chromium/third_party/webrtc/test/direct_transport.cc

/*
 *  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 "test/direct_transport.h"

#include "api/media_types.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "rtc_base/checks.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/time_utils.h"

namespace webrtc {
namespace test {

Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
    :{}

MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
                                const size_t packet_length) const {}

DirectTransport::DirectTransport(
    TaskQueueBase* task_queue,
    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
    Call* send_call,
    const std::map<uint8_t, MediaType>& payload_type_map,
    rtc::ArrayView<const RtpExtension> audio_extensions,
    rtc::ArrayView<const RtpExtension> video_extensions)
    :{}

DirectTransport::~DirectTransport() {}

void DirectTransport::SetReceiver(PacketReceiver* receiver) {}

bool DirectTransport::SendRtp(rtc::ArrayView<const uint8_t> data,
                              const PacketOptions& options) {}

bool DirectTransport::SendRtcp(rtc::ArrayView<const uint8_t> data) {}

int DirectTransport::GetAverageDelayMs() {}

void DirectTransport::Start() {}

void DirectTransport::ProcessPackets() {}
}  // namespace test
}  // namespace webrtc