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b3046c25aa
Change-Id: I36db102d023e4b716ce33a0afcff38b79b59b622 Bug: webrtc:7135 Change-Id: I36db102d023e4b716ce33a0afcff38b79b59b622 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290883 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39090}
110 lines
3.4 KiB
C++
110 lines
3.4 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_SCENARIO_AUDIO_STREAM_H_
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#define TEST_SCENARIO_AUDIO_STREAM_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "test/scenario/call_client.h"
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#include "test/scenario/column_printer.h"
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#include "test/scenario/network_node.h"
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#include "test/scenario/scenario_config.h"
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namespace webrtc {
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namespace test {
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// SendAudioStream represents sending of audio. It can be used for starting the
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// stream if neccessary.
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class SendAudioStream {
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public:
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~SendAudioStream();
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SendAudioStream(const SendAudioStream&) = delete;
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SendAudioStream& operator=(const SendAudioStream&) = delete;
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void Start();
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void Stop();
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void SetMuted(bool mute);
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ColumnPrinter StatsPrinter();
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private:
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friend class Scenario;
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friend class AudioStreamPair;
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friend class ReceiveAudioStream;
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SendAudioStream(CallClient* sender,
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AudioStreamConfig config,
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rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
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Transport* send_transport);
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AudioSendStream* send_stream_ = nullptr;
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CallClient* const sender_;
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const AudioStreamConfig config_;
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uint32_t ssrc_;
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};
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// ReceiveAudioStream represents an audio receiver. It can't be used directly.
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class ReceiveAudioStream {
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public:
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~ReceiveAudioStream();
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ReceiveAudioStream(const ReceiveAudioStream&) = delete;
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ReceiveAudioStream& operator=(const ReceiveAudioStream&) = delete;
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void Start();
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void Stop();
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AudioReceiveStreamInterface::Stats GetStats() const;
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private:
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friend class Scenario;
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friend class AudioStreamPair;
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ReceiveAudioStream(CallClient* receiver,
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AudioStreamConfig config,
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SendAudioStream* send_stream,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
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Transport* feedback_transport);
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AudioReceiveStreamInterface* receive_stream_ = nullptr;
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CallClient* const receiver_;
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const AudioStreamConfig config_;
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};
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// AudioStreamPair represents an audio streaming session. It can be used to
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// access underlying send and receive classes. It can also be used in calls to
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// the Scenario class.
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class AudioStreamPair {
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public:
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~AudioStreamPair();
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AudioStreamPair(const AudioStreamPair&) = delete;
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AudioStreamPair& operator=(const AudioStreamPair&) = delete;
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SendAudioStream* send() { return &send_stream_; }
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ReceiveAudioStream* receive() { return &receive_stream_; }
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private:
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friend class Scenario;
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AudioStreamPair(CallClient* sender,
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rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
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CallClient* receiver,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
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AudioStreamConfig config);
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private:
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const AudioStreamConfig config_;
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SendAudioStream send_stream_;
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ReceiveAudioStream receive_stream_;
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};
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std::vector<RtpExtension> GetAudioRtpExtensions(
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const AudioStreamConfig& config);
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} // namespace test
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} // namespace webrtc
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#endif // TEST_SCENARIO_AUDIO_STREAM_H_
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