Repo created

TMessagesProj/jni/voip/webrtc/pc/remote_audio_source.cc

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+/*
+ *  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.
+ */
+
+#include "pc/remote_audio_source.h"
+
+#include <stddef.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_format.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+// This proxy is passed to the underlying media engine to receive audio data as
+// they come in. The data will then be passed back up to the RemoteAudioSource
+// which will fan it out to all the sinks that have been added to it.
+class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
+ public:
+  explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
+    RTC_DCHECK(source);
+  }
+
+  AudioDataProxy() = delete;
+  AudioDataProxy(const AudioDataProxy&) = delete;
+  AudioDataProxy& operator=(const AudioDataProxy&) = delete;
+
+  ~AudioDataProxy() override { source_->OnAudioChannelGone(); }
+
+  // AudioSinkInterface implementation.
+  void OnData(const AudioSinkInterface::Data& audio) override {
+    source_->OnData(audio);
+  }
+
+ private:
+  const rtc::scoped_refptr<RemoteAudioSource> source_;
+};
+
+RemoteAudioSource::RemoteAudioSource(
+    TaskQueueBase* worker_thread,
+    OnAudioChannelGoneAction on_audio_channel_gone_action)
+    : main_thread_(TaskQueueBase::Current()),
+      worker_thread_(worker_thread),
+      on_audio_channel_gone_action_(on_audio_channel_gone_action),
+      state_(MediaSourceInterface::kInitializing) {
+  RTC_DCHECK(main_thread_);
+  RTC_DCHECK(worker_thread_);
+}
+
+RemoteAudioSource::~RemoteAudioSource() {
+  RTC_DCHECK(audio_observers_.empty());
+  if (!sinks_.empty()) {
+    RTC_LOG(LS_WARNING)
+        << "RemoteAudioSource destroyed while sinks_ is non-empty.";
+  }
+}
+
+void RemoteAudioSource::Start(
+    cricket::VoiceMediaReceiveChannelInterface* media_channel,
+    absl::optional<uint32_t> ssrc) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  // Register for callbacks immediately before AddSink so that we always get
+  // notified when a channel goes out of scope (signaled when "AudioDataProxy"
+  // is destroyed).
+  RTC_DCHECK(media_channel);
+  ssrc ? media_channel->SetRawAudioSink(*ssrc,
+                                        std::make_unique<AudioDataProxy>(this))
+       : media_channel->SetDefaultRawAudioSink(
+             std::make_unique<AudioDataProxy>(this));
+}
+
+void RemoteAudioSource::Stop(
+    cricket::VoiceMediaReceiveChannelInterface* media_channel,
+    absl::optional<uint32_t> ssrc) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(media_channel);
+  ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
+       : media_channel->SetDefaultRawAudioSink(nullptr);
+}
+
+void RemoteAudioSource::SetState(SourceState new_state) {
+  RTC_DCHECK_RUN_ON(main_thread_);
+  if (state_ != new_state) {
+    state_ = new_state;
+    FireOnChanged();
+  }
+}
+
+MediaSourceInterface::SourceState RemoteAudioSource::state() const {
+  RTC_DCHECK_RUN_ON(main_thread_);
+  return state_;
+}
+
+bool RemoteAudioSource::remote() const {
+  RTC_DCHECK_RUN_ON(main_thread_);
+  return true;
+}
+
+void RemoteAudioSource::SetVolume(double volume) {
+  RTC_DCHECK_GE(volume, 0);
+  RTC_DCHECK_LE(volume, 10);
+  RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__,
+                                        volume);
+  for (auto* observer : audio_observers_) {
+    observer->OnSetVolume(volume);
+  }
+}
+
+void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
+  RTC_DCHECK(observer != NULL);
+  RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
+  audio_observers_.push_back(observer);
+}
+
+void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
+  RTC_DCHECK(observer != NULL);
+  audio_observers_.remove(observer);
+}
+
+void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(main_thread_);
+  RTC_DCHECK(sink);
+
+  MutexLock lock(&sink_lock_);
+  RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
+  sinks_.push_back(sink);
+}
+
+void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(main_thread_);
+  RTC_DCHECK(sink);
+
+  MutexLock lock(&sink_lock_);
+  sinks_.remove(sink);
+}
+
+void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
+  // Called on the externally-owned audio callback thread, via/from webrtc.
+  TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData");
+  MutexLock lock(&sink_lock_);
+  for (auto* sink : sinks_) {
+    // When peerconnection acts as an audio source, it should not provide
+    // absolute capture timestamp.
+    sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
+                 audio.samples_per_channel,
+                 /*absolute_capture_timestamp_ms=*/absl::nullopt);
+  }
+}
+
+void RemoteAudioSource::OnAudioChannelGone() {
+  if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
+    return;
+  }
+  // Called when the audio channel is deleted. It may be the worker thread or
+  // may be a different task queue.
+  // This object needs to live long enough for the cleanup logic in the posted
+  // task to run, so take a reference to it. Sometimes the task may not be
+  // processed (because the task queue was destroyed shortly after this call),
+  // but that is fine because the task queue destructor will take care of
+  // destroying task which will release the reference on RemoteAudioSource.
+  rtc::scoped_refptr<RemoteAudioSource> thiz(this);
+  main_thread_->PostTask([thiz = std::move(thiz)] {
+    thiz->sinks_.clear();
+    thiz->SetState(MediaSourceInterface::kEnded);
+  });
+}
+
+}  // namespace webrtc