| /* |
| * 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/voice_engine/test/channel_transport/channel_transport.h" |
| |
| #include <stdio.h> |
| |
| #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) |
| #include "webrtc/test/gtest.h" |
| #endif |
| #include "webrtc/voice_engine/test/channel_transport/udp_transport.h" |
| #include "webrtc/voice_engine/include/voe_network.h" |
| |
| #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) |
| #undef NDEBUG |
| #include <assert.h> |
| #endif |
| |
| namespace webrtc { |
| namespace test { |
| |
| VoiceChannelTransport::VoiceChannelTransport(VoENetwork* voe_network, |
| int channel) |
| : channel_(channel), |
| voe_network_(voe_network) { |
| uint8_t socket_threads = 1; |
| socket_transport_ = UdpTransport::Create(channel, socket_threads); |
| int registered = voe_network_->RegisterExternalTransport(channel, |
| *socket_transport_); |
| #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) |
| EXPECT_EQ(0, registered); |
| #else |
| assert(registered == 0); |
| #endif |
| } |
| |
| VoiceChannelTransport::~VoiceChannelTransport() { |
| voe_network_->DeRegisterExternalTransport(channel_); |
| UdpTransport::Destroy(socket_transport_); |
| } |
| |
| void VoiceChannelTransport::IncomingRTPPacket( |
| const int8_t* incoming_rtp_packet, |
| const size_t packet_length, |
| const char* /*from_ip*/, |
| const uint16_t /*from_port*/) { |
| voe_network_->ReceivedRTPPacket( |
| channel_, incoming_rtp_packet, packet_length, PacketTime()); |
| } |
| |
| void VoiceChannelTransport::IncomingRTCPPacket( |
| const int8_t* incoming_rtcp_packet, |
| const size_t packet_length, |
| const char* /*from_ip*/, |
| const uint16_t /*from_port*/) { |
| voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, |
| packet_length); |
| } |
| |
| int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) { |
| static const int kNumReceiveSocketBuffers = 500; |
| int return_value = socket_transport_->InitializeReceiveSockets(this, |
| rtp_port); |
| if (return_value == 0) { |
| return socket_transport_->StartReceiving(kNumReceiveSocketBuffers); |
| } |
| return return_value; |
| } |
| |
| int VoiceChannelTransport::SetSendDestination(const char* ip_address, |
| uint16_t rtp_port) { |
| return socket_transport_->InitializeSendSockets(ip_address, rtp_port); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |