
- // Dump the RTP packet to a file (if RTP dump is enabled).
- if (_rtpDumpIn.DumpPacket(rtpBufferPtr,
- (WebRtc_UWord16)rtpBufferLength) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTP dump to input file failed");
- }
-
- // Deliver RTP packet to RTP/RTCP module for parsing
- // The packet will be pushed back to the channel thru the
- // OnReceivedPayloadData callback so we don't push it to the ACM here
- if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtpBufferPtr,
- (WebRtc_UWord16)rtpBufferLength) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "Channel::IncomingRTPPacket() RTP packet is invalid");
- return;
- }
-}
-
-void
-Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IncomingRTCPPacket(rtcpPacketLength=%d, fromIP=%s,"
- " fromPort=%u)",
- rtcpPacketLength, fromIP, fromPort);
-
- // Temporary buffer pointer and size for decryption
- WebRtc_UWord8* rtcpBufferPtr = (WebRtc_UWord8*)incomingRtcpPacket;
- WebRtc_Word32 rtcpBufferLength = rtcpPacketLength;
-
- // Store playout timestamp for the received RTCP packet
- // which will be read by the GetRemoteRTCPData API
- WebRtc_UWord32 playoutTimestamp(0);
- if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
- {
- _playoutTimeStampRTCP = playoutTimestamp;
- }
-
- // SRTP or External decryption
- if (_decrypting)
- {
- CriticalSectionScoped cs(&_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_decryptionRTCPBufferPtr)
- {
- // Allocate memory for decryption buffer one time only
- _decryptionRTCPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform decryption (SRTP or external).
- WebRtc_Word32 decryptedBufferLength = 0;
- _encryptionPtr->decrypt_rtcp(_channelId,
- rtcpBufferPtr,
- _decryptionRTCPBufferPtr,
- rtcpBufferLength,
- (int*)&decryptedBufferLength);
- if (decryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_DECRYPTION_FAILED, kTraceError,
- "Channel::IncomingRTCPPacket() decryption failed");
- return;
- }
-
- // Replace default data buffer with decrypted buffer
- rtcpBufferPtr = _decryptionRTCPBufferPtr;
- rtcpBufferLength = decryptedBufferLength;
- }
- }
-
- // Dump the RTCP packet to a file (if RTP dump is enabled).
- if (_rtpDumpIn.DumpPacket(rtcpBufferPtr,
- (WebRtc_UWord16)rtcpBufferLength) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTCP dump to input file failed");
- }
-
- // Deliver RTCP packet to RTP/RTCP module for parsing
- if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtcpBufferPtr,
- (WebRtc_UWord16)rtcpBufferLength) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "Channel::IncomingRTPPacket() RTCP packet is invalid");
- return;
- }
-}
-
-void
Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id,
const WebRtc_UWord8 event,
const WebRtc_UWord16 lengthMs,
@@ -1028,11 +871,6 @@
_channelId(channelId),
_audioCodingModule(*AudioCodingModule::Create(
VoEModuleId(instanceId, channelId))),
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- _numSocketThreads(KNumSocketThreads),
- _socketTransportModule(*UdpTransport::Create(
- VoEModuleId(instanceId, channelId), _numSocketThreads)),
-#endif
#ifdef WEBRTC_SRTP
_srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId,
channelId))),
@@ -1161,18 +999,6 @@
DeRegisterExternalMediaProcessing(kRecordingPerChannel);
}
StopSend();
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- StopReceiving();
- // De-register packet callback to ensure we're not in a callback when
- // deleting channel state, avoids race condition and deadlock.
- if (_socketTransportModule.InitializeReceiveSockets(NULL, 0, NULL, NULL, 0)
- != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "~Channel() failed to de-register receive callback");
- }
-#endif
StopPlayout();
{
@@ -1219,15 +1045,6 @@
" (Audio coding module)");
}
// De-register modules in process thread
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule)
- == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to deregister socket module");
- }
-#endif
if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice,
@@ -1236,10 +1053,6 @@
}
// Destroy modules
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport::Destroy(
- &_socketTransportModule);
-#endif
AudioCodingModule::Destroy(&_audioCodingModule);
#ifdef WEBRTC_SRTP
SrtpModule::DestroySrtpModule(&_srtpModule);
@@ -1284,12 +1097,7 @@
const bool processThreadFail =
((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) ||
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- (_moduleProcessThreadPtr->RegisterModule(
- &_socketTransportModule) != 0));
-#else
false);
-#endif
if (processThreadFail)
{
_engineStatisticsPtr->SetLastError(
@@ -1423,17 +1231,6 @@
}
#endif
}
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- // Ensure that the WebRtcSocketTransport implementation is used as
- // Transport on the sending side
- {
- // A lock is needed here since users can call
- // RegisterExternalTransport() at the same time.
- CriticalSectionScoped cs(&_callbackCritSect);
- _transportPtr = &_socketTransportModule;
- }
-#endif
-
// Initialize the far end AP module
// Using 8 kHz as initial Fs, the same as in transmission. Might be
// changed at the first receiving audio.
@@ -1662,25 +1459,6 @@
// If external transport is used, we will only initialize/set the variables
// after this section, since we are not using the WebRtc transport but
// still need to keep track of e.g. if we are receiving.
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_externalTransport)
- {
- if (!_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKETS_NOT_INITED, kTraceError,
- "StartReceive() must set local receiver first");
- return -1;
- }
- if (_socketTransportModule.StartReceiving(KNumberOfSocketBuffers) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "StartReceiving() failed to start receiving");
- return -1;
- }
- }
-#endif
_receiving = true;
_numberOfDiscardedPackets = 0;
return 0;
@@ -1695,20 +1473,6 @@
{
return 0;
}
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_externalTransport &&
- _socketTransportModule.ReceiveSocketsInitialized())
- {
- if (_socketTransportModule.StopReceiving() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "StopReceiving() failed to stop receiving.");
- return -1;
- }
- }
-#endif
// Recover DTMF detection status.
WebRtc_Word32 ret = _rtpRtcpModule->SetTelephoneEventForwardToDecoder(true);
if (ret != 0) {
@@ -1721,310 +1485,6 @@
return 0;
}
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::SetLocalReceiver(const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort,
- const char ipAddr[64],
- const char multicastIpAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetLocalReceiver()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetLocalReceiver() conflict with external transport");
- return -1;
- }
-
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_SENDING, kTraceError,
- "SetLocalReceiver() already sending");
- return -1;
- }
- if (_receiving)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_LISTENING, kTraceError,
- "SetLocalReceiver() already receiving");
- return -1;
- }
-
- if (_socketTransportModule.InitializeReceiveSockets(this,
- rtpPort,
- ipAddr,
- multicastIpAddr,
- rtcpPort) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetLocalReceiver() invalid IP address");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetLocalReceiver() invalid socket");
- break;
- case UdpTransport::kPortInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid port");
- break;
- case UdpTransport::kFailedToBindPort:
- _engineStatisticsPtr->SetLastError(
- VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED, kTraceError,
- "SetLocalReceiver() binding failed");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetLocalReceiver() undefined socket error");
- break;
- }
- return -1;
- }
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetLocalReceiver(int& port, int& RTCPport, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetLocalReceiver()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetLocalReceiver() conflict with external transport");
- return -1;
- }
-
- char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
- WebRtc_UWord16 rtpPort(0);
- WebRtc_UWord16 rtcpPort(0);
- char multicastIpAddr[UdpTransport::kIpAddressVersion6Length] = {0};
-
- // Acquire socket information from the socket module
- if (_socketTransportModule.ReceiveSocketInformation(ipAddrTmp,
- rtpPort,
- rtcpPort,
- multicastIpAddr) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_GET_SOCKET_INFO, kTraceError,
- "GetLocalReceiver() unable to retrieve socket information");
- return -1;
- }
-
- // Deliver valid results to the user
- port = static_cast<int> (rtpPort);
- RTCPport = static_cast<int> (rtcpPort);
- if (ipAddr != NULL)
- {
- strcpy(ipAddr, ipAddrTmp);
- }
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::SetSendDestination(const WebRtc_UWord16 rtpPort,
- const char ipAddr[64],
- const int sourcePort,
- const WebRtc_UWord16 rtcpPort)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendDestination()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendDestination() conflict with external transport");
- return -1;
- }
-
- // Initialize ports and IP address for the remote (destination) side.
- // By default, the sockets used for receiving are used for transmission as
- // well, hence the source ports for outgoing packets are the same as the
- // receiving ports specified in SetLocalReceiver.
- // If an extra send socket has been created, it will be utilized until a
- // new source port is specified or until the channel has been deleted and
- // recreated. If no socket exists, sockets will be created when the first
- // RTP and RTCP packets shall be transmitted (see e.g.
- // UdpTransportImpl::SendPacket()).
- //
- // NOTE: this function does not require that sockets exists; all it does is
- // to build send structures to be used with the sockets when they exist.
- // It is therefore possible to call this method before SetLocalReceiver.
- // However, sockets must exist if a multi-cast address is given as input.
-
- // Build send structures and enable QoS (if enabled and supported)
- if (_socketTransportModule.InitializeSendSockets(
- ipAddr, rtpPort, rtcpPort) != UdpTransport::kNoSocketError)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSendDestination() invalid IP address 1");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() invalid socket 1");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_GQOS_ERROR, kTraceError,
- "SetSendDestination() failed to set QoS");
- break;
- case UdpTransport::kMulticastAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_MULTICAST_ADDRESS, kTraceError,
- "SetSendDestination() invalid multicast address");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() undefined socket error 1");
- break;
- }
- return -1;
- }
-
- // Check if the user has specified a non-default source port different from
- // the local receive port.
- // If so, an extra local socket will be created unless the source port is
- // not unique.
- if (sourcePort != kVoEDefault)
- {
- WebRtc_UWord16 receiverRtpPort(0);
- WebRtc_UWord16 rtcpNA(0);
- if (_socketTransportModule.ReceiveSocketInformation(NULL,
- receiverRtpPort,
- rtcpNA,
- NULL) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_GET_SOCKET_INFO, kTraceError,
- "SetSendDestination() failed to retrieve socket information");
- return -1;
- }
-
- WebRtc_UWord16 sourcePortUW16 =
- static_cast<WebRtc_UWord16> (sourcePort);
-
- // An extra socket will only be created if the specified source port
- // differs from the local receive port.
- if (sourcePortUW16 != receiverRtpPort)
- {
- // Initialize extra local socket to get a different source port
- // than the local
- // receiver port. Always use default source for RTCP.
- // Note that, this calls UdpTransport::CloseSendSockets().
- if (_socketTransportModule.InitializeSourcePorts(
- sourcePortUW16,
- sourcePortUW16+1) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSendDestination() invalid IP address 2");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() invalid socket 2");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() undefined socket error 2");
- break;
- }
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "SetSendDestination() extra local socket is created"
- " to facilitate unique source port");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "SetSendDestination() sourcePort equals the local"
- " receive port => no extra socket is created");
- }
- }
-
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetSendDestination(int& port,
- char ipAddr[64],
- int& sourcePort,
- int& RTCPport)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendDestination()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendDestination() conflict with external transport");
- return -1;
- }
-
- char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
- WebRtc_UWord16 rtpPort(0);
- WebRtc_UWord16 rtcpPort(0);
- WebRtc_UWord16 rtpSourcePort(0);
- WebRtc_UWord16 rtcpSourcePort(0);
-
- // Acquire sending socket information from the socket module
- _socketTransportModule.SendSocketInformation(ipAddrTmp, rtpPort, rtcpPort);
- _socketTransportModule.SourcePorts(rtpSourcePort, rtcpSourcePort);
-
- // Deliver valid results to the user
- port = static_cast<int> (rtpPort);
- RTCPport = static_cast<int> (rtcpPort);
- sourcePort = static_cast<int> (rtpSourcePort);
- if (ipAddr != NULL)
- {
- strcpy(ipAddr, ipAddrTmp);
- }
-
- return 0;
-}
-#endif
-
-
WebRtc_Word32
Channel::SetNetEQPlayoutMode(NetEqModes mode)
{
@@ -2628,24 +2088,6 @@
CriticalSectionScoped cs(&_callbackCritSect);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- // Sanity checks for default (non external transport) to avoid conflict with
- // WebRtc sockets.
- if (_socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(VE_SEND_SOCKETS_CONFLICT,
- kTraceError,
- "RegisterExternalTransport() send sockets already initialized");
- return -1;
- }
- if (_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(VE_RECEIVE_SOCKETS_CONFLICT,
- kTraceError,
- "RegisterExternalTransport() receive sockets already initialized");
- return -1;
- }
-#endif
if (_externalTransport)
{
_engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION,
@@ -2675,396 +2117,81 @@
return 0;
}
_externalTransport = false;
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
_transportPtr = NULL;
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
"DeRegisterExternalTransport() all transport is disabled");
-#else
- _transportPtr = &_socketTransportModule;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "DeRegisterExternalTransport() internal Transport is enabled");
-#endif
return 0;
}
WebRtc_Word32
Channel::ReceivedRTPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ReceivedRTPPacket()");
- const char dummyIP[] = "127.0.0.1";
- IncomingRTPPacket(data, length, dummyIP, 0);
- return 0;
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ReceivedRTPPacket()");
+
+ // Store playout timestamp for the received RTP packet
+ // to be used for upcoming delay estimations
+ WebRtc_UWord32 playoutTimestamp(0);
+ if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
+ {
+ _playoutTimeStampRTP = playoutTimestamp;
+ }
+ // Dump the RTP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpIn.DumpPacket((const WebRtc_UWord8*)data,
+ (WebRtc_UWord16)length) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTP dump to input file failed");
+ }
+
+ // Deliver RTP packet to RTP/RTCP module for parsing
+ // The packet will be pushed back to the channel thru the
+ // OnReceivedPayloadData callback so we don't push it to the ACM here
+ if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data,
+ (WebRtc_UWord16)length) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
+ "Channel::IncomingRTPPacket() RTP packet is invalid");
+ }
+ return 0;
}
WebRtc_Word32
Channel::ReceivedRTCPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ReceivedRTCPPacket()");
- const char dummyIP[] = "127.0.0.1";
- IncomingRTCPPacket(data, length, dummyIP, 0);
- return 0;
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+ "Channel::ReceivedRTCPPacket()");
+
+ // Store playout timestamp for the received RTCP packet
+ // which will be read by the GetRemoteRTCPData API
+ WebRtc_UWord32 playoutTimestamp(0);
+ if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
+ {
+ _playoutTimeStampRTCP = playoutTimestamp;
+ }
+
+ // Dump the RTCP packet to a file (if RTP dump is enabled).
+ if (_rtpDumpIn.DumpPacket((const WebRtc_UWord8*)data,
+ (WebRtc_UWord16)length) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId,_channelId),
+ "Channel::SendPacket() RTCP dump to input file failed");
+ }
+
+ // Deliver RTCP packet to RTP/RTCP module for parsing
+ if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data,
+ (WebRtc_UWord16)length) == -1)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
+ "Channel::IncomingRTPPacket() RTCP packet is invalid");
+ }
+ return 0;
}
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSourceInfo()");
-
- WebRtc_UWord16 rtpPortModule;
- WebRtc_UWord16 rtcpPortModule;
- char ipaddr[UdpTransport::kIpAddressVersion6Length] = {0};
-
- if (_socketTransportModule.RemoteSocketInformation(ipaddr,
- rtpPortModule,
- rtcpPortModule) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSourceInfo() failed to retrieve remote socket information");
- return -1;
- }
- strcpy(ipAddr, ipaddr);
- rtpPort = rtpPortModule;
- rtcpPort = rtcpPortModule;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSourceInfo() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
- rtpPort, rtcpPort, ipAddr);
- return 0;
-}
-
-WebRtc_Word32
-Channel::EnableIPv6()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EnableIPv6()");
- if (_socketTransportModule.ReceiveSocketsInitialized() ||
- _socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "EnableIPv6() socket layer is already initialized");
- return -1;
- }
- if (_socketTransportModule.EnableIpV6() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "EnableIPv6() failed to enable IPv6");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d", lastError);
- return -1;
- }
- return 0;
-}
-
-bool
-Channel::IPv6IsEnabled() const
-{
- bool isEnabled = _socketTransportModule.IpV6Enabled();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "IPv6IsEnabled() => %d", isEnabled);
- return isEnabled;
-}
-
-WebRtc_Word32
-Channel::SetSourceFilter(int rtpPort, int rtcpPort, const char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSourceFilter()");
- if (_socketTransportModule.SetFilterPorts(
- static_cast<WebRtc_UWord16>(rtpPort),
- static_cast<WebRtc_UWord16>(rtcpPort)) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "SetSourceFilter() failed to set filter ports");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d",
- lastError);
- return -1;
- }
- const char* filterIpAddress = ipAddr;
- if (_socketTransportModule.SetFilterIP(filterIpAddress) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSourceFilter() failed to set filter IP address");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d", lastError);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSourceFilter()");
- WebRtc_UWord16 rtpFilterPort(0);
- WebRtc_UWord16 rtcpFilterPort(0);
- if (_socketTransportModule.FilterPorts(rtpFilterPort, rtcpFilterPort) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "GetSourceFilter() failed to retrieve filter ports");
- }
- char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
- if (_socketTransportModule.FilterIP(ipAddrTmp) != 0)
- {
- // no filter has been configured (not seen as an error)
- memset(ipAddrTmp,
- 0, UdpTransport::kIpAddressVersion6Length);
- }
- rtpPort = static_cast<int> (rtpFilterPort);
- rtcpPort = static_cast<int> (rtcpFilterPort);
- strcpy(ipAddr, ipAddrTmp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSourceFilter() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
- rtpPort, rtcpPort, ipAddr);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetSendTOS(int DSCP, int priority, bool useSetSockopt)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendTOS(DSCP=%d, useSetSockopt=%d)",
- DSCP, (int)useSetSockopt);
-
- // Set TOS value and possibly try to force usage of setsockopt()
- if (_socketTransportModule.SetToS(DSCP, useSetSockopt) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kTosError:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() TOS error");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendTOS() GQOS error");
- break;
- case UdpTransport::kTosInvalid:
- // can't switch SetSockOpt method without disabling TOS first, or
- // SetSockopt() call failed
- _engineStatisticsPtr->SetLastError(VE_TOS_INVALID, kTraceError,
- "SetSendTOS() invalid TOS");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
- "SetSendTOS() invalid Socket");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() TOS error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport => lastError = %d",
- lastSockError);
- return -1;
- }
-
- // Set priority (PCP) value, -1 means don't change
- if (-1 != priority)
- {
- if (_socketTransportModule.SetPCP(priority) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kPcpError:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() PCP error");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendTOS() GQOS conflict");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendTOS() invalid Socket");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() PCP error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "UdpTransport => lastError = %d",
- lastSockError);
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendTOS(DSCP=?, useSetSockopt=?)");
- WebRtc_Word32 dscp(0), prio(0);
- bool setSockopt(false);
- if (_socketTransportModule.ToS(dscp, setSockopt) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSendTOS() failed to get TOS info");
- return -1;
- }
- if (_socketTransportModule.PCP(prio) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSendTOS() failed to get PCP info");
- return -1;
- }
- DSCP = static_cast<int> (dscp);
- priority = static_cast<int> (prio);
- useSetSockopt = setSockopt;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSendTOS() => DSCP=%d, priority=%d, useSetSockopt=%d",
- DSCP, priority, (int)useSetSockopt);
- return 0;
-}
-
-#if defined(_WIN32)
-WebRtc_Word32
-Channel::SetSendGQoS(bool enable, int serviceType, int overrideDSCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendGQoS(enable=%d, serviceType=%d, "
- "overrideDSCP=%d)",
- (int)enable, serviceType, overrideDSCP);
- if(!_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKETS_NOT_INITED, kTraceError,
- "SetSendGQoS() GQoS state must be set after sockets are created");
- return -1;
- }
- if(!_socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_DESTINATION_NOT_INITED, kTraceError,
- "SetSendGQoS() GQoS state must be set after sending side is "
- "initialized");
- return -1;
- }
- if (enable &&
- (serviceType != SERVICETYPE_BESTEFFORT) &&
- (serviceType != SERVICETYPE_CONTROLLEDLOAD) &&
- (serviceType != SERVICETYPE_GUARANTEED) &&
- (serviceType != SERVICETYPE_QUALITATIVE))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendGQoS() Invalid service type");
- return -1;
- }
- if (enable && ((overrideDSCP < 0) || (overrideDSCP > 63)))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendGQoS() Invalid overrideDSCP value");
- return -1;
- }
-
- // Avoid GQoS/ToS conflict when user wants to override the default DSCP
- // mapping
- bool QoS(false);
- WebRtc_Word32 sType(0);
- WebRtc_Word32 ovrDSCP(0);
- if (_socketTransportModule.QoS(QoS, sType, ovrDSCP))
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "SetSendGQoS() failed to get QOS info");
- return -1;
- }
- if (QoS && ovrDSCP == 0 && overrideDSCP != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendGQoS() QOS is already enabled and overrideDSCP differs,"
- " not allowed");
- return -1;
- }
- const WebRtc_Word32 maxBitrate(0);
- if (_socketTransportModule.SetQoS(enable,
- static_cast<WebRtc_Word32>(serviceType),
- maxBitrate,
- static_cast<WebRtc_Word32>(overrideDSCP),
- true))
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(VE_GQOS_ERROR, kTraceError,
- "SetSendGQoS() QOS error");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
- "SetSendGQoS() Socket error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport() => lastError = %d",
- lastSockError);
- return -1;
- }
- return 0;
-}
-#endif
-
-#if defined(_WIN32)
-WebRtc_Word32
-Channel::GetSendGQoS(bool &enabled, int &serviceType, int &overrideDSCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendGQoS(enable=?, serviceType=?, "
- "overrideDSCP=?)");
-
- bool QoS(false);
- WebRtc_Word32 serviceTypeModule(0);
- WebRtc_Word32 overrideDSCPModule(0);
- _socketTransportModule.QoS(QoS, serviceTypeModule, overrideDSCPModule);
-
- enabled = QoS;
- serviceType = static_cast<int> (serviceTypeModule);
- overrideDSCP = static_cast<int> (overrideDSCPModule);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSendGQoS() => enabled=%d, serviceType=%d, overrideDSCP=%d",
- (int)enabled, serviceType, overrideDSCP);
- return 0;
-}
-#endif
-#endif
-
WebRtc_Word32
Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds)
{
@@ -3200,65 +2327,6 @@
return 0;
}
-WebRtc_Word32
-Channel::SendUDPPacket(const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendUDPPacket()");
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SendUDPPacket() external transport is enabled");
- return -1;
- }
- if (useRtcpSocket && !_rtpRtcpModule->RTCP())
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTCP_ERROR, kTraceError,
- "SendUDPPacket() RTCP is disabled");
- return -1;
- }
- if (!_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NOT_SENDING, kTraceError,
- "SendUDPPacket() not sending");
- return -1;
- }
-
- char* dataC = new char[length];
- if (NULL == dataC)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NO_MEMORY, kTraceError,
- "SendUDPPacket() memory allocation failed");
- return -1;
- }
- memcpy(dataC, data, length);
-
- transmittedBytes = SendPacketRaw(dataC, length, useRtcpSocket);
-
- delete [] dataC;
- dataC = NULL;
-
- if (transmittedBytes <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SEND_ERROR, kTraceError,
- "SendUDPPacket() transmission failed");
- transmittedBytes = 0;
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "SendUDPPacket() => transmittedBytes=%d", transmittedBytes);
- return 0;
-}
-
-
int Channel::StartPlayingFileLocally(const char* fileName,
const bool loop,
const FileFormats format,
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index f1c5809..cdb0790 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -27,9 +27,6 @@
#include "webrtc/voice_engine/shared_data.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-#include "webrtc/modules/udp_transport/interface/udp_transport.h"
-#endif
#ifdef WEBRTC_SRTP
#include "SrtpModule.h"
#endif
@@ -65,9 +62,6 @@
public RtpData,
public RtpFeedback,
public RtcpFeedback,
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- public UdpTransportData, // receiving packet from sockets
-#endif
public FileCallback, // receiving notification from file player & recorder
public Transport,
public RtpAudioFeedback,
@@ -181,9 +175,6 @@
int sampleTimeSeconds);
WebRtc_Word32 GetPeriodicDeadOrAliveStatus(bool& enabled,
int& sampleTimeSeconds);
- WebRtc_Word32 SendUDPPacket(const void* data, unsigned int length,
- int& transmittedBytes, bool useRtcpSocket);
-
// VoEFile
int StartPlayingFileLocally(const char* fileName, const bool loop,
const FileFormats format,
@@ -413,18 +404,6 @@
const WebRtc_UWord8 volume);
public:
- // From UdpTransportData in the Socket Transport module
- void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort);
-
- void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort);
-
-public:
// From Transport (called by the RTP/RTCP module)
int SendPacket(int /*channel*/, const void *data, int len);
int SendRTCPPacket(int /*channel*/, const void *data, int len);
@@ -497,16 +476,6 @@
{
return _outputAudioLevel.Level();
}
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- bool SendSocketsInitialized() const
- {
- return _socketTransportModule.SendSocketsInitialized();
- }
- bool ReceiveSocketsInitialized() const
- {
- return _socketTransportModule.ReceiveSocketsInitialized();
- }
-#endif
WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame);
WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);
WebRtc_UWord32 EncodeAndSend();
@@ -535,10 +504,6 @@
private:
scoped_ptr<RtpRtcp> _rtpRtcpModule;
AudioCodingModule& _audioCodingModule;
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_UWord8 _numSocketThreads;
- UdpTransport& _socketTransportModule;
-#endif
#ifdef WEBRTC_SRTP
SrtpModule& _srtpModule;
#endif
diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h
index 84cd4fc..70cbab1 100644
--- a/voice_engine/include/voe_base.h
+++ b/voice_engine/include/voe_base.h
@@ -14,7 +14,6 @@
// - Initialization and termination.
// - Trace information on text files or via callbacks.
// - Multi-channel support (mixing, sending to multiple destinations etc.).
-// - Call setup (port and address) for receiving and sending sides.
//
// To support other codecs than G.711, the VoECodec sub-API must be utilized.
//
@@ -140,28 +139,6 @@
// Deletes an existing channel and releases the utilized resources.
virtual int DeleteChannel(int channel) = 0;
- // Sets the local receiver port and address for a specified
- // |channel| number.
- virtual int SetLocalReceiver(int channel, int port,
- int RTCPport = kVoEDefault,
- const char ipAddr[64] = NULL,
- const char multiCastAddr[64] = NULL) = 0;
-
- // Gets the local receiver port and address for a specified
- // |channel| number.
- virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64]) = 0;
-
- // Sets the destination port and address for a specified |channel| number.
- virtual int SetSendDestination(int channel, int port,
- const char ipAddr[64],
- int sourcePort = kVoEDefault,
- int RTCPport = kVoEDefault) = 0;
-
- // Gets the destination port and address for a specified |channel| number.
- virtual int GetSendDestination(int channel, int& port, char ipAddr[64],
- int& sourcePort, int& RTCPport) = 0;
-
// Prepares and initiates the VoiceEngine for reception of
// incoming RTP/RTCP packets on the specified |channel|.
virtual int StartReceive(int channel) = 0;
diff --git a/voice_engine/include/voe_network.h b/voice_engine/include/voe_network.h
index 10acf1c..9a2ff73 100644
--- a/voice_engine/include/voe_network.h
+++ b/voice_engine/include/voe_network.h
@@ -11,9 +11,6 @@
// This sub-API supports the following functionalities:
//
// - External protocol support.
-// - Extended port and address APIs.
-// - Port and address filters.
-// - Windows GQoS functions.
// - Packet timeout notification.
// - Dead-or-Alive connection observations.
// - Transmission of raw RTP/RTCP packets into existing channels.
@@ -93,49 +90,6 @@
virtual int ReceivedRTCPPacket(
int channel, const void* data, unsigned int length) = 0;
- // Gets the source ports and IP address of incoming packets on a
- // specific |channel|.
- virtual int GetSourceInfo(
- int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
-
- // Gets the local (host) IP address.
- virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false) = 0;
-
- // Enables IPv6 for a specified |channel|.
- virtual int EnableIPv6(int channel) = 0;
-
- // Gets the current IPv6 staus for a specified |channel|.
- virtual bool IPv6IsEnabled(int channel) = 0;
-
- // Enables a port and IP address filter for incoming packets on a
- // specific |channel|.
- virtual int SetSourceFilter(int channel,
- int rtpPort, int rtcpPort = 0, const char ipAddr[64] = 0) = 0;
-
- // Gets the current port and IP-address filter for a specified |channel|.
- virtual int GetSourceFilter(
- int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
-
- // Sets the six-bit Differentiated Services Code Point (DSCP) in the
- // IP header of the outgoing stream for a specific |channel|.
- virtual int SetSendTOS(int channel,
- int DSCP, int priority = -1, bool useSetSockopt = false) = 0;
-
- // Gets the six-bit DSCP in the IP header of the outgoing stream for
- // a specific channel.
- virtual int GetSendTOS(
- int channel, int& DSCP, int& priority, bool& useSetSockopt) = 0;
-
- // Sets the Generic Quality of Service (GQoS) service level.
- // The Windows operating system then maps to a Differentiated Services
- // Code Point (DSCP) and to an 802.1p setting. [Windows only]
- virtual int SetSendGQoS(
- int channel, bool enable, int serviceType, int overrideDSCP = 0) = 0;
-
- // Gets the Generic Quality of Service (GQoS) service level.
- virtual int GetSendGQoS(
- int channel, bool& enabled, int& serviceType, int& overrideDSCP) = 0;
-
// Enables or disables warnings that report if packets have not been
// received in |timeoutSeconds| seconds for a specific |channel|.
virtual int SetPacketTimeoutNotification(
@@ -161,12 +115,6 @@
virtual int GetPeriodicDeadOrAliveStatus(
int channel, bool& enabled, int& sampleTimeSeconds) = 0;
- // Handles sending a raw UDP data packet over an existing RTP or RTCP
- // socket.
- virtual int SendUDPPacket(
- int channel, const void* data, unsigned int length,
- int& transmittedBytes, bool useRtcpSocket = false) = 0;
-
protected:
VoENetwork() {}
virtual ~VoENetwork() {}
diff --git a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index bbdd64d..c5f0500 100644
--- a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -13,9 +13,30 @@
#include "before_initialization_fixture.h"
#include "scoped_ptr.h"
+#include "webrtc/common_types.h"
class TestErrorObserver;
+class LoopBackTransport : public webrtc::Transport {
+ public:
+ LoopBackTransport(webrtc::VoENetwork* voe_network)
+ : voe_network_(voe_network) {
+ }
+
+ virtual int SendPacket(int channel, const void *data, int len) {
+ voe_network_->ReceivedRTPPacket(channel, data, len);
+ return len;
+ }
+
+ virtual int SendRTCPPacket(int channel, const void *data, int len) {
+ voe_network_->ReceivedRTCPPacket(channel, data, len);
+ return len;
+ }
+
+ private:
+ webrtc::VoENetwork* voe_network_;
+};
+
// This fixture initializes the voice engine in addition to the work
// done by the before-initialization fixture. It also registers an error
// observer which will fail tests on error callbacks. This fixture is
diff --git a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
index d1e6039..353888a 100644
--- a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
+++ b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
@@ -12,8 +12,6 @@
#include <cstring>
-static const char* kLoopbackIp = "127.0.0.1";
-
AfterStreamingFixture::AfterStreamingFixture()
: channel_(voe_base_->CreateChannel()) {
EXPECT_GE(channel_, 0);
@@ -30,7 +28,9 @@
voe_file_->StopPlayingFileAsMicrophone(channel_);
PausePlaying();
+ EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
voe_base_->DeleteChannel(channel_);
+ delete transport_;
}
void AfterStreamingFixture::SwitchToManualMicrophone() {
@@ -59,8 +59,8 @@
}
void AfterStreamingFixture::SetUpLocalPlayback() {
- EXPECT_EQ(0, voe_base_->SetSendDestination(channel_, 8000, kLoopbackIp));
- EXPECT_EQ(0, voe_base_->SetLocalReceiver(0, 8000));
+ transport_ = new LoopBackTransport(voe_network_);
+ EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
webrtc::CodecInst codec;
codec.channels = 1;
diff --git a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h
index 6b0a61f..26e37a7 100644
--- a/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h
+++ b/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h
@@ -42,6 +42,8 @@
private:
void SetUpLocalPlayback();
+
+ LoopBackTransport* transport_;
};
diff --git a/voice_engine/test/auto_test/standard/mixing_test.cc b/voice_engine/test/auto_test/standard/mixing_test.cc
index 6a90c07..1d2aea3 100644
--- a/voice_engine/test/auto_test/standard/mixing_test.cc
+++ b/voice_engine/test/auto_test/standard/mixing_test.cc
@@ -32,7 +32,13 @@
: input_filename_(test::OutputPath() + "mixing_test_input.pcm"),
output_filename_(test::OutputPath() + "mixing_test_output.pcm") {
}
-
+ void SetUp() {
+ transport_ = new LoopBackTransport(voe_network_);
+ }
+ void TearDown() {
+ delete transport_;
+ }
+
// Creates and mixes |num_remote_streams| which play a file "as microphone"
// with |num_local_streams| which play a file "locally", using a constant
// amplitude of |input_value|. The local streams manifest as "anonymous"
@@ -165,8 +171,7 @@
// Start up a single remote stream.
void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) {
EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst));
- EXPECT_EQ(0, voe_base_->SetLocalReceiver(stream, port));
- EXPECT_EQ(0, voe_base_->SetSendDestination(stream, port, "127.0.0.1"));
+ EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
EXPECT_EQ(0, voe_base_->StartReceive(stream));
EXPECT_EQ(0, voe_base_->StartPlayout(stream));
EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
@@ -180,12 +185,14 @@
EXPECT_EQ(0, voe_base_->StopSend(streams[i]));
EXPECT_EQ(0, voe_base_->StopPlayout(streams[i]));
EXPECT_EQ(0, voe_base_->StopReceive(streams[i]));
+ EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(streams[i]));
EXPECT_EQ(0, voe_base_->DeleteChannel(streams[i]));
}
}
const std::string input_filename_;
const std::string output_filename_;
+ LoopBackTransport* transport_;
};
// These tests assume a maximum of three mixed participants. We typically allow
diff --git a/voice_engine/test/auto_test/standard/network_before_streaming_test.cc b/voice_engine/test/auto_test/standard/network_before_streaming_test.cc
deleted file mode 100644
index 7a41e80..0000000
--- a/voice_engine/test/auto_test/standard/network_before_streaming_test.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 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.
- */
-
-#include "after_initialization_fixture.h"
-
-class NetworkBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp() {
- channel_ = voe_base_->CreateChannel();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(channel_);
- }
-
- int channel_;
-};
-
-TEST_F(NetworkBeforeStreamingTest,
- GetSourceInfoReturnsEmptyValuesForUnconfiguredChannel) {
- char src_ip[32] = "0.0.0.0";
- int src_rtp_port = 1234;
- int src_rtcp_port = 1235;
-
- EXPECT_EQ(0, voe_network_->GetSourceInfo(
- channel_, src_rtp_port, src_rtcp_port, src_ip));
- EXPECT_EQ(0, src_rtp_port);
- EXPECT_EQ(0, src_rtcp_port);
- EXPECT_STRCASEEQ("", src_ip);
-}
-
-TEST_F(NetworkBeforeStreamingTest,
- GetSourceFilterReturnsEmptyValuesForUnconfiguredChannel) {
- int filter_port = -1;
- int filter_port_rtcp = -1;
- char filter_ip[32] = "0.0.0.0";
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_port, filter_port_rtcp, filter_ip));
-
- EXPECT_EQ(0, filter_port);
- EXPECT_EQ(0, filter_port_rtcp);
- EXPECT_STRCASEEQ("", filter_ip);
-}
-
-TEST_F(NetworkBeforeStreamingTest, SetSourceFilterSucceeds) {
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
-}
diff --git a/voice_engine/test/auto_test/standard/network_test.cc b/voice_engine/test/auto_test/standard/network_test.cc
index 8bf2d11..f4713dc 100644
--- a/voice_engine/test/auto_test/standard/network_test.cc
+++ b/voice_engine/test/auto_test/standard/network_test.cc
@@ -23,79 +23,6 @@
using ::testing::Between;
-TEST_F(NetworkTest, GetSourceInfoReturnsPortsAndIpAfterReceivingPackets) {
- // Give some time to send speech packets.
- Sleep(200);
-
- int rtp_port = 0;
- int rtcp_port = 0;
- char source_ip[32] = "127.0.0.1";
-
- EXPECT_EQ(0, voe_network_->GetSourceInfo(channel_, rtp_port, rtcp_port,
- source_ip));
-
- EXPECT_EQ(kDefaultRtpPort, rtp_port);
- EXPECT_EQ(kDefaultRtcpPort, rtcp_port);
-}
-
-TEST_F(NetworkTest, NoFilterIsEnabledByDefault) {
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(0, filter_rtp_port);
- EXPECT_EQ(0, filter_rtcp_port);
- EXPECT_STREQ("", filter_ip);
-}
-
-TEST_F(NetworkTest, ManualCanFilterRtpPort) {
- TEST_LOG("No filter, should hear audio.\n");
- Sleep(1000);
-
- int port_to_block = kDefaultRtpPort + 10;
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, port_to_block));
-
- // Changes should take effect immediately.
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(port_to_block, filter_rtp_port);
-
- TEST_LOG("Now filtering port %d, should not hear audio.\n", port_to_block);
- Sleep(1000);
-
- TEST_LOG("Removing filter, should hear audio.\n");
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
- Sleep(1000);
-}
-
-TEST_F(NetworkTest, ManualCanFilterIp) {
- TEST_LOG("You should hear audio.\n");
- Sleep(1000);
-
- int rtcp_port_to_block = kDefaultRtcpPort + 10;
- TEST_LOG("Filtering IP 10.10.10.10, should not hear audio.\n");
- EXPECT_EQ(0, voe_network_->SetSourceFilter(
- channel_, 0, rtcp_port_to_block, "10.10.10.10"));
-
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(0, filter_rtp_port);
- EXPECT_EQ(rtcp_port_to_block, filter_rtcp_port);
- EXPECT_STREQ("10.10.10.10", filter_ip);
-}
-
TEST_F(NetworkTest,
CallsObserverOnTimeoutAndRestartWhenPacketTimeoutNotificationIsEnabled) {
// First, get rid of the default, asserting observer and install our observer.
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 3549cb3..7815288 100644
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -101,10 +101,10 @@
second_channel_ = voe_base_->CreateChannel();
EXPECT_GE(second_channel_, 0);
- EXPECT_EQ(0, voe_base_->SetSendDestination(
- second_channel_, 8002, "127.0.0.1"));
- EXPECT_EQ(0, voe_base_->SetLocalReceiver(
- second_channel_, 8002));
+ transport_ = new LoopBackTransport(voe_network_);
+ EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
+ *transport_));
+
EXPECT_EQ(0, voe_base_->StartReceive(second_channel_));
EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
@@ -115,10 +115,13 @@
}
void TearDown() {
+ EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(second_channel_));
voe_base_->DeleteChannel(second_channel_);
+ delete transport_;
}
int second_channel_;
+ LoopBackTransport* transport_;
};
void RtcpAppHandler::OnApplicationDataReceived(
diff --git a/voice_engine/test/auto_test/voe_cpu_test.cc b/voice_engine/test/auto_test/voe_cpu_test.cc
index 14e4a00..2737ee8 100644
--- a/voice_engine/test/auto_test/voe_cpu_test.cc
+++ b/voice_engine/test/auto_test/voe_cpu_test.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "voe_cpu_test.h"
+
#include <stdio.h>
#include <string.h>
#include <time.h>
@@ -16,7 +18,8 @@
#include <conio.h>
#endif
-#include "voe_cpu_test.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
using namespace webrtc;
@@ -45,6 +48,7 @@
VoEFile* file = _mgr.FilePtr();
VoECodec* codec = _mgr.CodecPtr();
VoEAudioProcessing* apm = _mgr.APMPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
int channel(-1);
CodecInst isac;
@@ -59,8 +63,12 @@
CHECK(base->Init());
channel = base->CreateChannel();
- CHECK(base->SetLocalReceiver(channel, 5566));
- CHECK(base->SetSendDestination(channel, 5566, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_socket_transport(
+ new VoiceChannelTransport(voe_network, channel));
+
+ CHECK(voice_socket_transport->SetSendDestination("127.0.0.1", 5566));
+ CHECK(voice_socket_transport->SetLocalReceiver(5566));
+
CHECK(codec->SetRecPayloadType(channel, isac));
CHECK(codec->SetSendCodec(channel, isac));
@@ -86,7 +94,6 @@
base->DeleteChannel(channel);
CHECK(base->Terminate());
-
return 0;
}
diff --git a/voice_engine/test/auto_test/voe_extended_test.cc b/voice_engine/test/auto_test/voe_extended_test.cc
index d4c49c4..ce26992 100644
--- a/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/voice_engine/test/auto_test/voe_extended_test.cc
@@ -8,22 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/voice_engine/test/auto_test/voe_extended_test.h"
+
#include <stdio.h>
#include <string.h>
#include <vector>
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
-#include "webrtc/voice_engine/test/auto_test/voe_extended_test.h"
#if defined(_WIN32)
#include <conio.h>
-#include <winsock2.h>
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
#include <netdb.h>
#endif
@@ -158,21 +153,6 @@
return len;
}
-XTransport::XTransport(VoENetwork* netw, VoEFile* file) :
- _netw(netw), _file(file) {
-}
-
-int XTransport::SendPacket(int channel, const void *data, int len) {
- // loopback
- // _netw->ReceivedRTPPacket(channel, data, len);
-
- return 0;
-}
-
-int XTransport::SendRTCPPacket(int, const void *, int) {
- return 0;
-}
-
// ----------------------------------------------------------------------------
// VoERTPObserver
// ----------------------------------------------------------------------------
@@ -255,8 +235,14 @@
_playing[channel] = false;
_sending[channel] = false;
- voe_base_->SetLocalReceiver(channel, rtpPort);
- voe_base_->SetSendDestination(channel, rtpPort, "127.0.0.1");
+ VoENetwork* voe_network = _mgr.NetworkPtr();
+
+ voice_channel_transports_[channel].reset(
+ new VoiceChannelTransport(voe_network, channel));
+
+ voice_channel_transports_[channel]->SetSendDestination("127.0.0.1", rtpPort);
+ voice_channel_transports_[channel]->SetLocalReceiver(rtpPort);
+
if (listen) {
_listening[channel] = true;
voe_base_->StartReceive(channel);
@@ -286,6 +272,7 @@
_sending[channel] = false;
voe_base_->StopSend(channel);
}
+ voice_channel_transports_[channel].reset(NULL);
}
void VoEExtendedTest::Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone,
@@ -333,7 +320,7 @@
// instance variable since it is required in order to appease the
// gods of darkness.
VoEBase* voe_base_ = _mgr.BasePtr();
- VoENetwork* netw = _mgr.NetworkPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
#ifdef _TEST_RTP_RTCP_
VoERTP_RTCP* rtp = _mgr.RTP_RTCPPtr();
#endif
@@ -555,305 +542,6 @@
ANL();
// ------------------------------------------------------------------------
- // >> SetLocalReceiver
- //
- // State: VE not initialized, no existing channels
- TEST_MUSTPASS(voe_base_->Init());
-
- int ch;
-
- TEST(SetLocalReceiver);
- ANL();
-
- // no channel created yet => should fail
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 100));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
-#ifdef WEBRTC_IOS
- printf("\nNOTE: Local IP must be set in source code (line %d) \n",
- __LINE__ + 1);
- char* localIp = "127.0.0.1";
-#else
- char localIp[64] = { 0 };
- TEST_MUSTPASS(netw->GetLocalIP(localIp));
- MARK();
- // NOTE: This API is supported on Win, Mac and Linux and may fail or not
- // return local IP for other platforms.
-#endif
-
- // trivial invalid function calls
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch+1, 12345));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, -1));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
-
- // check conflict with ongoing receiving
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_ERROR(VE_ALREADY_LISTENING);
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // check conflict with ongoing transmission
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_ERROR(VE_ALREADY_SENDING);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
-
- // valid function calls
- // Need to sleep between, otherwise it may fail for unknown reason
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, NULL,
- "230.1.2.3"));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp,
- "230.1.2.3"));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 5555, NULL));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SleepMs(100);
-
- // STATE: no media but sockets exists and are binded to 12345 and 12346
- // respectively
-
- // Add some dynamic tests as well:
-
- // ensure that last setting is used (cancels old settings)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
- MARK();
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 54321));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 54321, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- Play(ch, 1000, true, true);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetLocalReceiver
- //
- // State: VE initialized, no existing channels
- TEST(GetLocalReceiver);
- ANL();
-
- int port;
- char ipaddr[64];
- int RTCPport;
-
- ch = voe_base_->CreateChannel();
-
- // verify non-configured (blank) local receiver
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 0);
- TEST_MUSTPASS(RTCPport != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // check some trivial set/get combinations
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 12346);
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0); // now binded to "any" IP
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 55555))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 55555);
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 12346);
- TEST_MUSTPASS(strcmp(ipaddr, localIp) != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendDestination
- //
- // State: VE initialized, no existing channels
- TEST(SetSendDestination);
- ANL();
-
- // call without existing channel
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
- // trivial fail tests
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 65536, "127.0.0.1"));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTP port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid source port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
- 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTCP port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.300"));
- MARK();
- TEST_ERROR(VE_INVALID_IP_ADDRESS); // invalid IP address
-
- // sockets must be created first to support multi-cast (not required
- // otherwise)
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
- MARK();
- TEST_ERROR(VE_SOCKET_ERROR);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555)); // create sockets
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
- MARK(); // should work now
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- // STATE: one channel created, no sockets exist
-
- // valid function calls
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", kVoEDefault,
- 55555));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444,
- 55555));
- MARK();
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- // create receive sockets first and then an extra pair of send sockets
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1", 11111));
- MARK(); // binds to 11111
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetSendDestination
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSendDestination
- //
- // State: VE initialized, no existing channels
- TEST(GetSendDestination);
- ANL();
-
- int sourcePort;
-
- ch = voe_base_->CreateChannel();
-
- // verify non-configured (blank) local receiver
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 0);
- TEST_MUSTPASS(sourcePort != 0);
- TEST_MUSTPASS(RTCPport != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // check some trivial set/get combinations
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 0); // should be 0 since no local receiver has
- // NOT been defined yet
- TEST_MUSTPASS(RTCPport != 44445);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 55555); // should be equal to local port
- TEST_MUSTPASS(RTCPport != 44445);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
- // NULL as IP-address input should work as well
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, NULL, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 0);
- TEST_MUSTPASS(RTCPport != 44445);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
// >> StartReceive
// >> StopReceive
//
@@ -871,14 +559,14 @@
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
- ch = voe_base_->CreateChannel();
+ int ch = voe_base_->CreateChannel();
- // sockets must be created first
+ // must register external transport first.
TEST_MUSTPASS(!voe_base_->StartReceive(0));
MARK();
- TEST_ERROR(VE_SOCKETS_NOT_INITED);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
+
+ ExtendedTestTransport* ptrTransport = new ExtendedTestTransport(voe_network);
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(ch, *ptrTransport));
MARK(); // should work this time
// enable again (should work)
@@ -892,7 +580,6 @@
MARK();
// Verify in loopback
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
TEST_MUSTPASS(voe_base_->StartSend(ch));
Play(ch, 1000, true, true);
TEST_MUSTPASS(voe_base_->StopSend(ch));
@@ -900,16 +587,15 @@
MARK();
voe_base_->DeleteChannel(0);
+ delete ptrTransport;
ch = voe_base_->CreateChannel();
// Ensure that it is OK to add delay between SetLocalReceiver and StarListen
TEST_LOG("\nspeak after 2 seconds and ensure that no delay is added:\n");
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
Sleep(2000, true); // adding emulated delay here
TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
TEST_MUSTPASS(voe_base_->StartSend(ch));
Play(ch, 2000, true, true);
TEST_MUSTPASS(voe_base_->StopSend(ch));
@@ -922,23 +608,23 @@
for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
MARK();
}
for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- TEST_MUSTPASS(voe_base_->StopReceive(i));
- MARK();
voe_base_->DeleteChannel(i);
+ MARK();
}
for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
+ ExtendedTestTransport* ptrTransport =
+ new ExtendedTestTransport(voe_network);
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(ch, *ptrTransport));
TEST_MUSTPASS(voe_base_->StartReceive(ch));
MARK();
TEST_MUSTPASS(voe_base_->StopReceive(ch));
MARK();
voe_base_->DeleteChannel(ch);
+ delete ptrTransport;
}
ANL();
@@ -1033,25 +719,24 @@
MARK();
TEST_ERROR(VE_DESTINATION_NOT_INITED);
+
// initialize destination and try again (should work even without existing
// sockets)
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
+ ptrTransport = new ExtendedTestTransport(voe_network);
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(ch, *ptrTransport));
TEST_MUSTPASS(voe_base_->StartSend(ch));
MARK();
SleepMs(100);
- // STATE: sockets should now have been created automatically at the first
- // transmitted packet should be binded to 33333 and "0.0.0.0"
TEST_MUSTPASS(voe_base_->StopSend(ch));
MARK();
voe_base_->DeleteChannel(ch);
+ delete ptrTransport;
ch = voe_base_->CreateChannel();
- // try loopback with unique send sockets (closed when channel is deleted or
- // new source is set)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
+ ptrTransport = new ExtendedTestTransport(voe_network);
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(ch, *ptrTransport));
TEST_MUSTPASS(voe_base_->StartSend(ch));
MARK();
TEST_MUSTPASS(voe_base_->StartReceive(ch));
@@ -1061,31 +746,7 @@
TEST_MUSTPASS(voe_base_->StopReceive(ch));
voe_base_->DeleteChannel(ch);
- ANL();
- // Multi-channel tests
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333 + 2*i));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333 + 2*i, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- TEST_MUSTPASS(voe_base_->StopSend(i));
- MARK();
- voe_base_->DeleteChannel(i);
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 45633 + 2*i));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 45633 + 2*i, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- MARK();
- voe_base_->DeleteChannel(ch);
- }
ANL();
AOK();
ANL();
@@ -1163,9 +824,11 @@
ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, ch));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartSend(ch));
TEST_MUSTPASS(voe_base_->StartPlayout(ch));
@@ -1206,9 +869,9 @@
#ifdef _TEST_RTP_RTCP_
TEST_MUSTPASS(rtp->SetRTCP_CNAME(ch, "Johnny"));
#endif
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 12349));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
- 12349));
+ voice_channel_transport.reset(new VoiceChannelTransport(voe_network, ch));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartSend(ch));
@@ -1226,18 +889,11 @@
TEST_MUSTPASS(rtp->GetRemoteRTCP_CNAME(ch, tmpStr));
TEST_MUSTPASS(_stricmp("Johnny", tmpStr));
#endif
- int rtpPort(0), rtcpPort(0);
- char ipAddr[64] = { 0 };
- TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(12349 != rtcpPort);
TEST_MUSTPASS(voe_base_->StopSend(ch));
TEST_MUSTPASS(voe_base_->StopPlayout(ch));
TEST_MUSTPASS(voe_base_->StopReceive(ch));
// Call StartSend before StartReceive
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
TEST_MUSTPASS(voe_base_->StartSend(ch));
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartPlayout(ch));
@@ -1251,9 +907,6 @@
TEST_MUSTPASS(voe_base_->StopReceive(ch));
// Try again using same ports
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
TEST_MUSTPASS(voe_base_->StartSend(ch));
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartPlayout(ch));
@@ -1264,38 +917,17 @@
SleepMs(7000); // Make sure we get RTCP packet
PAUSE
- // Verify correct RTCP source port
- TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(12345+1 != rtcpPort);
TEST_MUSTPASS(voe_base_->StopSend(ch));
TEST_MUSTPASS(voe_base_->StopPlayout(ch));
TEST_MUSTPASS(voe_base_->StopReceive(ch));
voe_base_->DeleteChannel(ch);
+
ch = voe_base_->CreateChannel();
+ voice_channel_transport.reset(new VoiceChannelTransport(voe_network, ch));
- // Try with extra send socket
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 22222));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 22222, "127.0.0.1", 11111));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("\nfull duplex is now activated (4)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // repeat default case starting with a fresh channel
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartSend(ch));
@@ -1310,7 +942,6 @@
TEST_MUSTPASS(voe_base_->StopReceive(ch));
// restart call again
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartPlayout(ch));
TEST_MUSTPASS(voe_base_->StartSend(ch));
@@ -1323,35 +954,15 @@
TEST_MUSTPASS(voe_base_->StopPlayout(ch));
TEST_MUSTPASS(voe_base_->StopReceive(ch));
- // force sending from new socket
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 12350,
- 12359));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_LOG("\nfull duplex is now activated (7)\n");
-
- PAUSE
-
- // Test getting send settings
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, rtpPort, ipAddr, sourcePort,
- rtcpPort));
- TEST_MUSTPASS(12345 != rtpPort);
- TEST_MUSTPASS(_stricmp("127.0.0.1", ipAddr));
- TEST_MUSTPASS(12350 != sourcePort);
- TEST_MUSTPASS(12359 != rtcpPort);
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
// new channel and new port
ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 33221));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport_2(
+ new VoiceChannelTransport(voe_network, ch));
+ voice_channel_transport_2->SetSendDestination("127.0.0.1", 33221);
+ voice_channel_transport_2->SetLocalReceiver(33221);
+
TEST_MUSTPASS(voe_base_->StartReceive(ch));
TEST_MUSTPASS(voe_base_->StartPlayout(ch));
TEST_MUSTPASS(voe_base_->StartSend(ch));
@@ -1360,36 +971,6 @@
PAUSE
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
-
-#ifndef WEBRTC_IOS
- // bind to local IP and try again
- strcpy(localIp, "127.0.0.1");
-#else
- localIp = "127.0.0.1";
-#endif
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33221, 12349, localIp));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, localIp));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
-
- TEST_LOG("\nfull duplex is now activated (9)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(33221 != rtpPort);
- TEST_MUSTPASS(_stricmp(localIp, ipAddr));
- TEST_MUSTPASS(12349 != rtcpPort);
-
ANL();
AOK();
ANL();
@@ -1477,7 +1058,6 @@
voe_base_->DeleteChannel(0);
TEST_MUSTPASS(voe_base_->Terminate());
-
return 0;
}
@@ -1491,7 +1071,7 @@
VoECallReport* report = _mgr.CallReportPtr();
VoEFile* file = _mgr.FilePtr();
VoEAudioProcessing* apm = _mgr.APMPtr();
- VoENetwork* netw = _mgr.NetworkPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
PrepareTest("CallReport");
@@ -1516,8 +1096,10 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -1581,14 +1163,14 @@
// All results should be -1 since dead-or-alive is not active
TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive) != -1);
MARK();
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 1));
SleepMs(2000);
// All results should be >= 0 since dead-or-alive is active
TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive));
MARK();
TEST_MUSTPASS(nDead == -1);
TEST_MUSTPASS(nAlive == -1)
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, false));
AOK();
ANL();
@@ -1641,6 +1223,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoECodec* codec = _mgr.CodecPtr();
VoEFile* file = _mgr.FilePtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
#ifdef _USE_EXTENDED_TRACE_
TEST_MUSTPASS(VoiceEngine::SetTraceFile(
@@ -1657,14 +1240,8 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- ExtendedTestTransport* ptrTransport(NULL);
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
-#else
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
-#endif
+ ExtendedTestTransport* ptrTransport = new ExtendedTestTransport(voe_network);
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -2140,10 +1717,6 @@
TEST_MUSTPASS(voe_base_->StopReceive(0));
// start loopback streaming (PCMU is default)
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(voe_base_->SetSendDestination(0,8000,"127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
-#endif
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@@ -2416,16 +1989,12 @@
TEST(SetRecPayloadType - removing receive codecs);
ANL();
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
-#endif
TEST_MUSTPASS(voe_base_->StartSend(0));
if (file) {
TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0,
- _mgr.AudioFilename(),
- true,
- true));
+ _mgr.AudioFilename(),
+ true,
+ true));
}
// Scan all supported and valid codecs and remove from receiving db, then
@@ -2940,12 +2509,7 @@
// set iSAC as sending codec
// set iSAC-wb as sending codec
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
-#else
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8001, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8001));
-#endif
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartReceive(0));
@@ -2991,10 +2555,8 @@
TEST_LOG("Skipping extended iSAC API tests - "
"WEBRTC_CODEC_ISAC not defined\n");
#endif // #if defined(WEBRTC_CODEC_ISAC)
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
delete ptrTransport;
-#endif
TEST_MUSTPASS(voe_base_->DeleteChannel(0));
TEST_MUSTPASS(voe_base_->Terminate());
@@ -3013,6 +2575,7 @@
VoEDtmf* dtmf = _mgr.DtmfPtr();
VoECodec* codec = _mgr.CodecPtr();
VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
std::string output_path = webrtc::test::OutputPath();
TEST_MUSTPASS(VoiceEngine::SetTraceFile(
@@ -3028,8 +2591,10 @@
//#endif
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -3339,6 +2904,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoEFile* file = _mgr.FilePtr();
VoEEncryption* encrypt = _mgr.EncryptionPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
#ifdef _USE_EXTENDED_TRACE_
TEST_MUSTPASS(VoiceEngine::SetTraceFile(
@@ -3354,8 +2920,10 @@
#endif
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -3850,6 +3418,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoEExternalMedia* xmedia = _mgr.ExternalMediaPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
// check if this interface is supported
if (!xmedia) {
@@ -3867,8 +3436,10 @@
#endif
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@@ -3973,6 +3544,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoEFile* file = _mgr.FilePtr();
VoECodec* codec = _mgr.CodecPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
#ifdef _USE_EXTENDED_TRACE_
TEST_MUSTPASS(VoiceEngine::SetTraceFile(
@@ -3989,8 +3561,10 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@@ -4389,9 +3963,10 @@
TEST_MUSTPASS(ch == -1);
TEST_MUSTPASS(voe_base_->StopPlayout(ch));
}
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(1, 12356, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(1, 12356));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport_1(
+ new VoiceChannelTransport(voe_network, 1));
+ voice_channel_transport_1->SetSendDestination("127.0.0.1", 12356);
+ voice_channel_transport_1->SetLocalReceiver(12356);
TEST_MUSTPASS(voe_base_->StartReceive(1));
TEST_MUSTPASS(voe_base_->StopPlayout(1));
TEST_MUSTPASS(voe_base_->StartSend(1));
@@ -4978,20 +4553,8 @@
int VoEExtendedTest::TestNetwork() {
PrepareTest("Network");
-#ifdef WEBRTC_ANDROID
- int sleepTime = 200;
- int sleepTime2 = 250;
-#elif defined(WEBRTC_IOS) // WEBRTC_IOS needs more delay for getSourceInfo()
- int sleepTime = 150;
- int sleepTime2 = 200;
-#else
- int sleepTime = 100;
- int sleepTime2 = 200;
-#endif
-
VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoENetwork* netw = _mgr.NetworkPtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
#ifdef _USE_EXTENDED_TRACE_
TEST_MUSTPASS(VoiceEngine::SetTraceFile((output_path +
@@ -5009,259 +4572,6 @@
TEST_MUSTPASS(voe_base_->Init());
// ------------------------------------------------------------------------
- // >> GetLocalIP
- //
- // State: VE initialized, no existing channels
- TEST(GetLocalIP);
- ANL();
-
-#ifdef WEBRTC_IOS
- // Should fail
- TEST_MUSTPASS(!netw->GetLocalIP(NULL, 0)); MARK();
- TEST_ERROR(VE_FUNC_NOT_SUPPORTED);
-
- ANL();
- printf("NOTE: Local IP must be set in source code (line %d) \n",
- __LINE__ + 1);
- const char* localIP = "192.168.1.4";
-
-#else
- // Must be big enough so that we can print an IPv6 address.
- char localIP[256] = {0};
-
- // invalid parameter
- TEST_MUSTPASS(!netw->GetLocalIP(NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // default function calls (IPv4)
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
- TEST_LOG("[local IPv4: %s]\n", localIP);
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
-
-#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
- // default function calls (IPv6)
- TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
- MARK();
- TEST_LOG("[local IPv6: %s]\n", localIP);
- TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
- MARK();
-#endif
-
- // one last call to ensure that local
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
-#endif
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalIP
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSourceInfo
- //
- // - VE initialized
- // - no existing channels
- TEST(GetSourceInfo);
- ANL();
-
- int rtpPort(0);
- int rtcpPort(0);
- char ipaddr[64] = { 0 };
- ExtendedTestTransport* ptrTransport(NULL);
-
- // call without valid channel
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // NULL as input string
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // call when external transport is enabled
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- delete ptrTransport;
-
- // call when external transport is disabled (no packet received yet)
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
- // send and receive packets with default settings for a while
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime2); // does not guarantee RTCP
-
- // verify remote parameters (exclude RTCP)
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // ensure that valid results are maintained after StopListen
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // verify that results are maintained after new call to SetLocalReceiver
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: not listening, not sending
- // send and receive packets with other settings for a while
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- // STATE: listening, sending
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9005);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // restart sending to and from local IP
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9005);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0); // should not be "127.0.0.1"
-
- // use non-default source port in outgoing packets
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9010));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: listening and sending using an extra local socket
-
- // stop/start sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- // verify that the unique source port is maintained for the extra socket
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // set new source port for outgoing packets (9010 -> 9020)
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9020));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-#ifdef WEBRTC_IOS
- SleepMs(500); // Need extra pause for some reason
-#endif
-
- // verify that the unique source port is set for the new extra socket
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9020);
- // STATE: listening and sending using an extra local socket
-
- // remove extra send socket and restart call again
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // delete channel => destroys the
- // extra socket
- TEST_MUSTPASS(voe_base_->CreateChannel()); // new channel uses one socket only
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000)); // use new port as well
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- // verify that remote info is correct
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: listening and sending using shared socket
-
- // use non-default source port in outgoing packets to create extra send
- // socket
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 7000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 7000, "127.0.0.1", 7010));
- // RTP src is 7010 => RTCP src = 7011
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 7010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // check RTCP port as well (should be 7010 + 1 = 7011)
- Sleep(8000, true);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 7010);
- TEST_MUSTPASS(rtcpPort != 7011);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetSourceInfo
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
// >> SetExternalTransport
//
// - VE initialized
@@ -5271,127 +4581,54 @@
TEST(SetExternalTransport);
ANL();
- ptrTransport = new ExtendedTestTransport(netw);
+ ExtendedTestTransport* ptrTransport = new ExtendedTestTransport(voe_network);
// call without valid channel
- TEST_MUSTPASS(!netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(!voe_network ->DeRegisterExternalTransport(0));
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
TEST_MUSTPASS(voe_base_->CreateChannel());
// different valid call combinations
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK();
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK();
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(!voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK(); // must deregister first
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
// STATE: external transport is disabled
-
- // initialize sending and ensure that external transport can't be enabled
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 1234, "127.0.0.2"));
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_ERROR(VE_SEND_SOCKETS_CONFLICT);
-
- // restart channel to ensure that "initialized sender" state is cleared
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // initialize receiving and ensure that external transport can't be enabled
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 5678));
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_ERROR(VE_RECEIVE_SOCKETS_CONFLICT);
-
- // restart channel to ensure that "initialized receiver" state is cleared
TEST_MUSTPASS(voe_base_->DeleteChannel(0));
TEST_MUSTPASS(voe_base_->CreateChannel());
// enable external transport and verify that "emulated loopback" works
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK();
TEST_MUSTPASS(voe_base_->StartSend(0)); // should only start recording
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(!voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK(); // should fail
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
+ TEST_MUSTPASS(voe_network ->RegisterExternalTransport(0, *ptrTransport));
MARK();
Play(0, 2000, true, true); // play file as mic and verify loopback audio
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
- // STATE: external transport is disabled
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
- int testError = VE_FUNC_NOT_SUPPORTED;
-#else
- int testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
-
- // check all APIs that should fail when external transport is enabled
- int DSCP, priority, serviceType, overrideDSCP, nBytes(0);
- bool useSetSockopt, enabled;
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 12345));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->GetLocalReceiver(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->GetSendDestination(0, rtpPort, ipaddr, rtpPort,
- rtcpPort));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->EnableIPv6(0))
- TEST_ERROR(testError);
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != false)
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
-
// modified i VoE 3.4 (can be called also for external transport)
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StopReceive(0));
-#if (!defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)) || \
- defined(WEBRTC_EXTERNAL_TRANSPORT)
- testError = VE_FUNC_NOT_SUPPORTED;
-#else
- testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0));
- TEST_ERROR(testError);
- TEST_MUSTPASS(!netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_ERROR(testError);
-#if !defined(_WIN32) || defined(WEBRTC_EXTERNAL_TRANSPORT)
- testError = VE_FUNC_NOT_SUPPORTED;
-#else
- testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
- TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0));
- TEST_ERROR(testError);
- TEST_MUSTPASS(!netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_ERROR(testError);
- char dummy[1] = { 'a' };
- TEST_MUSTPASS(!netw->SendUDPPacket(0, dummy, 1, nBytes));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
-
// always disable external transport before deleting the Transport object;
// will lead to crash for RTCP transmission otherwise
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterExternalTransport(0));
MARK();
delete ptrTransport;
@@ -5406,350 +4643,6 @@
// ------------------------------------------------------------------------
// ------------------------------------------------------------------------
- // >> EnableIPv6
- //
- // - VE initialized
- // - no existing channels
- // - no media
- // - NOTE: set _ENABLE_IPV6_TESTS_ to include these tests
- // - http://www.microsoft.com/resources/documentation/windows/xp/all/
- // proddocs/en-us/sag_ip_v6_pro_rt_enable.mspx?mfr=true
- // >> ipv6 install
- // >> ipv6 [-v] if [IfIndex]
- // >> ping6 ::1
- // >> ping6 fe80::1
-
-#ifdef _ENABLE_IPV6_TESTS_
-
- TEST(EnableIPv6); ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // call with enabled external transport
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- delete ptrTransport;
-
- // Test "locking" to IPv4
- TEST_MUSTPASS(netw->IPv6IsEnabled(0)); MARK(); // After this call we cannot
- // enable IPv6
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
-
- // Check that IPv6 address is invalid
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "::1")); MARK(); // fail
-
- // New channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // valid default call
- TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
- TEST_MUSTPASS(netw->GetLocalIP(localIP)); MARK(); // should still read IPv4
- TEST_LOG("[local IPv4: %s]", localIP);
-
- // ensure that Ipv6 is enabled
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
-
- // check that IPv4 address is invalid
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_ERROR(VE_INVALID_IP_ADDRESS);
-
- // verify usage of IPv6 loopback address
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- // IPv6 loopback address is 0:0:0:0:0:0:0:1
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true);
- ANL();
-
- // Restart channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
- // ensure that Ipv6 is enabled
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- // IPv6 loopback address is 0:0:0:0:0:0:0:1
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(), true,
- true);
- SleepMs(500); // ensure that we receieve some packets
-
- // SetSourceFilter and GetSourceFilter
- TEST(SetSourceFilter and GetSourceFilter for IPv6); ANL();
- char sourceIp[64] =
- { 0};
- char filterIp[64] =
- { 0};
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, sourceIp));
- TEST_LOG("Source port: %d \n", rtpPort);
- TEST_LOG("Source RTCP port: %d \n", rtcpPort);
- TEST_LOG("Source IP: %s \n", sourceIp);
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_LOG("Filter port RTP: %d \n", rtpPort);
- TEST_LOG("Filter port RTCP: %d \n", rtcpPort);
- TEST_LOG("Filter IP: %s \n", filterIp);
- TEST_MUSTPASS(0 != rtpPort);
- TEST_MUSTPASS(0 != rtcpPort);
- TEST_MUSTPASS(filterIp[0] != '\0');
- TEST_LOG("Set filter IP to %s => should hear audio\n", sourceIp);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, sourceIp));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(0 != rtpPort);
- TEST_MUSTPASS(0 != rtcpPort);
- TEST_MUSTPASS(_stricmp(filterIp, sourceIp));
- SleepMs(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::10:10:10"));
- SleepMs(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::0"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::"));
- SleepMs(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- SleepMs(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(filterIp[0] != '\0');
- SleepMs(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- SleepMs(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::"));
- SleepMs(1500);
-
- file->StopPlayingFileAsMicrophone(0);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
-#endif // #ifdef _ENABLE_IPV6_TESTS_
- // >> end of EnableIPv6
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSourceFilter
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(SetSourceFilter);
- ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // invalid parameters
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346, "300.300.300.300"));
- MARK();
- TEST_ERROR(VE_INVALID_IP_ADDRESS);
-
- // STATE: RTP filter port is 12345, RTCP filter port is 12346
-
- // disable all filters and ensure that media is received
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
-
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2000);
- TEST_MUSTPASS(rtcpPort != 2001);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- // clear states and restart loopback session
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // set RTP filter to port 2002 and verify that source 2000 is blocked
- TEST_MUSTPASS(netw->SetSourceFilter(0, 2002, 0, NULL));;
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // ensure that received packets originates from 2002 and that they now pass
- // the filter
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // RTP source is 2002
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2002, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2002, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2002);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- // clear states and restart loopback session
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // set IP filter to local IP and verify that default loopback stream is
- // blocked
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, localIP));;
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // ensure that received packets originates from the local IP and that they
- // now pass the filter
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // should pass the filter
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SleepMs(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2000);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- // STATE: no active media, IP filter is active
-
- // disable all filters
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));;
- MARK();
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetSourceFilter
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSourceFilter
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(GetSourceFilter);
- ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // invalid input parameters
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // valid call without any filter set
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // STATE: no active media and no enabled filters
-
- // set different filters and verify that they "bite"
- TEST_MUSTPASS(netw->SetSourceFilter(0, 54321, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 54321);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 15425, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtcpPort != 15425);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "192.168.199.19"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "192.168.199.19") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "0.0.0.0"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetSourceFilter
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
// >> RegisterDeadOrAliveObserver
// >> DeRegisterDeadOrAliveObserver
//
@@ -5762,24 +4655,24 @@
ANL();
// call without valid channel
- TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
+ TEST_MUSTPASS(!voe_network ->RegisterDeadOrAliveObserver(0, *this));
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ TEST_MUSTPASS(voe_network ->RegisterDeadOrAliveObserver(0, *this));
MARK();
- TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
+ TEST_MUSTPASS(!voe_network ->RegisterDeadOrAliveObserver(0, *this));
MARK(); // already registered
TEST_ERROR(VE_INVALID_OPERATION);
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterDeadOrAliveObserver(0));
MARK();
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterDeadOrAliveObserver(0));
MARK(); // OK to do it again
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ TEST_MUSTPASS(voe_network ->RegisterDeadOrAliveObserver(0, *this));
MARK();
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ TEST_MUSTPASS(voe_network ->DeRegisterDeadOrAliveObserver(0));
MARK();
TEST_MUSTPASS(voe_base_->DeleteChannel(0));
@@ -5798,39 +4691,43 @@
// - no media
// call without valid channel
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(!voe_network ->SetPeriodicDeadOrAliveStatus(0, false));
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
TEST_MUSTPASS(voe_base_->CreateChannel());
// Invalid paramters
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 0));
+ TEST_MUSTPASS(!voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 0));
MARK();
TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 151));
+ TEST_MUSTPASS(!voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 151));
MARK();
TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(1, true, 10));
+ TEST_MUSTPASS(!voe_network ->SetPeriodicDeadOrAliveStatus(1, true, 10));
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
int sampleTime(0);
+ bool enabled;
// Valid parameters
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 1));
MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(
+ voe_network ->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
TEST_MUSTPASS(enabled != true);
TEST_MUSTPASS(sampleTime != 1);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 150));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 150));
MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(
+ voe_network ->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
TEST_MUSTPASS(enabled != true);
TEST_MUSTPASS(sampleTime != 150);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, false));
MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
+ TEST_MUSTPASS(
+ voe_network ->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
TEST_MUSTPASS(enabled != false);
TEST_MUSTPASS(sampleTime != 150); // ensure last set time isnt modified
@@ -5839,25 +4736,25 @@
// STATE: full duplex media is active
// test the dead-or-alive mechanism
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
+ TEST_MUSTPASS(voe_network ->RegisterDeadOrAliveObserver(0, *this));
MARK();
TEST_LOG("\nVerify that Alive callbacks are received (dT=2sec): ");
fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 2));
SleepMs(6000);
TEST_LOG("\nChange dT to 1 second: ");
fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 1));
SleepMs(6000);
TEST_LOG("\nDisable dead-or-alive callbacks: ");
fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, false));
SleepMs(6000);
TEST_LOG("\nStop sending and enable callbacks again.\n");
TEST_LOG("Verify that Dead callbacks are received (dT=2sec): ");
fflush(NULL);
TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, true, 2));
SleepMs(6000);
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_LOG("\nRestart sending.\n");
@@ -5866,8 +4763,8 @@
SleepMs(6000);
TEST_LOG("\nDisable dead-or-alive callbacks.");
fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
+ TEST_MUSTPASS(voe_network ->SetPeriodicDeadOrAliveStatus(0, false));
+ TEST_MUSTPASS(voe_network ->DeRegisterDeadOrAliveObserver(0));
MARK();
StopMedia(0);
@@ -5898,47 +4795,47 @@
ANL();
// call without existing valid channel
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, false));
+ TEST_MUSTPASS(!voe_network ->SetPacketTimeoutNotification(0, false));
MARK();
TEST_ERROR(VE_CHANNEL_NOT_VALID);
TEST_MUSTPASS(voe_base_->CreateChannel());
// invalid function calls
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 0));
+ TEST_MUSTPASS(!voe_network ->SetPacketTimeoutNotification(0, true, 0));
MARK();
TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 151));
+ TEST_MUSTPASS(!voe_network ->SetPacketTimeoutNotification(0, true, 151));
MARK();
TEST_ERROR(VE_INVALID_ARGUMENT);
// valid function calls (no active media)
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
+ TEST_MUSTPASS(voe_network ->SetPacketTimeoutNotification(0, true, 2));
MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ TEST_MUSTPASS(voe_network ->GetPacketTimeoutNotification(0, enabled, timeOut));
MARK();
TEST_MUSTPASS(enabled != true);
TEST_MUSTPASS(timeOut != 2);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
+ TEST_MUSTPASS(voe_network ->SetPacketTimeoutNotification(0, false));
MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ TEST_MUSTPASS(voe_network ->GetPacketTimeoutNotification(0, enabled, timeOut));
MARK();
TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 10));
+ TEST_MUSTPASS(voe_network ->SetPacketTimeoutNotification(0, true, 10));
MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ TEST_MUSTPASS(voe_network ->GetPacketTimeoutNotification(0, enabled, timeOut));
MARK();
TEST_MUSTPASS(enabled != true);
TEST_MUSTPASS(timeOut != 10);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
+ TEST_MUSTPASS(voe_network ->SetPacketTimeoutNotification(0, true, 2));
MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ TEST_MUSTPASS(voe_network ->GetPacketTimeoutNotification(0, enabled, timeOut));
MARK();
TEST_MUSTPASS(enabled != true);
TEST_MUSTPASS(timeOut != 2);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
+ TEST_MUSTPASS(voe_network ->SetPacketTimeoutNotification(0, false));
MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
+ TEST_MUSTPASS(voe_network ->GetPacketTimeoutNotification(0, enabled, timeOut));
MARK();
TEST_MUSTPASS(enabled != false);
@@ -5947,512 +4844,10 @@
AOK();
ANL();
ANL();
-
- // >> end of SetPacketTimeoutNotification
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SendUDPPacket
- //
- // - VE initialized
- // - no existing channels
- // - no media
-
-
- // >> end of SendUDPPacket
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendTOS
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(SetSendTOS);
- ANL();
-#if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
-
- // call without existing valid channel
-
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // trivial invalid function calls
- TEST_MUSTPASS(!netw->SetSendTOS(0, -1)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 64)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -2)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, 8)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1)); MARK();
- TEST_ERROR(VE_SOCKET_ERROR); // must create sockets first
-
-#ifdef _WIN32
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 3000));
-
- // enable ToS using SetSockopt (should work without local binding)
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, true)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 1);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // try to disable SetSockopt while ToS is enabled (should fail)
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_ERROR(VE_TOS_INVALID); // must disable ToS before enabling SetSockopt
-
- // disable ToS to be able to stop using SetSockopt
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true)); MARK(); // disable ToS
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 0);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // to use the "non-SetSockopt" method, local binding is required,
- // trying without it should fail
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_ERROR(VE_TOS_ERROR); // must bind to local IP first
-
- // bind to local IP and try again (should work this time)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
- TEST_LOG("\nThis test needs to be run as administrator\n");
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 1);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != false);
-
- // STATE: binded to local IP, local port is 12345 and DSCP is 1 (not using
- // SetSockopt)
-
- // verify loopback audio with the current settings
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP using Wireshark.
- // Use filter ip.src == "RemoteIP".
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the "
- "remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
- TEST_LOG(" DSCP is set to 0x%02x\n", 1);
- SleepMs(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 2));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SleepMs(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 63));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SleepMs(100);
-
- // stop and resume sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0));
-#endif // _SEND_TO_REMOTE_IP_
- // Windows priority tests (priority cannot be set using setsockopt on Win)
- TEST_LOG("Testing priority\n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, true)); // Should fail
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-#endif // _WIN32
- // STATE: no media, disabled ToS, no defined receiver
-
- // Repeat tests above but using setsockopt() this time.
- // Binding to local IP should not be required.
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault));
- TEST_MUSTPASS(netw->SetSendTOS(0, 10, -1, true)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 10);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // STATE: *not* binded to local IP, local port is 12345 and DSCP is 10
- // (using SetSockopt)
-
- // verify loopback audio with the current settings
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP using Wireshark.
- // Use filter ip.src == "RemoteIP".
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the"
- " remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SleepMs(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 20, -1, true)); // use setsockopt()
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SleepMs(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 61, -1, true)); // use setsockopt()
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SleepMs(100);
-
- // stop and resume sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SleepMs(100);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true));
-#endif // _SEND_TO_REMOTE_IP_
-#if defined(WEBRTC_LINUX)
- // Linux priority tests (using setsockopt)
- TEST_LOG("Testing priority\n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-#endif // #if defined(WEBRTC_LINUX)
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX)
- // Fail tests for other than Wind and Linux
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, false)); // Should fail
- TEST_ERROR(VE_INVALID_ARGUMENT);
-#endif // #if !defined(_WIN32) && !defined(WEBRTC_LINUX)
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL(); AOK(); ANL(); ANL();
-
- // END #if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
-#else
- TEST_LOG("Skipping ToS tests - _WIN32, LINUX, MAC is not defined or "
- "WEBRTC_ANDROID is defined");
-#endif
-
- // >> end of SetSendTOS
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendGQoS (Windows only)
- //
- // - VE initialized
- // - no existing channels
- // - no media
- //
- // From qos.h:
- //
- // #define SERVICETYPE_NOTRAFFIC 0x00000000
- // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
- // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
- // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
- // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
- // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
- // #define SERVICETYPE_NOCHANGE 0x00000006
- // #define SERVICETYPE_NONCONFORMING 0x00000009
- // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
- // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
- //
- // #define SERVICE_BESTEFFORT 0x80010000
- // #define SERVICE_CONTROLLEDLOAD 0x80020000
- // #define SERVICE_GUARANTEED 0x80040000
- // #define SERVICE_QUALITATIVE 0x80200000
- //
- // (*) supported in WEBRTC VoE
- TEST(SetSendGQoS);
- ANL();
-#ifdef _WIN32
-
- // call without existing valid channel
- TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // supported service type but no sockets
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_SOCKETS_NOT_INITED);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
-
- // supported service type but sender is not initialized
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_DESTINATION_NOT_INITED);
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
-
- // invalid service types
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOTRAFFIC)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_UNAVAILABLE));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_GENERAL_INFORMATION));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOCHANGE)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NONCONFORMING));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_CONTROL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_CONTROLLEDLOAD)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_GUARANTEED)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_QUALITATIVE)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // Is ToS enabled here?
-
- // Settings which don't require binding to local IP
-
- // set SERVICETYPE_BESTEFFORT
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_BESTEFFORT);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_CONTROLLEDLOAD
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
- MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_CONTROLLEDLOAD);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_GUARANTEED
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_GUARANTEED);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_QUALITATIVE
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // disable GQoS
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // STATE: diabled QGoS, sockets exists, sending side is initialized, no media
-
- // Loopback tests using the four different GQoS settings
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- ANL();
- TEST_LOG("[SERVICETYPE_BESTEFFORT]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_CONTROLLEDLOAD]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_GUARANTEED]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_QUALITATIVE]");
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP mapping using Wireshark.
- // Use filter ip.src == "RemoteIP".
-
- // Modify the send destination on the fly
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
-
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
- " the remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT (0x%02x), should "
- "be mapped to DSCP = 0x00\n", serviceType);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_CONTROLLEDLOAD (0x%02x), "
- "should be mapped to DSCP = 0x18\n", serviceType);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_GUARANTEED (0x%02x), should "
- "be mapped to DSCP = 0x28\n", serviceType);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_QUALITATIVE (0x%02x), should"
- " be mapped to DSCP = 0x00\n", serviceType);
- SleepMs(100);
-#endif // _SEND_TO_REMOTE_IP_
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- // STATE: sockets exists, sending side is initialized, no media
-
- // Repeat test above but this time using overrideDSCP.
-
- // Some initial loopack tests.
- // NOTE - override DSCP requres binding to local IP.
-
- // should not work since QoS is enabled
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
- TEST_ERROR(VE_TOS_GQOS_CONFLICT);
-
- // disble QoS and try to override again (should fail again since local
- // binding is not done yet)
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
- TEST_ERROR(VE_GQOS_ERROR);
-
- // make proper settings and try again (should work this time)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
-
- // Now, let's try some loopback tests using override DSCP
-
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- ANL();
- TEST_LOG("[overrideDSCP=3]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 17));
- MARK();
- ANL();
- TEST_LOG("[overrideDSCP=17]");
- Play(0, 2000, true, true); // file should be played out here...
-
- // And finally, send to remote side as well to verify that the new mapping
- // works as it should.
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Modify the send destination on the fly
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
-
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
- " the remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 18));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 62));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 32));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 1));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SleepMs(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
- SleepMs(100);
-#endif // _SEND_TO_REMOTE_IP_
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL(); AOK(); ANL(); ANL();
-
-#else
- TEST_LOG("Skipping GQoS tests - _WIN32 is not defined");
-#endif // #ifdef _WIN32
- // >> end of SetSendGQoS
- // ------------------------------------------------------------------------
-
- if (file) {
- file->StopPlayingFileAsMicrophone(0);
- }
- voe_base_->StopSend(0);
- voe_base_->StopPlayout(0);
- voe_base_->StopReceive(0);
- voe_base_->DeleteChannel(0);
- voe_base_->Terminate();
-
- ANL();
- AOK();
return 0;
}
+ // >> end of SetPacketTimeoutNotification
+ // ------------------------------------------------------------------------
// ----------------------------------------------------------------------------
// VoEExtendedTest::TestRTP_RTCP
@@ -6460,8 +4855,7 @@
// Used to validate packets during the RTP audio level indication test.
class RTPAudioTransport: public Transport {
-public:
-
+ public:
RTPAudioTransport() :
mute_(false) {
}
@@ -6512,7 +4906,6 @@
assert(vad == 0 || vad == 1);
assert(level >= 0 && level <= 127);
}
-
return 0;
}
@@ -6521,7 +4914,7 @@
return 0;
}
-private:
+ private:
bool mute_;
};
@@ -6558,8 +4951,13 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(network, 0));
+
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
+
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -6662,8 +5060,12 @@
TEST_MUSTPASS(voe_base_->DeleteChannel(1));
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+
+ voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
+
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
+
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -6672,7 +5074,7 @@
ANL();
// ------------------------------------------------------------------------
- // >> InsertExtraRTPPacket
+ // >> SetLocalSSRC
int i(0);
@@ -6690,58 +5092,6 @@
true, true));
// ------------------------------------------------------------------------
- // >> InsertExtraRTPPacket
- TEST(InsertExtraRTPPacket);
- ANL();
-
- const char payloadData[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
-
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(-1, 0, false,
- payloadData, 8));
- MARK(); // invalid channel
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, -1, false,
- payloadData, 8));
- MARK(); // invalid payload type
- TEST_ERROR(VE_INVALID_PLTYPE);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 128, false,
- payloadData, 8));
- MARK(); // invalid payload type
- TEST_ERROR(VE_INVALID_PLTYPE);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- NULL, 8));
- MARK(); // invalid pointer
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- payloadData, 1500-28+1));
- MARK(); // invalid size
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- payloadData, 8));
- MARK(); // not sending
- TEST_ERROR(VE_NOT_SENDING);
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- SleepMs(1000);
- for (int p = 0; p < 128; p++) {
- TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, false,
- payloadData, 8));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, true,
- payloadData, 8));
- MARK();
- }
-
- // Ensure we have sent all extra packets before we move forward to avoid
- //incorrect error code
- SleepMs(1000);
-
- ANL();
-
- // ------------------------------------------------------------------------
// >> RTP dump APIs
TEST(Start/StopRtpDump);
ANL();
@@ -6879,8 +5229,12 @@
SleepMs(100);
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+
+ voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
+
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
+
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
@@ -7126,8 +5480,12 @@
// We have to re-register the audio codec payload type as stopReceive will
// clean the database
TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
+
+ voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
+
+ voice_channel_transport->SetSendDestination("127.0.0.1", 8000);
+ voice_channel_transport->SetLocalReceiver(8000);
+
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@@ -7163,6 +5521,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoEVideoSync* vsync = _mgr.VideoSyncPtr();
+ VoENetwork* network = _mgr.NetworkPtr();
// check if this interface is supported
if (!vsync)
@@ -7186,8 +5545,13 @@
TEST_MUSTPASS(voe_base_->Init());
TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(network, 0));
+
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
+
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@@ -7265,6 +5629,7 @@
VoEBase* voe_base_ = _mgr.BasePtr();
VoEVolumeControl* volume = _mgr.VolumeControlPtr();
+ VoENetwork* network = _mgr.NetworkPtr();
#ifdef _TEST_FILE_
VoEFile* file = _mgr.FilePtr();
#endif
@@ -7296,8 +5661,10 @@
TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
#endif
#endif
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(network, 0));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
+ voice_channel_transport->SetLocalReceiver(12345);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
diff --git a/voice_engine/test/auto_test/voe_extended_test.h b/voice_engine/test/auto_test/voe_extended_test.h
index c685d88..af549f8 100644
--- a/voice_engine/test/auto_test/voe_extended_test.h
+++ b/voice_engine/test/auto_test/voe_extended_test.h
@@ -12,7 +12,14 @@
#define WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
#include "voe_standard_test.h"
-#include "modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
namespace voetest {
@@ -425,20 +432,22 @@
int TestRTP_RTCP();
int TestVideoSync();
int TestVolumeControl();
- public:
+
int ErrorCode() const {
return _errCode;
}
void ClearErrorCode() {
_errCode = 0;
}
+
protected:
// from VoiceEngineObserver
void CallbackOnError(const int errCode, const int channel);
void CallbackOnTrace(const TraceLevel level, const char* message, const int length);
- protected:
+
// from VoEConnectionObserver
void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+
private:
void Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone = false,
bool addTimeMarker = false);
@@ -448,12 +457,12 @@
int RunMixingTest(int num_remote_channels, int num_local_channels,
int16_t input_value, int16_t max_output_value,
int16_t min_output_value);
- private:
+
VoETestManager& _mgr;
- private:
int _errCode;
bool _alive;
bool _listening[32];
+ scoped_ptr<webrtc::VoiceChannelTransport> voice_channel_transports_[32];
bool _playing[32];
bool _sending[32];
};
diff --git a/voice_engine/test/auto_test/voe_stress_test.cc b/voice_engine/test/auto_test/voe_stress_test.cc
index c921913..519bf83 100644
--- a/voice_engine/test/auto_test/voe_stress_test.cc
+++ b/voice_engine/test/auto_test/voe_stress_test.cc
@@ -23,10 +23,12 @@
#endif
#include "webrtc/voice_engine/test/auto_test/voe_stress_test.h"
-#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
#include "webrtc/voice_engine/voice_engine_defines.h" // defines build macros
using namespace webrtc;
@@ -122,6 +124,7 @@
// Get sub-API pointers
VoEBase* base = _mgr.BasePtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
// Set trace
// VALIDATE_STRESS(base->SetTraceFileName(
@@ -147,9 +150,12 @@
printf("Test will take approximately %d minutes. \n",
numberOfLoops * loopSleep / 1000 / 60 + 1);
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(voe_network, 0));
+
for (i = 0; i < numberOfLoops; ++i) {
- VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
- VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+ voice_channel_transport->SetSendDestination("127.0.0.1", 4800);
+ voice_channel_transport->SetLocalReceiver(4800);
VALIDATE_STRESS(base->StartReceive(0));
VALIDATE_STRESS(base->StartPlayout(0));
VALIDATE_STRESS(base->StartSend(0));
@@ -162,8 +168,9 @@
}
ANL();
- VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
- VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
+ VALIDATE_STRESS(voice_channel_transport->SetSendDestination("127.0.0.1",
+ 4800));
+ VALIDATE_STRESS(voice_channel_transport->SetLocalReceiver(4800));
VALIDATE_STRESS(base->StartReceive(0));
VALIDATE_STRESS(base->StartPlayout(0));
VALIDATE_STRESS(base->StartSend(0));
diff --git a/voice_engine/test/auto_test/voe_unit_test.cc b/voice_engine/test/auto_test/voe_unit_test.cc
index b152aab..8c4a7fb 100644
--- a/voice_engine/test/auto_test/voe_unit_test.cc
+++ b/voice_engine/test/auto_test/voe_unit_test.cc
@@ -248,13 +248,17 @@
bool send, bool fileAsMic, bool localFile) {
VoEBase* base = _mgr.BasePtr();
VoEFile* file = _mgr.FilePtr();
+ VoENetwork* voe_network = _mgr.NetworkPtr();
_listening[channel] = false;
_playing[channel] = false;
_sending[channel] = false;
-
- CHECK(base->SetLocalReceiver(channel, rtpPort));
- CHECK(base->SetSendDestination(channel, rtpPort, "127.0.0.1"));
+ voice_channel_transports_[channel].reset(
+ new VoiceChannelTransport(voe_network, channel));
+
+ CHECK(voice_channel_transports_[channel]->SetLocalReceiver(rtpPort));
+ CHECK(voice_channel_transports_[channel]->SetSendDestination("127.0.0.1",
+ rtpPort));
if (listen) {
_listening[channel] = true;
CHECK(base->StartReceive(channel));
diff --git a/voice_engine/test/auto_test/voe_unit_test.h b/voice_engine/test/auto_test/voe_unit_test.h
index 346713a..8db8d67 100644
--- a/voice_engine/test/auto_test/voe_unit_test.h
+++ b/voice_engine/test/auto_test/voe_unit_test.h
@@ -11,7 +11,9 @@
#ifndef WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
#define WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
-#include "voice_engine/test/auto_test/voe_standard_test.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
namespace voetest {
@@ -57,6 +59,7 @@
bool _listening[32];
bool _playing[32];
bool _sending[32];
+ scoped_ptr<webrtc::VoiceChannelTransport> voice_channel_transports_[32];
private:
bool _extOnOff;
diff --git a/voice_engine/test/cmd_test/voe_cmd_test.cc b/voice_engine/test/cmd_test/voe_cmd_test.cc
index 2035e4e..d4bfc4b 100644
--- a/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -18,8 +18,6 @@
#include <vector>
#include "gtest/gtest.h"
-#include "test/testsupport/fileutils.h"
-
#include "voe_errors.h"
#include "voe_base.h"
#include "voe_codec.h"
@@ -35,14 +33,15 @@
#include "voe_network.h"
#include "voe_neteq_stats.h"
#include "engine_configurations.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/udp_transport/include/channel_transport.h"
+#include "webrtc/test/testsupport/fileutils.h"
// Enable this this flag to run this test with hard coded
// IP/Port/codec and start test automatically with key input
// it could be useful in repeat tests.
//#define DEBUG
-// #define EXTERNAL_TRANSPORT
-
using namespace webrtc;
#define VALIDATE \
@@ -70,29 +69,6 @@
void RunTest(std::string out_path);
-#ifdef EXTERNAL_TRANSPORT
-
-class my_transportation : public Transport
-{
- int SendPacket(int channel,const void *data,int len);
- int SendRTCPPacket(int channel, const void *data, int len);
-};
-
-int my_transportation::SendPacket(int channel,const void *data,int len)
-{
- netw->ReceivedRTPPacket(channel, data, len);
- return 0;
-}
-
-int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
-{
- netw->ReceivedRTCPPacket(channel, data, len);
- return 0;
-}
-
-my_transportation my_transport;
-#endif
-
class MyObserver : public VoiceEngineObserver {
public:
virtual void CallbackOnError(const int channel, const int err_code);
@@ -270,44 +246,24 @@
cnt++;
int j = 0;
-#ifdef EXTERNAL_TRANSPORT
- my_transportation ch0transport;
- printf("Enabling external transport \n");
- netw->RegisterExternalTransport(0, ch0transport);
-#else
char ip[64];
#ifdef DEBUG
strcpy(ip, "127.0.0.1");
#else
- char localip[64];
- netw->GetLocalIP(localip);
- printf("local IP:%s\n", localip);
-
printf("1. 127.0.0.1 \n");
printf("2. Specify IP \n");
ASSERT_EQ(1, scanf("%i", &i));
- if (1 == i)
+ if (1 == i) {
strcpy(ip, "127.0.0.1");
- else {
+ } else {
printf("Specify remote IP: ");
ASSERT_EQ(1, scanf("%s", ip));
}
#endif
- int colons(0);
- while (ip[j] != '\0' && j < 64 && !(colons = (ip[j++] == ':')))
- ;
- if (colons) {
- printf("Enabling IPv6\n");
- res = netw->EnableIPv6(0);
- VALIDATE;
- }
-
- int rPort;
-#ifdef DEBUG
- rPort=8500;
-#else
+ int rPort = 8500;
+#ifndef DEBUG
printf("Specify remote port (1=1234): ");
ASSERT_EQ(1, scanf("%i", &rPort));
if (1 == rPort)
@@ -315,23 +271,24 @@
printf("Set Send port \n");
#endif
+ scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ new VoiceChannelTransport(netw, chan));
+
printf("Set Send IP \n");
- res = base1->SetSendDestination(chan, rPort, ip);
+ res = voice_channel_transport->SetSendDestination(ip, rPort);
VALIDATE;
- int lPort;
-#ifdef DEBUG
- lPort=8500;
-#else
+ int lPort = 8500;
+#ifndef DEBUG
printf("Specify local port (1=1234): ");
ASSERT_EQ(1, scanf("%i", &lPort));
if (1 == lPort)
lPort = 1234;
printf("Set Rec Port \n");
#endif
- res = base1->SetLocalReceiver(chan, lPort);
+
+ res = voice_channel_transport->SetLocalReceiver(lPort);
VALIDATE;
-#endif
printf("\n");
for (i = 0; i < codec->NumOfCodecs(); i++) {
@@ -367,12 +324,19 @@
#endif
int channel_index = 0;
std::vector<int> channels(kMaxNumChannels);
+ std::vector<scoped_ptr<VoiceChannelTransport> > voice_channel_transports;
+
for (i = 0; i < kMaxNumChannels; ++i) {
channels[i] = base1->CreateChannel();
int port = rPort + (i + 1) * 2;
- res = base1->SetSendDestination(channels[i], port, ip);
+
+ voice_channel_transports[i].reset(
+ new VoiceChannelTransport(netw, channels[i]));
+
+ printf("Set Send IP \n");
+ res = voice_channel_transports[i]->SetSendDestination(ip, port);
VALIDATE;
- res = base1->SetLocalReceiver(channels[i], port);
+ res = voice_channel_transports[i]->SetLocalReceiver(port);
VALIDATE;
res = codec->SetSendCodec(channels[i], cinst);
VALIDATE;
diff --git a/voice_engine/test/voice_engine_tests.gypi b/voice_engine/test/voice_engine_tests.gypi
index d02ef08..af6cb93 100644
--- a/voice_engine/test/voice_engine_tests.gypi
+++ b/voice_engine/test/voice_engine_tests.gypi
@@ -20,6 +20,7 @@
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/third_party/google-gflags/google-gflags.gyp:google-gflags',
'<(webrtc_root)/test/libtest/libtest.gyp:libtest',
+ '<(webrtc_root)/test/udp_transport.gyp:udp_transport',
],
'include_dirs': [
'auto_test',
@@ -59,7 +60,6 @@
'auto_test/standard/mixing_test.cc',
'auto_test/standard/neteq_stats_test.cc',
'auto_test/standard/neteq_test.cc',
- 'auto_test/standard/network_before_streaming_test.cc',
'auto_test/standard/network_test.cc',
'auto_test/standard/rtp_rtcp_before_streaming_test.cc',
'auto_test/standard/rtp_rtcp_test.cc',
@@ -102,6 +102,7 @@
'<(DEPTH)/testing/gtest.gyp:gtest',
'voice_engine_core',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/test/udp_transport.gyp:udp_transport',
],
'sources': [
'cmd_test/voe_cmd_test.cc',
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 864b6da..7446c22 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -683,265 +683,9 @@
{
return -1;
}
-
return 0;
}
-int VoEBaseImpl::SetLocalReceiver(int channel, int port, int RTCPport,
- const char ipAddr[64],
- const char multiCastAddr[64])
-{
- // Inititialize local receive sockets (RTP and RTCP).
- //
- // The sockets are always first closed and then created again by this
- // function call. The created sockets are by default also used for
- // transmission (unless source port is set in SetSendDestination).
- //
- // Note that, sockets can also be created automatically if a user calls
- // SetSendDestination and StartSend without having called SetLocalReceiver
- // first. The sockets are then created at the first packet transmission.
-
- CriticalSectionScoped cs(_shared->crit_sec());
- if (ipAddr == NULL && multiCastAddr == NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d)",
- channel, port, RTCPport);
- }
- else if (ipAddr != NULL && multiCastAddr == NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, ipAddr=%s)",
- channel, port, RTCPport, ipAddr);
- }
- else if (ipAddr == NULL && multiCastAddr != NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
- "multiCastAddr=%s)", channel, port, RTCPport, multiCastAddr);
- }
- else
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
- "ipAddr=%s, multiCastAddr=%s)", channel, port, RTCPport, ipAddr,
- multiCastAddr);
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((port < 0) || (port > 65535))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid RTP port");
- return -1;
- }
- if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
- != kVoEDefault) && (RTCPport > 65535)))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid RTCP port");
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalReceiver() failed to locate channel");
- return -1;
- }
-
- // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in
- // the module, which is the default.
- WebRtc_UWord16 rtcpPortUW16(0);
- if (RTCPport != kVoEDefault)
- {
- rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
- }
-
- return channelPtr->SetLocalReceiver(port, rtcpPortUW16, ipAddr,
- multiCastAddr);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
- kTraceWarning, "SetLocalReceiver() VoE is built for external "
- "transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetLocalReceiver(channel=%d, ipAddr[]=?)", channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalReceiver() failed to locate channel");
- return -1;
- }
- WebRtc_Word32 ret = channelPtr->GetLocalReceiver(port, RTCPport, ipAddr);
- if (ipAddr != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetLocalReceiver() => port=%d, RTCPport=%d, ipAddr=%s",
- port, RTCPport, ipAddr);
- }
- else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetLocalReceiver() => port=%d, RTCPport=%d", port, RTCPport);
- }
- return ret;
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetLocalReceiver() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::SetSendDestination(int channel, int port, const char* ipaddr,
- int sourcePort, int RTCPport)
-{
- WEBRTC_TRACE(
- kTraceApiCall,
- kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetSendDestination(channel=%d, port=%d, ipaddr=%s,"
- "sourcePort=%d, RTCPport=%d)",
- channel, port, ipaddr, sourcePort, RTCPport);
- CriticalSectionScoped cs(_shared->crit_sec());
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendDestination() failed to locate channel");
- return -1;
- }
- if ((port < 0) || (port > 65535))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid RTP port");
- return -1;
- }
- if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
- != kVoEDefault) && (RTCPport > 65535)))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid RTCP port");
- return -1;
- }
- if (((sourcePort != kVoEDefault) && (sourcePort < 0)) || ((sourcePort
- != kVoEDefault) && (sourcePort > 65535)))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid source port");
- return -1;
- }
-
- // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in the
- // module, which is the default.
- WebRtc_UWord16 rtcpPortUW16(0);
- if (RTCPport != kVoEDefault)
- {
- rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVoice,
- VoEId(_shared->instance_id(), channel),
- "SetSendDestination() non default RTCP port %u will be "
- "utilized",
- rtcpPortUW16);
- }
-
- return channelPtr->SetSendDestination(port, ipaddr, sourcePort,
- rtcpPortUW16);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSendDestination() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::GetSendDestination(int channel, int& port, char ipAddr[64],
- int& sourcePort, int& RTCPport)
-{
- WEBRTC_TRACE(
- kTraceApiCall,
- kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetSendDestination(channel=%d, ipAddr[]=?, sourcePort=?,"
- "RTCPport=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendDestination() failed to locate channel");
- return -1;
- }
- WebRtc_Word32 ret = channelPtr->GetSendDestination(port, ipAddr,
- sourcePort, RTCPport);
- if (ipAddr != NULL)
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetSendDestination() => port=%d, RTCPport=%d, ipAddr=%s, "
- "sourcePort=%d, RTCPport=%d",
- port, RTCPport, ipAddr, sourcePort, RTCPport);
- }
- else
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetSendDestination() => port=%d, RTCPport=%d, "
- "sourcePort=%d, RTCPport=%d",
- port, RTCPport, sourcePort, RTCPport);
- }
- return ret;
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSendDestination() VoE is built for external transport");
- return -1;
-#endif
-}
-
int VoEBaseImpl::StartReceive(int channel)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
@@ -1064,15 +808,6 @@
{
return 0;
}
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!channelPtr->ExternalTransport()
- && !channelPtr->SendSocketsInitialized())
- {
- _shared->SetLastError(VE_DESTINATION_NOT_INITED, kTraceError,
- "StartSend() must set send destination first");
- return -1;
- }
-#endif
if (StartSend() != 0)
{
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
@@ -1145,16 +880,6 @@
accLen += len;
assert(accLen < kVoiceEngineVersionMaxMessageSize);
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- len = AddExternalTransportBuild(versionPtr);
- if (len == -1)
- {
- return -1;
- }
- versionPtr += len;
- accLen += len;
- assert(accLen < kVoiceEngineVersionMaxMessageSize);
-#endif
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
len = AddExternalRecAndPlayoutBuild(versionPtr);
if (len == -1)
@@ -1207,13 +932,6 @@
return sprintf(str, "VoiceEngine 4.1.0\n");
}
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32 VoEBaseImpl::AddExternalTransportBuild(char* str) const
-{
- return sprintf(str, "External transport build\n");
-}
-#endif
-
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
WebRtc_Word32 VoEBaseImpl::AddExternalRecAndPlayoutBuild(char* str) const
{
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index 17c6ed3..92f1539 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -44,25 +44,6 @@
virtual int DeleteChannel(int channel);
- virtual int SetLocalReceiver(int channel, int port,
- int RTCPport = kVoEDefault,
- const char ipAddr[64] = NULL,
- const char multiCastAddr[64] = NULL);
-
- virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64]);
-
- virtual int SetSendDestination(int channel, int port,
- const char ipAddr[64],
- int sourcePort = kVoEDefault,
- int RTCPport = kVoEDefault);
-
- virtual int GetSendDestination(int channel,
- int& port,
- char ipAddr[64],
- int& sourcePort,
- int& RTCPport);
-
virtual int StartReceive(int channel);
virtual int StartPlayout(int channel);
@@ -125,9 +106,6 @@
WebRtc_Word32 AddBuildInfo(char* str) const;
WebRtc_Word32 AddVoEVersion(char* str) const;
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_Word32 AddExternalTransportBuild(char* str) const;
-#endif
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
WebRtc_Word32 AddExternalRecAndPlayoutBuild(char* str) const;
#endif
@@ -139,7 +117,6 @@
WebRtc_UWord32 _oldMicLevel;
AudioFrame _audioFrame;
voe::SharedData* _shared;
-
};
} // namespace webrtc
diff --git a/voice_engine/voe_network_impl.cc b/voice_engine/voe_network_impl.cc
index c563900..6c4cfc2 100644
--- a/voice_engine/voe_network_impl.cc
+++ b/voice_engine/voe_network_impl.cc
@@ -170,504 +170,6 @@
return channelPtr->ReceivedRTCPPacket((const WebRtc_Word8*) data, length);
}
-int VoENetworkImpl::GetSourceInfo(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetSourceInfo(channel=%d, rtpPort=?, rtcpPort=?, ipAddr[]=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "GetSourceInfo() invalid IP-address buffer");
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSourceInfo() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSourceInfo() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSourceInfo(rtpPort, rtcpPort, ipAddr);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSourceInfo() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetLocalIP(char ipAddr[64], bool ipv6)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetLocalIP(ipAddr[]=?, ipv6=%d)", ipv6);
- IPHONE_NOT_SUPPORTED(_shared->statistics());
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "GetLocalIP() invalid IP-address buffer");
- return -1;
- }
-
- // Create a temporary socket module to ensure that this method can be
- // called also when no channels are created.
- WebRtc_UWord8 numSockThreads(1);
- UdpTransport* socketPtr =
- UdpTransport::Create(
- -1,
- numSockThreads);
- if (NULL == socketPtr)
- {
- _shared->SetLastError(VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetLocalIP() failed to create socket module");
- return -1;
- }
-
- // Use a buffer big enough for IPv6 addresses and initialize it with zeros.
- char localIPAddr[256] = {0};
-
- if (ipv6)
- {
- char localIP[16];
- if (socketPtr->LocalHostAddressIPV6(localIP) != 0)
- {
- _shared->SetLastError(VE_INVALID_IP_ADDRESS, kTraceError,
- "GetLocalIP() failed to retrieve local IP - 1");
- UdpTransport::Destroy(socketPtr);
- return -1;
- }
- // Convert 128-bit address to character string (a:b:c:d:e:f:g:h)
- sprintf(localIPAddr,
- "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x"
- "%.2x:%.2x%.2x",
- localIP[0], localIP[1], localIP[2], localIP[3], localIP[4],
- localIP[5], localIP[6], localIP[7], localIP[8], localIP[9],
- localIP[10], localIP[11], localIP[12], localIP[13],
- localIP[14], localIP[15]);
- }
- else
- {
- WebRtc_UWord32 localIP(0);
- // Read local IP (as 32-bit address) from the socket module
- if (socketPtr->LocalHostAddress(localIP) != 0)
- {
- _shared->SetLastError(VE_INVALID_IP_ADDRESS, kTraceError,
- "GetLocalIP() failed to retrieve local IP - 2");
- UdpTransport::Destroy(socketPtr);
- return -1;
- }
- // Convert 32-bit address to character string (x.y.z.w)
- sprintf(localIPAddr, "%d.%d.%d.%d", (int) ((localIP >> 24) & 0x0ff),
- (int) ((localIP >> 16) & 0x0ff),
- (int) ((localIP >> 8) & 0x0ff),
- (int) (localIP & 0x0ff));
- }
-
- strcpy(ipAddr, localIPAddr);
-
- UdpTransport::Destroy(socketPtr);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "GetLocalIP() => ipAddr=%s", ipAddr);
- return 0;
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetLocalIP() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::EnableIPv6(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "EnableIPv6(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
- IPHONE_NOT_SUPPORTED(_shared->statistics());
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "EnableIPv6() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "EnableIPv6() external transport is enabled");
- return -1;
- }
- return channelPtr->EnableIPv6();
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "EnableIPv6() VoE is built for external transport");
- return -1;
-#endif
-}
-
-bool VoENetworkImpl::IPv6IsEnabled(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "IPv6IsEnabled(channel=%d)", channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return false;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "IPv6IsEnabled() failed to locate channel");
- return false;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "IPv6IsEnabled() external transport is enabled");
- return false;
- }
- return channelPtr->IPv6IsEnabled();
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "IPv6IsEnabled() VoE is built for external transport");
- return false;
-#endif
-}
-
-int VoENetworkImpl::SetSourceFilter(int channel,
- int rtpPort,
- int rtcpPort,
- const char ipAddr[64])
-{
- (ipAddr == NULL) ? WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetSourceFilter(channel=%d, rtpPort=%d,"
- " rtcpPort=%d)",
- channel, rtpPort, rtcpPort)
- : WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_shared->instance_id(), -1),
- "SetSourceFilter(channel=%d, rtpPort=%d,"
- " rtcpPort=%d, ipAddr=%s)",
- channel, rtpPort, rtcpPort, ipAddr);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((rtpPort < 0) || (rtpPort > 65535))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSourceFilter() invalid RTP port");
- return -1;
- }
- if ((rtcpPort < 0) || (rtcpPort > 65535))
- {
- _shared->SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSourceFilter() invalid RTCP port");
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSourceFilter() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSourceFilter() external transport is enabled");
- return -1;
- }
- return channelPtr->SetSourceFilter(rtpPort, rtcpPort, ipAddr);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSourceFilter() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSourceFilter(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetSourceFilter(channel=%d, rtpPort=?, rtcpPort=?, "
- "ipAddr[]=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "GetSourceFilter() invalid IP-address buffer");
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSourceFilter() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSourceFilter() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSourceFilter(rtpPort, rtcpPort, ipAddr);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSourceFilter() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::SetSendTOS(int channel,
- int DSCP,
- int priority,
- bool useSetSockopt)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SetSendTOS(channel=%d, DSCP=%d, useSetSockopt=%d)",
- channel, DSCP, useSetSockopt);
-
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
- _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "SetSendTOS() is not supported on this platform");
- return -1;
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((DSCP < 0) || (DSCP > 63))
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() Invalid DSCP value");
- return -1;
- }
-#if defined(_WIN32) || defined(WEBRTC_LINUX)
- if ((priority < -1) || (priority > 7))
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() Invalid priority value");
- return -1;
- }
-#else
- if (-1 != priority)
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() priority not supported");
- return -1;
- }
-#endif
-#if defined(_WIN32)
- if ((priority >= 0) && useSetSockopt)
- {
- // On Windows, priority and useSetSockopt cannot be combined
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() priority and useSetSockopt conflict");
- return -1;
- }
-#endif
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendTOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendTOS() external transport is enabled");
- return -1;
- }
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- useSetSockopt = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
- " force useSetSockopt=true since there is no alternative"
- " implementation");
-#endif
-
- return channelPtr->SetSendTOS(DSCP, priority, useSetSockopt);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSendTOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSendTOS(int channel,
- int& DSCP,
- int& priority,
- bool& useSetSockopt)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetSendTOS(channel=%d)", channel);
-
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
- _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "GetSendTOS() is not supported on this platform");
- return -1;
-#endif
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendTOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendTOS() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSendTOS(DSCP, priority, useSetSockopt);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSendTOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::SetSendGQoS(int channel,
- bool enable,
- int serviceType,
- int overrideDSCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SetSendGQOS(channel=%d, enable=%d, serviceType=%d,"
- " overrideDSCP=%d)",
- channel, (int) enable, serviceType, overrideDSCP);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
- IPHONE_NOT_SUPPORTED(_shared->statistics());
-#if !defined(_WIN32)
- _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "SetSendGQOS() is not supported on this platform");
- return -1;
-#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendGQOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendGQOS() external transport is enabled");
- return -1;
- }
- return channelPtr->SetSendGQoS(enable, serviceType, overrideDSCP);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSendGQOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSendGQoS(int channel,
- bool& enabled,
- int& serviceType,
- int& overrideDSCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "GetSendGQOS(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_shared->statistics());
- IPHONE_NOT_SUPPORTED(_shared->statistics());
-#if !defined(_WIN32)
- _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "GetSendGQOS() is not supported on this platform");
- return -1;
-#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendGQOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendGQOS() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSendGQoS(enabled, serviceType, overrideDSCP);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSendGQOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
int VoENetworkImpl::SetPacketTimeoutNotification(int channel,
bool enable,
int timeoutSeconds)
@@ -821,52 +323,6 @@
sampleTimeSeconds);
}
-int VoENetworkImpl::SendUDPPacket(int channel,
- const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "SendUDPPacket(channel=%d, data=0x%x, length=%u, useRTCP=%d)",
- channel, data, length, useRtcpSocket);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_shared->statistics().Initialized())
- {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == data)
- {
- _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SendUDPPacket() invalid data buffer");
- return -1;
- }
- if (0 == length)
- {
- _shared->SetLastError(VE_INVALID_PACKET, kTraceError,
- "SendUDPPacket() invalid packet size");
- return -1;
- }
- voe::ScopedChannel sc(_shared->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SendUDPPacket() failed to locate channel");
- return -1;
- }
- return channelPtr->SendUDPPacket(data,
- length,
- transmittedBytes,
- useRtcpSocket);
-#else
- _shared->SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SendUDPPacket() VoE is built for external transport");
- return -1;
-#endif
-}
-
#endif // WEBRTC_VOICE_ENGINE_NETWORK_API
} // namespace webrtc
diff --git a/voice_engine/voe_network_impl.h b/voice_engine/voe_network_impl.h
index b159c81..4dbe4f3 100644
--- a/voice_engine/voe_network_impl.h
+++ b/voice_engine/voe_network_impl.h
@@ -34,47 +34,6 @@
const void* data,
unsigned int length);
- virtual int GetSourceInfo(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64]);
-
- virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false);
-
- virtual int EnableIPv6(int channel);
-
- virtual bool IPv6IsEnabled(int channel);
-
- virtual int SetSourceFilter(int channel,
- int rtpPort,
- int rtcpPort,
- const char ipAddr[64] = 0);
-
- virtual int GetSourceFilter(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64]);
-
- virtual int SetSendTOS(int channel,
- int DSCP,
- int priority = -1,
- bool useSetSockopt = false);
-
- virtual int GetSendTOS(int channel,
- int& DSCP,
- int& priority,
- bool& useSetSockopt);
-
- virtual int SetSendGQoS(int channel,
- bool enable,
- int serviceType,
- int overrideDSCP);
-
- virtual int GetSendGQoS(int channel,
- bool& enabled,
- int& serviceType,
- int& overrideDSCP);
-
virtual int SetPacketTimeoutNotification(int channel,
bool enable,
int timeoutSeconds = 2);
@@ -96,12 +55,6 @@
bool& enabled,
int& sampleTimeSeconds);
- virtual int SendUDPPacket(int channel,
- const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket = false);
-
protected:
VoENetworkImpl(voe::SharedData* shared);
virtual ~VoENetworkImpl();
diff --git a/voice_engine/voice_engine_core.gypi b/voice_engine/voice_engine_core.gypi
index 1c28c0d..21bfca4 100644
--- a/voice_engine/voice_engine_core.gypi
+++ b/voice_engine/voice_engine_core.gypi
@@ -20,7 +20,6 @@
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
@@ -134,7 +133,6 @@
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],