/*
* 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 "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:
~SendAudioStream();
SendAudioStream(
const SendAudioStream&) =
delete;
SendAudioStream&
operator=(
const SendAudioStream&) =
delete;
void Start();
void Stop();
void SetMuted(
bool mute);
ColumnPrinter StatsPrinter();
private:
friend class Scenario;
friend class AudioStreamPair;
friend class ReceiveAudioStream;
SendAudioStream(CallClient* sender,
AudioStreamConfig config,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
Transport* send_transport);
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:
~ReceiveAudioStream();
ReceiveAudioStream(
const ReceiveAudioStream&) =
delete;
ReceiveAudioStream&
operator=(
const ReceiveAudioStream&) =
delete;
void Start();
void Stop();
AudioReceiveStreamInterface::Stats GetStats()
const;
private:
friend class Scenario;
friend class AudioStreamPair;
ReceiveAudioStream(CallClient* receiver,
AudioStreamConfig config,
SendAudioStream* send_stream,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
Transport* feedback_transport);
AudioReceiveStreamInterface* 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:
~AudioStreamPair();
AudioStreamPair(
const AudioStreamPair&) =
delete;
AudioStreamPair&
operator=(
const AudioStreamPair&) =
delete;
SendAudioStream* send() {
return &send_stream_; }
ReceiveAudioStream* receive() {
return &receive_stream_; }
private:
friend class Scenario;
AudioStreamPair(CallClient* sender,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
CallClient* receiver,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
AudioStreamConfig config);
private:
const AudioStreamConfig config_;
SendAudioStream send_stream_;
ReceiveAudioStream receive_stream_;
};
std::vector<RtpExtension> GetAudioRtpExtensions(
const AudioStreamConfig& config);
}
// namespace test
}
// namespace webrtc
#endif // TEST_SCENARIO_AUDIO_STREAM_H_
