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// 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 <memory>
#include <string>
#include "base/containers/hash_tables.h"
#include "base/containers/queue.h"
#include "base/gtest_prod_util.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/browser/webrtc/webrtc_event_log_manager.h"
#include "content/common/content_export.h"
#include "content/public/browser/render_process_host_observer.h"
#include "media/media_features.h"
#include "services/device/public/interfaces/wake_lock.mojom.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:
// Ensures that no previous instantiation of the class was performed, then
// instantiates the class and returns the object. Subsequent calls to
// GetInstance() will return this object.
static WebRTCInternals* CreateSingletonInstance();
// Returns the object previously constructed using CreateSingletonInstance().
// Can be null in tests.
static WebRTCInternals* GetInstance();
~WebRTCInternals() override;
// 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 EnableLocalEventLogRecordings(content::WebContents* web_contents);
void DisableLocalEventLogRecordings();
bool IsEventLogRecordingsEnabled() 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);
device::mojom::WakeLockPtr wake_lock_;
private:
FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest,
CallWithAudioDebugRecordings);
FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest,
CallWithAudioDebugRecordingsEnabledThenDisabled);
FRIEND_TEST_ALL_PREFIXES(WebRtcAudioDebugRecordingsBrowserTest,
TwoCallsWithAudioDebugRecordings);
FRIEND_TEST_ALL_PREFIXES(WebRtcInternalsTest,
AudioDebugRecordingsFileSelectionCanceled);
static WebRTCInternals* g_webrtc_internals;
void SendUpdate(const char* command,
std::unique_ptr<base::Value> 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();
#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::WakeLock* GetWakeLock();
// 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<WebRTCInternalsUIObserver> 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<ui::SelectFileDialog> select_file_dialog_;
enum class SelectionType {
kRtcEventLogs,
kAudioDebugRecordings
} selection_type_;
// Diagnostic audio recording state.
bool audio_debug_recordings_;
base::FilePath audio_debug_recordings_file_path_;
// Diagnostic event log recording state.
bool event_log_recordings_;
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<int> 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<base::Value> 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<base::Value> value_;
DISALLOW_COPY_AND_ASSIGN(PendingUpdate);
};
base::queue<PendingUpdate> pending_updates_;
const int aggregate_updates_ms_;
// Weak factory for this object that we use for bulking up updates.
base::WeakPtrFactory<WebRTCInternals> weak_factory_;
};
} // namespace content
#endif // CONTENT_BROWSER_WEBRTC_WEBRTC_INTERNALS_H_