Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 110 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/scenario/audio_stream.h b/third_party/libwebrtc/test/scenario/audio_stream.h
new file mode 100644
index 0000000000..cbaf9d29eb
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/audio_stream.h
@@ -0,0 +1,110 @@
+/*
+ *  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_