// Copyright (c) 2013 The Chromium Authors. All rights reserved. // Use of this source code is governed by a BSD-style license that can be // found in the LICENSE file. #ifndef CONTENT_BROWSER_WEBRTC_WEBRTC_INTERNALS_H_ #define CONTENT_BROWSER_WEBRTC_WEBRTC_INTERNALS_H_ #include #include #include #include "base/containers/hash_tables.h" #include "base/gtest_prod_util.h" #include "base/lazy_instance.h" #include "base/memory/weak_ptr.h" #include "base/observer_list.h" #include "base/process/process.h" #include "base/threading/thread_checker.h" #include "base/values.h" #include "content/common/content_export.h" #include "content/public/browser/render_process_host_observer.h" #include "device/wake_lock/public/interfaces/wake_lock_service.mojom.h" #include "media/media_features.h" #include "ui/shell_dialogs/select_file_dialog.h" namespace content { class WebContents; class WebRTCInternalsUIObserver; // This is a singleton class running in the browser UI thread. // It collects peer connection infomation from the renderers, // forwards the data to WebRTCInternalsUIObserver and // sends data collecting commands to the renderers. class CONTENT_EXPORT WebRTCInternals : public RenderProcessHostObserver, public ui::SelectFileDialog::Listener { public: static WebRTCInternals* GetInstance(); // This method is called when a PeerConnection is created. // |render_process_id| is the id of the render process (not OS pid), which is // needed because we might not be able to get the OS process id when the // render process terminates and we want to clean up. // |pid| is the renderer process id, |lid| is the renderer local id used to // identify a PeerConnection, |url| is the url of the tab owning the // PeerConnection, |rtc_configuration| is the serialized RTCConfiguration, // |constraints| is the media constraints used to initialize the // PeerConnection. void OnAddPeerConnection(int render_process_id, base::ProcessId pid, int lid, const std::string& url, const std::string& rtc_configuration, const std::string& constraints); // This method is called when PeerConnection is destroyed. // |pid| is the renderer process id, |lid| is the renderer local id. void OnRemovePeerConnection(base::ProcessId pid, int lid); // This method is called when a PeerConnection is updated. // |pid| is the renderer process id, |lid| is the renderer local id, // |type| is the update type, |value| is the detail of the update. void OnUpdatePeerConnection(base::ProcessId pid, int lid, const std::string& type, const std::string& value); // This method is called when results from PeerConnectionInterface::GetStats // are available. |pid| is the renderer process id, |lid| is the renderer // local id, |value| is the list of stats reports. void OnAddStats(base::ProcessId pid, int lid, const base::ListValue& value); // This method is called when getUserMedia is called. |render_process_id| is // the id of the render process (not OS pid), which is needed because we might // not be able to get the OS process id when the render process terminates and // we want to clean up. |pid| is the renderer OS process id, |origin| is the // security origin of the getUserMedia call, |audio| is true if audio stream // is requested, |video| is true if the video stream is requested, // |audio_constraints| is the constraints for the audio, |video_constraints| // is the constraints for the video. void OnGetUserMedia(int render_process_id, base::ProcessId pid, const std::string& origin, bool audio, bool video, const std::string& audio_constraints, const std::string& video_constraints); // Methods for adding or removing WebRTCInternalsUIObserver. void AddObserver(WebRTCInternalsUIObserver* observer); void RemoveObserver(WebRTCInternalsUIObserver* observer); // Sends all update data to |observer|. void UpdateObserver(WebRTCInternalsUIObserver* observer); // Enables or disables diagnostic audio recordings for debugging purposes. void EnableAudioDebugRecordings(content::WebContents* web_contents); void DisableAudioDebugRecordings(); bool IsAudioDebugRecordingsEnabled() const; const base::FilePath& GetAudioDebugRecordingsFilePath() const; // Enables or disables diagnostic event log. void EnableEventLogRecordings(content::WebContents* web_contents); void DisableEventLogRecordings(); bool IsEventLogRecordingsEnabled() const; const base::FilePath& GetEventLogFilePath() const; int num_open_connections() const { return num_open_connections_; } protected: // Constructor/Destructor are protected to allow tests to derive from the // class and do per-instance testing without having to use the global // instance. // The default ctor sets |aggregate_updates_ms| to 500ms. WebRTCInternals(); WebRTCInternals(int aggregate_updates_ms, bool should_block_power_saving); ~WebRTCInternals() override; device::mojom::WakeLockServicePtr wake_lock_service_; private: friend struct base::LazyInstanceTraitsBase; FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest, CallWithAudioDebugRecordings); FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest, CallWithAudioDebugRecordingsEnabledThenDisabled); FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest, TwoCallsWithAudioDebugRecordings); FRIEND_TEST_ALL_PREFIXES(WebRtcInternalsTest, AudioDebugRecordingsFileSelectionCanceled); void SendUpdate(const char* command, std::unique_ptr value); // RenderProcessHostObserver implementation. void RenderProcessExited(RenderProcessHost* host, base::TerminationStatus status, int exit_code) override; // ui::SelectFileDialog::Listener implementation. void FileSelected(const base::FilePath& path, int index, void* unused_params) override; void FileSelectionCanceled(void* params) override; // Called when a renderer exits (including crashes). void OnRendererExit(int render_process_id); #if BUILDFLAG(ENABLE_WEBRTC) // Enables diagnostic audio recordings on all render process hosts using // |audio_debug_recordings_file_path_|. void EnableAudioDebugRecordingsOnAllRenderProcessHosts(); // Enables event log recordings on all render process hosts using // |event_log_recordings_file_path_|. void EnableEventLogRecordingsOnAllRenderProcessHosts(); #endif // Updates the number of open PeerConnections. Called when a PeerConnection // is stopped or removed. void MaybeClosePeerConnection(base::DictionaryValue* record); // Called whenever a PeerConnection is created or stopped in order to // request/cancel a wake lock on suspending the current application for power // saving. void UpdateWakeLock(); device::mojom::WakeLockService* GetWakeLockService(); // Called on a timer to deliver updates to javascript. // We throttle and bulk together updates to avoid DOS like scenarios where // a page uses a lot of peerconnection instances with many event // notifications. void ProcessPendingUpdates(); base::DictionaryValue* FindRecord(base::ProcessId pid, int lid, size_t* index = nullptr); base::ObserverList observers_; // |peer_connection_data_| is a list containing all the PeerConnection // updates. // Each item of the list represents the data for one PeerConnection, which // contains these fields: // "rid" -- the renderer id. // "pid" -- OS process id of the renderer that creates the PeerConnection. // "lid" -- local Id assigned to the PeerConnection. // "url" -- url of the web page that created the PeerConnection. // "servers" and "constraints" -- server configuration and media constraints // used to initialize the PeerConnection respectively. // "log" -- a ListValue contains all the updates for the PeerConnection. Each // list item is a DictionaryValue containing "time", which is the number of // milliseconds since epoch as a string, and "type" and "value", both of which // are strings representing the event. base::ListValue peer_connection_data_; // A list of getUserMedia requests. Each item is a DictionaryValue that // contains these fields: // "rid" -- the renderer id. // "pid" -- proceddId of the renderer. // "origin" -- the security origin of the request. // "audio" -- the serialized audio constraints if audio is requested. // "video" -- the serialized video constraints if video is requested. base::ListValue get_user_media_requests_; // For managing select file dialog. scoped_refptr select_file_dialog_; // Diagnostic audio recording state. bool audio_debug_recordings_; base::FilePath audio_debug_recordings_file_path_; // Diagnostic event log recording state. bool event_log_recordings_; bool selecting_event_log_; base::FilePath event_log_recordings_file_path_; // While |num_open_connections_| is greater than zero, request a wake lock // service. This prevents the application from being suspended while remoting. int num_open_connections_; const bool should_block_power_saving_; // Set of render process hosts that |this| is registered as an observer on. base::hash_set render_process_id_set_; // Used to bulk up updates that we send to javascript. // The class owns the value/dictionary and command name of an update. // For each update, a PendingUpdate is stored in the |pending_updates_| queue // and deleted as soon as the update has been delivered. // The class is moveble and not copyable to avoid copying while still allowing // us to use an stl container without needing scoped_ptr or similar. // The class is single threaded, so all operations must occur on the same // thread. class PendingUpdate { public: PendingUpdate(const char* command, std::unique_ptr value); PendingUpdate(PendingUpdate&& other); ~PendingUpdate(); const char* command() const; const base::Value* value() const; private: base::ThreadChecker thread_checker_; const char* command_; std::unique_ptr value_; DISALLOW_COPY_AND_ASSIGN(PendingUpdate); }; std::queue pending_updates_; const int aggregate_updates_ms_; // Weak factory for this object that we use for bulking up updates. base::WeakPtrFactory weak_factory_; }; } // namespace content #endif // CONTENT_BROWSER_WEBRTC_WEBRTC_INTERNALS_H_