blob: 513c74bdebb312bd19d94f407e3fa97d55172a6a [file] [log] [blame]
// 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.
#include "content/browser/webrtc/webrtc_internals.h"
#include <stddef.h>
#include <memory>
#include <utility>
#include "base/bind.h"
#include "base/command_line.h"
#include "base/strings/string_number_conversions.h"
#include "build/build_config.h"
#include "content/browser/renderer_host/render_process_host_impl.h"
#include "content/browser/web_contents/web_contents_view.h"
#include "content/browser/webrtc/webrtc_internals_connections_observer.h"
#include "content/browser/webrtc/webrtc_internals_ui_observer.h"
#include "content/public/browser/audio_service.h"
#include "content/public/browser/browser_task_traits.h"
#include "content/public/browser/browser_thread.h"
#include "content/public/browser/content_browser_client.h"
#include "content/public/browser/device_service.h"
#include "content/public/browser/web_contents.h"
#include "content/public/browser/webrtc_event_logger.h"
#include "content/public/common/content_client.h"
#include "content/public/common/content_switches.h"
#include "ipc/ipc_platform_file.h"
#include "media/audio/audio_debug_recording_session.h"
#include "media/audio/audio_manager.h"
#include "media/media_buildflags.h"
#include "mojo/public/cpp/bindings/pending_receiver.h"
#include "services/audio/public/cpp/debug_recording_session_factory.h"
#include "services/device/public/mojom/wake_lock_provider.mojom.h"
#include "ui/shell_dialogs/select_file_policy.h"
using base::ProcessId;
using std::string;
namespace content {
namespace {
const base::FilePath::CharType kEventLogFilename[] =
FILE_PATH_LITERAL("event_log");
// This is intended to limit DoS attacks against the browser process consisting
// of many getUserMedia() calls. See https://crbug.com/804440.
const size_t kMaxGetUserMediaEntries = 1000;
// Makes sure that |dict| has a ListValue under path "log".
base::Value* EnsureLogList(base::Value* dict) {
base::Value* log = dict->FindListKey("log");
if (!log) {
log = dict->SetKey("log", base::Value(base::Value::Type::LIST));
}
return log;
}
// Removes the log entry associated with a given record.
void FreeLogList(base::Value* value) {
DCHECK(value->is_dict());
value->RemoveKey("log");
}
} // namespace
WebRTCInternals* WebRTCInternals::g_webrtc_internals = nullptr;
WebRTCInternals::PendingUpdate::PendingUpdate(const std::string& event_name,
base::Value event_data)
: event_name_(event_name), event_data_(std::move(event_data)) {}
WebRTCInternals::PendingUpdate::PendingUpdate(PendingUpdate&& other)
: event_name_(other.event_name_),
event_data_(std::move(other.event_data_)) {}
WebRTCInternals::PendingUpdate::~PendingUpdate() {
DCHECK(thread_checker_.CalledOnValidThread());
}
const std::string& WebRTCInternals::PendingUpdate::event_name() const {
DCHECK(thread_checker_.CalledOnValidThread());
return event_name_;
}
const base::Value* WebRTCInternals::PendingUpdate::event_data() const {
DCHECK(thread_checker_.CalledOnValidThread());
return event_data_.is_none() ? nullptr : &event_data_;
}
WebRTCInternals::WebRTCInternals() : WebRTCInternals(500, true) {}
WebRTCInternals::WebRTCInternals(int aggregate_updates_ms,
bool should_block_power_saving)
: selection_type_(SelectionType::kAudioDebugRecordings),
command_line_derived_logging_path_(
base::CommandLine::ForCurrentProcess()->GetSwitchValuePath(
switches::kWebRtcLocalEventLogging)),
event_log_recordings_(false),
num_connected_connections_(0),
should_block_power_saving_(should_block_power_saving),
aggregate_updates_ms_(aggregate_updates_ms) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
DCHECK(!g_webrtc_internals);
audio_debug_recordings_file_path_ =
GetContentClient()->browser()->GetDefaultDownloadDirectory();
event_log_recordings_file_path_ = audio_debug_recordings_file_path_;
if (audio_debug_recordings_file_path_.empty()) {
// In this case the default path (|audio_debug_recordings_file_path_|) will
// be empty and the platform default path will be used in the file dialog
// (with no default file name). See SelectFileDialog::SelectFile. On Android
// where there's no dialog we'll fail to open the file.
VLOG(1) << "Could not get the download directory.";
} else {
audio_debug_recordings_file_path_ =
audio_debug_recordings_file_path_.Append(
FILE_PATH_LITERAL("audio_debug"));
event_log_recordings_file_path_ =
event_log_recordings_file_path_.Append(kEventLogFilename);
}
// Allow command-line based setting of (local) WebRTC event logging.
if (!command_line_derived_logging_path_.empty()) {
const base::FilePath local_logs_path =
command_line_derived_logging_path_.Append(kEventLogFilename);
WebRtcEventLogger* const logger = WebRtcEventLogger::Get();
if (logger) {
logger->EnableLocalLogging(local_logs_path);
}
// For clarity's sake, though these aren't supposed to be regarded now:
event_log_recordings_ = true;
event_log_recordings_file_path_.clear();
}
g_webrtc_internals = this;
}
WebRTCInternals::~WebRTCInternals() {
DCHECK(g_webrtc_internals);
g_webrtc_internals = nullptr;
}
WebRTCInternals* WebRTCInternals::CreateSingletonInstance() {
DCHECK(!g_webrtc_internals);
g_webrtc_internals = new WebRTCInternals;
return g_webrtc_internals;
}
WebRTCInternals* WebRTCInternals::GetInstance() {
return g_webrtc_internals;
}
void WebRTCInternals::OnPeerConnectionAdded(GlobalRenderFrameHostId frame_id,
int lid,
ProcessId pid,
const string& url,
const string& rtc_configuration,
const string& constraints) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
// TODO(tommi): Consider changing this design so that webrtc-internals has
// minimal impact if chrome://webrtc-internals isn't open.
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("lid", lid);
dict.SetIntKey("pid", static_cast<int>(pid));
dict.SetStringKey("rtcConfiguration", rtc_configuration);
dict.SetStringKey("constraints", constraints);
dict.SetStringKey("url", url);
dict.SetBoolKey("isOpen", true);
dict.SetBoolKey("connected", false);
if (!observers_.empty())
SendUpdate("add-peer-connection", dict.Clone());
peer_connection_data_.Append(std::move(dict));
if (render_process_id_set_.insert(frame_id.child_id).second) {
RenderProcessHost* host = RenderProcessHost::FromID(frame_id.child_id);
if (host)
host->AddObserver(this);
}
}
void WebRTCInternals::OnPeerConnectionRemoved(GlobalRenderFrameHostId frame_id,
int lid) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
auto it = FindRecord(frame_id, lid);
if (it != peer_connection_data_.GetList().end()) {
MaybeClosePeerConnection(*it);
peer_connection_data_.EraseListIter(it);
}
if (!observers_.empty()) {
base::Value id(base::Value::Type::DICTIONARY);
id.SetIntKey("rid", frame_id.child_id);
id.SetIntKey("lid", lid);
SendUpdate("remove-peer-connection", std::move(id));
}
}
void WebRTCInternals::OnPeerConnectionUpdated(GlobalRenderFrameHostId frame_id,
int lid,
const string& type,
const string& value) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
auto it = FindRecord(frame_id, lid);
if (it == peer_connection_data_.GetList().end())
return;
if (type == "iceconnectionstatechange") {
if (value == "connected" || value == "checking" || value == "completed") {
MaybeMarkPeerConnectionAsConnected(*it);
} else if (value == "failed" || value == "disconnected" ||
value == "closed" || value == "new") {
MaybeMarkPeerConnectionAsNotConnected(*it);
}
} else if (type == "close") {
MaybeClosePeerConnection(*it);
} else if (type == "setConfiguration") {
// Update the configuration we have for this connection.
it->SetStringKey("rtcConfiguration", value);
}
// Don't update entries if there aren't any observers.
if (observers_.empty())
return;
base::Value log_entry(base::Value::Type::DICTIONARY);
double epoch_time = base::Time::Now().ToJsTime();
string time = base::NumberToString(epoch_time);
log_entry.SetStringKey("time", time);
log_entry.SetStringKey("type", type);
log_entry.SetStringKey("value", value);
base::Value update(base::Value::Type::DICTIONARY);
update.SetIntKey("rid", frame_id.child_id);
update.SetIntKey("lid", lid);
update.MergeDictionary(&log_entry);
SendUpdate("update-peer-connection", std::move(update));
// Append the update to the end of the log.
EnsureLogList(&*it)->Append(std::move(log_entry));
}
void WebRTCInternals::OnAddStandardStats(GlobalRenderFrameHostId frame_id,
int lid,
base::Value value) {
if (observers_.empty())
return;
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("lid", lid);
dict.SetKey("reports", std::move(value));
SendUpdate("add-standard-stats", std::move(dict));
}
void WebRTCInternals::OnAddLegacyStats(GlobalRenderFrameHostId frame_id,
int lid,
base::Value value) {
if (observers_.empty())
return;
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("lid", lid);
dict.SetKey("reports", std::move(value));
SendUpdate("add-legacy-stats", std::move(dict));
}
void WebRTCInternals::OnGetUserMedia(GlobalRenderFrameHostId frame_id,
base::ProcessId pid,
int request_id,
bool audio,
bool video,
const std::string& audio_constraints,
const std::string& video_constraints) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (get_user_media_requests_.GetList().size() >= kMaxGetUserMediaEntries) {
LOG(WARNING) << "Maximum number of tracked getUserMedia() requests reached "
"in webrtc-internals.";
return;
}
RenderFrameHost* host = RenderFrameHost::FromID(frame_id);
// Frame may be gone (and does not exist in tests).
std::string origin = host ? host->GetLastCommittedOrigin().Serialize() : "";
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("pid", static_cast<int>(pid));
dict.SetIntKey("request_id", request_id);
dict.SetStringKey("origin", origin);
dict.SetDoubleKey("timestamp", base::Time::Now().ToJsTime());
if (audio)
dict.SetStringKey("audio", audio_constraints);
if (video)
dict.SetStringKey("video", video_constraints);
if (!observers_.empty())
SendUpdate("add-get-user-media", dict.Clone());
get_user_media_requests_.Append(std::move(dict));
if (render_process_id_set_.insert(frame_id.child_id).second) {
RenderProcessHost* host = RenderProcessHost::FromID(frame_id.child_id);
if (host)
host->AddObserver(this);
}
}
void WebRTCInternals::OnGetUserMediaSuccess(
GlobalRenderFrameHostId frame_id,
base::ProcessId pid,
int request_id,
const std::string& stream_id,
const std::string& audio_track_info,
const std::string& video_track_info) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (get_user_media_requests_.GetList().size() >= kMaxGetUserMediaEntries) {
LOG(WARNING) << "Maximum number of tracked getUserMedia() requests reached "
"in webrtc-internals.";
return;
}
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("pid", static_cast<int>(pid));
dict.SetIntKey("request_id", request_id);
dict.SetDoubleKey("timestamp", base::Time::Now().ToJsTime());
dict.SetStringKey("stream_id", stream_id);
if (!audio_track_info.empty())
dict.SetStringKey("audio_track_info", audio_track_info);
if (!video_track_info.empty())
dict.SetStringKey("video_track_info", video_track_info);
if (!observers_.empty())
SendUpdate("update-get-user-media", dict.Clone());
get_user_media_requests_.Append(std::move(dict));
if (render_process_id_set_.insert(frame_id.child_id).second) {
RenderProcessHost* host = RenderProcessHost::FromID(frame_id.child_id);
if (host)
host->AddObserver(this);
}
}
void WebRTCInternals::OnGetUserMediaFailure(GlobalRenderFrameHostId frame_id,
base::ProcessId pid,
int request_id,
const std::string& error,
const std::string& error_message) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (get_user_media_requests_.GetList().size() >= kMaxGetUserMediaEntries) {
LOG(WARNING) << "Maximum number of tracked getUserMedia() requests reached "
"in webrtc-internals.";
return;
}
base::Value dict(base::Value::Type::DICTIONARY);
dict.SetIntKey("rid", frame_id.child_id);
dict.SetIntKey("pid", static_cast<int>(pid));
dict.SetIntKey("request_id", request_id);
dict.SetDoubleKey("timestamp", base::Time::Now().ToJsTime());
dict.SetStringKey("error", error);
dict.SetStringKey("error_message", error_message);
if (!observers_.empty())
SendUpdate("update-get-user-media", dict.Clone());
get_user_media_requests_.Append(std::move(dict));
if (render_process_id_set_.insert(frame_id.child_id).second) {
RenderProcessHost* host = RenderProcessHost::FromID(frame_id.child_id);
if (host)
host->AddObserver(this);
}
}
void WebRTCInternals::AddObserver(WebRTCInternalsUIObserver* observer) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
observers_.AddObserver(observer);
}
void WebRTCInternals::RemoveObserver(WebRTCInternalsUIObserver* observer) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
observers_.RemoveObserver(observer);
if (!observers_.empty())
return;
// Disables event log and audio debug recordings if enabled and the last
// webrtc-internals page is going away.
DisableAudioDebugRecordings();
DisableLocalEventLogRecordings();
// TODO(tommi): Consider removing all the peer_connection_data_.
for (auto& dictionary : peer_connection_data_.GetList())
FreeLogList(&dictionary);
}
void WebRTCInternals::AddConnectionsObserver(
WebRtcInternalsConnectionsObserver* observer) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
connections_observers_.AddObserver(observer);
}
void WebRTCInternals::RemoveConnectionsObserver(
WebRtcInternalsConnectionsObserver* observer) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
connections_observers_.RemoveObserver(observer);
}
void WebRTCInternals::UpdateObserver(WebRTCInternalsUIObserver* observer) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (peer_connection_data_.GetList().size() > 0)
observer->OnUpdate("update-all-peer-connections", &peer_connection_data_);
for (const auto& request : get_user_media_requests_.GetList()) {
// If there is a stream_id key or an error key this is an update.
if (request.FindStringKey("stream_id") || request.FindStringKey("error")) {
observer->OnUpdate("update-get-user-media", &request);
} else {
observer->OnUpdate("add-get-user-media", &request);
}
}
}
void WebRTCInternals::EnableAudioDebugRecordings(
content::WebContents* web_contents) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
#if defined(OS_ANDROID)
EnableAudioDebugRecordingsOnAllRenderProcessHosts();
#else
selection_type_ = SelectionType::kAudioDebugRecordings;
DCHECK(!select_file_dialog_);
select_file_dialog_ = ui::SelectFileDialog::Create(
this,
GetContentClient()->browser()->CreateSelectFilePolicy(web_contents));
select_file_dialog_->SelectFile(
ui::SelectFileDialog::SELECT_SAVEAS_FILE, std::u16string(),
audio_debug_recordings_file_path_, nullptr, 0,
base::FilePath::StringType(), web_contents->GetTopLevelNativeWindow(),
nullptr);
#endif
}
void WebRTCInternals::DisableAudioDebugRecordings() {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (!audio_debug_recording_session_)
return;
audio_debug_recording_session_.reset();
// Tear down the dialog since the user has unchecked the audio debug
// recordings box.
select_file_dialog_ = nullptr;
for (RenderProcessHost::iterator i(
content::RenderProcessHost::AllHostsIterator());
!i.IsAtEnd(); i.Advance()) {
i.GetCurrentValue()->DisableAudioDebugRecordings();
}
}
bool WebRTCInternals::IsAudioDebugRecordingsEnabled() const {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
return !!audio_debug_recording_session_;
}
const base::FilePath& WebRTCInternals::GetAudioDebugRecordingsFilePath() const {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
return audio_debug_recordings_file_path_;
}
void WebRTCInternals::EnableLocalEventLogRecordings(
content::WebContents* web_contents) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
DCHECK(CanToggleEventLogRecordings());
#if defined(OS_ANDROID)
WebRtcEventLogger* const logger = WebRtcEventLogger::Get();
if (logger) {
logger->EnableLocalLogging(event_log_recordings_file_path_);
}
#else
DCHECK(web_contents);
DCHECK(!select_file_dialog_);
selection_type_ = SelectionType::kRtcEventLogs;
select_file_dialog_ = ui::SelectFileDialog::Create(
this,
GetContentClient()->browser()->CreateSelectFilePolicy(web_contents));
select_file_dialog_->SelectFile(
ui::SelectFileDialog::SELECT_SAVEAS_FILE, std::u16string(),
event_log_recordings_file_path_, nullptr, 0, FILE_PATH_LITERAL(""),
web_contents->GetTopLevelNativeWindow(), nullptr);
#endif
}
void WebRTCInternals::DisableLocalEventLogRecordings() {
event_log_recordings_ = false;
// Tear down the dialog since the user has unchecked the event log checkbox.
select_file_dialog_ = nullptr;
DCHECK(CanToggleEventLogRecordings());
WebRtcEventLogger* const logger = WebRtcEventLogger::Get();
if (logger) {
logger->DisableLocalLogging();
}
}
bool WebRTCInternals::IsEventLogRecordingsEnabled() const {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
return event_log_recordings_;
}
bool WebRTCInternals::CanToggleEventLogRecordings() const {
return command_line_derived_logging_path_.empty();
}
void WebRTCInternals::SendUpdate(const std::string& event_name,
base::Value event_data) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
DCHECK(!observers_.empty());
bool queue_was_empty = pending_updates_.empty();
pending_updates_.push(PendingUpdate(event_name, std::move(event_data)));
if (queue_was_empty) {
GetUIThreadTaskRunner({})->PostDelayedTask(
FROM_HERE,
base::BindOnce(&WebRTCInternals::ProcessPendingUpdates,
weak_factory_.GetWeakPtr()),
base::Milliseconds(aggregate_updates_ms_));
}
}
void WebRTCInternals::RenderProcessExited(
RenderProcessHost* host,
const ChildProcessTerminationInfo& info) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
OnRendererExit(host->GetID());
render_process_id_set_.erase(host->GetID());
host->RemoveObserver(this);
}
void WebRTCInternals::FileSelected(const base::FilePath& path,
int /* unused_index */,
void* /*unused_params */) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
switch (selection_type_) {
case SelectionType::kRtcEventLogs: {
event_log_recordings_file_path_ = path;
event_log_recordings_ = true;
WebRtcEventLogger* const logger = WebRtcEventLogger::Get();
if (logger) {
logger->EnableLocalLogging(path);
}
break;
}
case SelectionType::kAudioDebugRecordings: {
audio_debug_recordings_file_path_ = path;
EnableAudioDebugRecordingsOnAllRenderProcessHosts();
break;
}
default: { NOTREACHED(); }
}
}
void WebRTCInternals::FileSelectionCanceled(void* params) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
switch (selection_type_) {
case SelectionType::kRtcEventLogs:
SendUpdate("event-log-recordings-file-selection-cancelled",
base::Value());
break;
case SelectionType::kAudioDebugRecordings:
SendUpdate("audio-debug-recordings-file-selection-cancelled",
base::Value());
break;
default:
NOTREACHED();
}
select_file_dialog_ = nullptr;
}
void WebRTCInternals::OnRendererExit(int render_process_id) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
// Iterates from the end of the list to remove the PeerConnections created
// by the exiting renderer.
base::Value::ListView peer_conn_view = peer_connection_data_.GetList();
for (int i = peer_conn_view.size() - 1; i >= 0; --i) {
DCHECK(peer_conn_view[i].is_dict());
absl::optional<int> this_rid, this_lid;
this_rid = peer_conn_view[i].FindIntKey("rid");
this_lid = peer_conn_view[i].FindIntKey("lid");
if (this_rid.value_or(0) == render_process_id) {
if (!observers_.empty()) {
base::Value update(base::Value::Type::DICTIONARY);
update.SetIntKey("rid", this_rid.value_or(0));
update.SetIntKey("lid", this_lid.value_or(0));
SendUpdate("remove-peer-connection", std::move(update));
}
MaybeClosePeerConnection(peer_conn_view[i]);
peer_connection_data_.EraseListIter(peer_conn_view.begin() + i);
}
}
UpdateWakeLock();
bool found_any = false;
// Iterates from the end of the list to remove the getUserMedia requests
// created by the exiting renderer.
base::Value::ListView get_user_media_requests_view =
get_user_media_requests_.GetList();
for (int i = get_user_media_requests_view.size() - 1; i >= 0; --i) {
DCHECK(get_user_media_requests_view[i].is_dict());
absl::optional<int> this_rid =
get_user_media_requests_view[i].FindIntKey("rid");
if (this_rid.value_or(0) == render_process_id) {
get_user_media_requests_.EraseListIter(
get_user_media_requests_view.begin() + i);
found_any = true;
}
}
if (found_any && !observers_.empty()) {
base::Value update(base::Value::Type::DICTIONARY);
update.SetIntKey("rid", render_process_id);
SendUpdate("remove-get-user-media-for-renderer", std::move(update));
}
}
void WebRTCInternals::EnableAudioDebugRecordingsOnAllRenderProcessHosts() {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
DCHECK(!audio_debug_recording_session_);
mojo::PendingRemote<audio::mojom::DebugRecording> debug_recording;
content::GetAudioService().BindDebugRecording(
debug_recording.InitWithNewPipeAndPassReceiver());
audio_debug_recording_session_ = audio::CreateAudioDebugRecordingSession(
audio_debug_recordings_file_path_, std::move(debug_recording));
for (RenderProcessHost::iterator i(
content::RenderProcessHost::AllHostsIterator());
!i.IsAtEnd(); i.Advance()) {
i.GetCurrentValue()->EnableAudioDebugRecordings(
audio_debug_recordings_file_path_);
}
}
void WebRTCInternals::MaybeClosePeerConnection(base::Value& record) {
absl::optional<bool> is_open = record.FindBoolKey("isOpen");
DCHECK(is_open.has_value());
if (!*is_open)
return;
record.SetBoolKey("isOpen", false);
MaybeMarkPeerConnectionAsNotConnected(record);
}
void WebRTCInternals::MaybeMarkPeerConnectionAsConnected(base::Value& record) {
bool was_connected = record.FindBoolKey("connected").value_or(true);
if (!was_connected) {
++num_connected_connections_;
record.SetBoolKey("connected", true);
UpdateWakeLock();
for (auto& observer : connections_observers_)
observer.OnConnectionsCountChange(num_connected_connections_);
}
}
void WebRTCInternals::MaybeMarkPeerConnectionAsNotConnected(
base::Value& record) {
bool was_connected = record.FindBoolKey("connected").value_or(false);
if (was_connected) {
record.SetBoolKey("connected", false);
--num_connected_connections_;
DCHECK_GE(num_connected_connections_, 0);
UpdateWakeLock();
for (auto& observer : connections_observers_)
observer.OnConnectionsCountChange(num_connected_connections_);
}
}
void WebRTCInternals::UpdateWakeLock() {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
if (!should_block_power_saving_)
return;
if (num_connected_connections_ == 0) {
DVLOG(1)
<< ("Cancel the wake lock on application suspension since no "
"PeerConnections are active anymore.");
GetWakeLock()->CancelWakeLock();
} else {
DCHECK_GT(num_connected_connections_, 0);
DVLOG(1) << ("Preventing the application from being suspended while one or "
"more PeerConnections are active.");
GetWakeLock()->RequestWakeLock();
}
}
device::mojom::WakeLock* WebRTCInternals::GetWakeLock() {
// Here is a lazy binding, and will not reconnect after connection error.
if (!wake_lock_) {
mojo::Remote<device::mojom::WakeLockProvider> wake_lock_provider;
GetDeviceService().BindWakeLockProvider(
wake_lock_provider.BindNewPipeAndPassReceiver());
wake_lock_provider->GetWakeLockWithoutContext(
device::mojom::WakeLockType::kPreventAppSuspension,
device::mojom::WakeLockReason::kOther,
"WebRTC has active PeerConnections",
wake_lock_.BindNewPipeAndPassReceiver());
}
return wake_lock_.get();
}
void WebRTCInternals::ProcessPendingUpdates() {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
while (!pending_updates_.empty()) {
const auto& update = pending_updates_.front();
for (auto& observer : observers_)
observer.OnUpdate(update.event_name(), update.event_data());
pending_updates_.pop();
}
}
base::CheckedContiguousIterator<base::Value> WebRTCInternals::FindRecord(
GlobalRenderFrameHostId frame_id,
int lid) {
DCHECK_CURRENTLY_ON(BrowserThread::UI);
base::Value::ListView peer_conn_view = peer_connection_data_.GetList();
for (auto it = peer_conn_view.begin(); it != peer_conn_view.end(); ++it) {
DCHECK(it->is_dict());
int this_rid = it->FindIntKey("rid").value_or(0);
int this_lid = it->FindIntKey("lid").value_or(0);
if (this_rid == frame_id.child_id && this_lid == lid)
return it;
}
return peer_conn_view.end();
}
} // namespace content