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/*
* 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/channel_transport/include/channel_transport.h"
#include <stdio.h>
#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
#include "testing/gtest/include/gtest/gtest.h"
#endif
#include "webrtc/test/channel_transport/udp_transport.h"
#include "webrtc/video_engine/vie_defines.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) {
int return_value = socket_transport_->InitializeReceiveSockets(this,
rtp_port);
if (return_value == 0) {
return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers);
}
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
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