blob: 17b8bc370efd3d1921598b860fb0f87b30238926 [file] [log] [blame]
/*
* Copyright 2018 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_SCENARIO_AUDIO_STREAM_H_
#define TEST_SCENARIO_AUDIO_STREAM_H_
#include <memory>
#include <string>
#include <vector>
#include "rtc_base/constructormagic.h"
#include "test/scenario/call_client.h"
#include "test/scenario/column_printer.h"
#include "test/scenario/network_node.h"
#include "test/scenario/scenario_config.h"
namespace webrtc {
namespace test {
// SendAudioStream represents sending of audio. It can be used for starting the
// stream if neccessary.
class SendAudioStream : public NetworkReceiverInterface {
public:
RTC_DISALLOW_COPY_AND_ASSIGN(SendAudioStream);
~SendAudioStream();
void Start();
private:
friend class Scenario;
friend class AudioStreamPair;
friend class ReceiveAudioStream;
SendAudioStream(CallClient* sender,
AudioStreamConfig config,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
Transport* send_transport);
// Handles RTCP feedback for this stream.
bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
uint64_t receiver,
Timestamp at_time) override;
AudioSendStream* send_stream_ = nullptr;
CallClient* const sender_;
const AudioStreamConfig config_;
uint32_t ssrc_;
};
// ReceiveAudioStream represents an audio receiver. It can't be used directly.
class ReceiveAudioStream : public NetworkReceiverInterface {
public:
RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveAudioStream);
~ReceiveAudioStream();
private:
friend class Scenario;
friend class AudioStreamPair;
ReceiveAudioStream(CallClient* receiver,
AudioStreamConfig config,
SendAudioStream* send_stream,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
Transport* feedback_transport);
bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
uint64_t receiver,
Timestamp at_time) override;
AudioReceiveStream* receive_stream_ = nullptr;
CallClient* const receiver_;
const AudioStreamConfig config_;
};
// AudioStreamPair represents an audio streaming session. It can be used to
// access underlying send and receive classes. It can also be used in calls to
// the Scenario class.
class AudioStreamPair {
public:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioStreamPair);
~AudioStreamPair();
SendAudioStream* send() { return &send_stream_; }
ReceiveAudioStream* receive() { return &receive_stream_; }
private:
friend class Scenario;
AudioStreamPair(CallClient* sender,
std::vector<NetworkNode*> send_link,
uint64_t send_receiver_id,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
CallClient* receiver,
std::vector<NetworkNode*> return_link,
uint64_t return_receiver_id,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
AudioStreamConfig config);
private:
const AudioStreamConfig config_;
std::vector<NetworkNode*> send_link_;
std::vector<NetworkNode*> return_link_;
NetworkNodeTransport send_transport_;
NetworkNodeTransport return_transport_;
SendAudioStream send_stream_;
ReceiveAudioStream receive_stream_;
};
} // namespace test
} // namespace webrtc
#endif // TEST_SCENARIO_AUDIO_STREAM_H_