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, 91 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/remote_audio_source.h b/third_party/libwebrtc/pc/remote_audio_source.h
new file mode 100644
index 0000000000..0fac606ad4
--- /dev/null
+++ b/third_party/libwebrtc/pc/remote_audio_source.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright 2014 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 PC_REMOTE_AUDIO_SOURCE_H_
+#define PC_REMOTE_AUDIO_SOURCE_H_
+
+#include <stdint.h>
+
+#include <list>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/call/audio_sink.h"
+#include "api/media_stream_interface.h"
+#include "api/notifier.h"
+#include "api/task_queue/task_queue_base.h"
+#include "media/base/media_channel.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// This class implements the audio source used by the remote audio track.
+// This class works by configuring itself as a sink with the underlying media
+// engine, then when receiving data will fan out to all added sinks.
+class RemoteAudioSource : public Notifier<AudioSourceInterface> {
+ public:
+  // In Unified Plan, receivers map to m= sections and their tracks and sources
+  // survive SSRCs being reconfigured. The life cycle of the remote audio source
+  // is associated with the life cycle of the m= section, and thus even if an
+  // audio channel is destroyed the RemoteAudioSource should kSurvive.
+  //
+  // In Plan B however, remote audio sources map 1:1 with an SSRCs and if an
+  // audio channel is destroyed, the RemoteAudioSource should kEnd.
+  enum class OnAudioChannelGoneAction {
+    kSurvive,
+    kEnd,
+  };
+
+  explicit RemoteAudioSource(
+      TaskQueueBase* worker_thread,
+      OnAudioChannelGoneAction on_audio_channel_gone_action);
+
+  // Register and unregister remote audio source with the underlying media
+  // engine.
+  void Start(cricket::VoiceMediaReceiveChannelInterface* media_channel,
+             absl::optional<uint32_t> ssrc);
+  void Stop(cricket::VoiceMediaReceiveChannelInterface* media_channel,
+            absl::optional<uint32_t> ssrc);
+  void SetState(SourceState new_state);
+
+  // MediaSourceInterface implementation.
+  MediaSourceInterface::SourceState state() const override;
+  bool remote() const override;
+
+  // AudioSourceInterface implementation.
+  void SetVolume(double volume) override;
+  void RegisterAudioObserver(AudioObserver* observer) override;
+  void UnregisterAudioObserver(AudioObserver* observer) override;
+
+  void AddSink(AudioTrackSinkInterface* sink) override;
+  void RemoveSink(AudioTrackSinkInterface* sink) override;
+
+ protected:
+  ~RemoteAudioSource() override;
+
+ private:
+  // These are callbacks from the media engine.
+  class AudioDataProxy;
+
+  void OnData(const AudioSinkInterface::Data& audio);
+  void OnAudioChannelGone();
+
+  TaskQueueBase* const main_thread_;
+  TaskQueueBase* const worker_thread_;
+  const OnAudioChannelGoneAction on_audio_channel_gone_action_;
+  std::list<AudioObserver*> audio_observers_;
+  Mutex sink_lock_;
+  std::list<AudioTrackSinkInterface*> sinks_;
+  SourceState state_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_REMOTE_AUDIO_SOURCE_H_