10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
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/constructor_magic.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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