webrtc_m130/test/rtp_rtcp_observer.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 TEST_RTP_RTCP_OBSERVER_H_
#define TEST_RTP_RTCP_OBSERVER_H_
#include <map>
#include <memory>
Reland "Delete test/constants.h" This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6. Reason for revert: Failing tests fixed. Original change's description: > Revert "Delete test/constants.h" > > This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de. > > Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate > > Original change's description: > > Delete test/constants.h > > > > It's not possible to use constants.h for all RTP extensions > > after the number of extensions exceeds 14, which is the maximum > > number of one-byte RTP extensions. This is because some extensions > > would have to be assigned a number greater than 14, even if the > > test only involves 14 extensions or less. > > > > For uniformity's sake, this CL also edits some files to use an > > enum as the files involved in this CL, rather than free-floating > > const-ints. > > > > Bug: webrtc:10288 > > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123048 > > Commit-Queue: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26728} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org > > Bug: webrtc:10288, chromium:933127 > Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/123381 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26744} TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954 Bug: webrtc:10288, chromium:933127 Reviewed-on: https://webrtc-review.googlesource.com/c/123384 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-18 23:45:57 +01:00
#include <utility>
#include <vector>
#include "api/test/simulated_network.h"
#include "call/simulated_packet_receiver.h"
#include "call/video_send_stream.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
#include "system_wrappers/include/field_trial.h"
#include "test/direct_transport.h"
#include "test/gtest.h"
#include "test/rtp_header_parser.h"
namespace {
const int kShortTimeoutMs = 500;
}
namespace webrtc {
namespace test {
class PacketTransport;
class RtpRtcpObserver {
public:
enum Action {
SEND_PACKET,
DROP_PACKET,
};
virtual ~RtpRtcpObserver() {}
virtual bool Wait() {
if (field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
observation_complete_.Wait(kShortTimeoutMs);
return true;
}
return observation_complete_.Wait(timeout_ms_);
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
protected:
RtpRtcpObserver() : RtpRtcpObserver(0) {}
explicit RtpRtcpObserver(int event_timeout_ms)
: parser_(RtpHeaderParser::CreateForTest()),
timeout_ms_(event_timeout_ms) {}
rtc::Event observation_complete_;
const std::unique_ptr<RtpHeaderParser> parser_;
private:
const int timeout_ms_;
};
class PacketTransport : public test::DirectTransport {
public:
enum TransportType { kReceiver, kSender };
PacketTransport(TaskQueueBase* task_queue,
Call* send_call,
RtpRtcpObserver* observer,
TransportType transport_type,
const std::map<uint8_t, MediaType>& payload_type_map,
std::unique_ptr<SimulatedPacketReceiverInterface> nw_pipe)
: test::DirectTransport(task_queue,
std::move(nw_pipe),
send_call,
payload_type_map),
observer_(observer),
transport_type_(transport_type) {}
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
RtpRtcpObserver::Action action;
{
if (transport_type_ == kSender) {
action = observer_->OnSendRtp(packet, length);
} else {
action = observer_->OnReceiveRtp(packet, length);
}
}
switch (action) {
case RtpRtcpObserver::DROP_PACKET:
// Drop packet silently.
return true;
case RtpRtcpObserver::SEND_PACKET:
return test::DirectTransport::SendRtp(packet, length, options);
}
return true; // Will never happen, makes compiler happy.
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
RtpRtcpObserver::Action action;
{
if (transport_type_ == kSender) {
action = observer_->OnSendRtcp(packet, length);
} else {
action = observer_->OnReceiveRtcp(packet, length);
}
}
switch (action) {
case RtpRtcpObserver::DROP_PACKET:
// Drop packet silently.
return true;
case RtpRtcpObserver::SEND_PACKET:
return test::DirectTransport::SendRtcp(packet, length);
}
return true; // Will never happen, makes compiler happy.
}
RtpRtcpObserver* const observer_;
TransportType transport_type_;
};
} // namespace test
} // namespace webrtc
#endif // TEST_RTP_RTCP_OBSERVER_H_