49a7843030
This CL changes the behavior for RunFor and RunUntil so they do not anymore restart the underlying streams every time they are called. This has a side effect on the semantics of the calls. Previously, both RunUntil and RunFor would restart the session and run until the given time had passed. Now RunFor will still run for the provided duration, however, to make the name of RunUntil more correct, it will run until the time since start is equal to the max_duration parameter. An extra overload of RunUntil was added to allow using this behavior without providing an ending condition. Bug: webrtc:9510 Change-Id: I9fe56a44116907fba3d102894b5c96af2ba6cffb Reviewed-on: https://webrtc-review.googlesource.com/c/111502 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25726}
95 lines
3.0 KiB
C++
95 lines
3.0 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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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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RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveAudioStream);
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~ReceiveAudioStream();
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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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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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