800e121dca
This CL makes it possible to change transport routes while running a scenario based test. To make this possible in a consistent manner, the scenario test framework is modified to only allow shared transport for all streams between two CallClients. This is what typically is done in practice and it is quite complex to even reason about the implications of using mixed transports for a single call. Bug: webrtc:9718 Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b Reviewed-on: https://webrtc-review.googlesource.com/c/107200 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25287}
93 lines
2.9 KiB
C++
93 lines
2.9 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 "rtc_base/constructormagic.h"
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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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RTC_DISALLOW_COPY_AND_ASSIGN(SendAudioStream);
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~SendAudioStream();
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void Start();
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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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RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveAudioStream);
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~ReceiveAudioStream();
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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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AudioReceiveStream* 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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RTC_DISALLOW_COPY_AND_ASSIGN(AudioStreamPair);
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~AudioStreamPair();
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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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} // namespace test
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} // namespace webrtc
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#endif // TEST_SCENARIO_AUDIO_STREAM_H_
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