1 files changed, 70 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/audio_track.cc b/third_party/libwebrtc/pc/audio_track.cc
new file mode 100644
index 0000000000..c012442d13
--- /dev/null
+++ b/third_party/libwebrtc/pc/audio_track.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright 2011 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.
+ */
+
+#include "pc/audio_track.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// static
+rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
+    absl::string_view id,
+    const rtc::scoped_refptr<AudioSourceInterface>& source) {
+  return rtc::make_ref_counted<AudioTrack>(id, source);
+}
+
+AudioTrack::AudioTrack(absl::string_view label,
+                       const rtc::scoped_refptr<AudioSourceInterface>& source)
+    : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
+  if (audio_source_) {
+    audio_source_->RegisterObserver(this);
+    OnChanged();
+  }
+}
+
+AudioTrack::~AudioTrack() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  set_state(MediaStreamTrackInterface::kEnded);
+  if (audio_source_)
+    audio_source_->UnregisterObserver(this);
+}
+
+std::string AudioTrack::kind() const {
+  return kAudioKind;
+}
+
+AudioSourceInterface* AudioTrack::GetSource() const {
+  // Callable from any thread.
+  return audio_source_.get();
+}
+
+void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  if (audio_source_)
+    audio_source_->AddSink(sink);
+}
+
+void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  if (audio_source_)
+    audio_source_->RemoveSink(sink);
+}
+
+void AudioTrack::OnChanged() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  if (audio_source_->state() == MediaSourceInterface::kEnded) {
+    set_state(kEnded);
+  } else {
+    set_state(kLive);
+  }
+}
+
+}  // namespace webrtc