webrtc_m130/webrtc/test/direct_transport.h

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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.
*/
#ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
#define WEBRTC_TEST_DIRECT_TRANSPORT_H_
#include <assert.h>
#include <memory>
#include "webrtc/api/call/transport.h"
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d > > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057 TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-29 23:57:43 -07:00
#include "webrtc/call/call.h"
#include "webrtc/rtc_base/sequenced_task_checker.h"
#include "webrtc/rtc_base/thread_annotations.h"
#include "webrtc/test/fake_network_pipe.h"
#include "webrtc/test/single_threaded_task_queue.h"
namespace webrtc {
class Clock;
class PacketReceiver;
namespace test {
// Objects of this class are expected to be allocated and destroyed on the
// same task-queue - the one that's passed in via the constructor.
class DirectTransport : public Transport {
public:
DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map);
DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
const FakeNetworkPipe::Config& config,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map);
DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
const FakeNetworkPipe::Config& config,
Call* send_call,
std::unique_ptr<Demuxer> demuxer);
~DirectTransport() override;
void SetConfig(const FakeNetworkPipe::Config& config);
RTC_DEPRECATED void StopSending();
// TODO(holmer): Look into moving this to the constructor.
virtual void SetReceiver(PacketReceiver* receiver);
bool SendRtp(const uint8_t* data,
size_t length,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t length) override;
int GetAverageDelayMs();
private:
void SendPackets();
Call* const send_call_;
Clock* const clock_;
// TODO(eladalon): Make |task_queue_| const.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8125
SingleThreadedTaskQueueForTesting* task_queue_;
SingleThreadedTaskQueueForTesting::TaskId next_scheduled_task_
GUARDED_BY(&sequence_checker_);
FakeNetworkPipe fake_network_;
rtc::SequencedTaskChecker sequence_checker_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_DIRECT_TRANSPORT_H_