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// Copyright 2022 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/peerconnection/rtp_contributing_source_cache.h"
#include "base/check.h"
#include "base/task/single_thread_task_runner.h"
#include "third_party/blink/renderer/core/execution_context/agent.h"
#include "third_party/blink/renderer/core/execution_context/execution_context.h"
#include "third_party/blink/renderer/core/loader/document_loader.h"
#include "third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.h"
#include "third_party/blink/renderer/platform/scheduler/public/event_loop.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h"
#include "third_party/blink/renderer/platform/wtf/functional.h"
namespace blink {
namespace {
HeapVector<Member<RTCRtpSynchronizationSource>>
RTCRtpSynchronizationSourcesFromRTCRtpSources(
ScriptState* script_state,
const RtpContributingSourceCache::RTCRtpSources* rtp_sources) {
LocalDOMWindow* window = LocalDOMWindow::From(script_state);
DocumentLoadTiming& time_converter =
window->GetFrame()->Loader().GetDocumentLoader()->GetTiming();
HeapVector<Member<RTCRtpSynchronizationSource>> synchronization_sources;
if (!rtp_sources)
return synchronization_sources;
for (const auto& rtp_source : *rtp_sources) {
if (rtp_source->SourceType() != RTCRtpSource::Type::kSSRC)
continue;
RTCRtpSynchronizationSource* synchronization_source =
MakeGarbageCollected<RTCRtpSynchronizationSource>();
synchronization_source->setTimestamp(
time_converter.MonotonicTimeToPseudoWallTime(rtp_source->Timestamp())
.InMilliseconds());
synchronization_source->setSource(rtp_source->Source());
if (rtp_source->AudioLevel().has_value()) {
synchronization_source->setAudioLevel(rtp_source->AudioLevel().value());
}
if (rtp_source->CaptureTimestamp().has_value()) {
synchronization_source->setCaptureTimestamp(
rtp_source->CaptureTimestamp().value());
}
if (rtp_source->SenderCaptureTimeOffset().has_value()) {
synchronization_source->setSenderCaptureTimeOffset(
rtp_source->SenderCaptureTimeOffset().value());
}
synchronization_source->setRtpTimestamp(rtp_source->RtpTimestamp());
synchronization_sources.push_back(synchronization_source);
}
return synchronization_sources;
}
HeapVector<Member<RTCRtpContributingSource>>
RTCRtpContributingSourcesFromRTCRtpSources(
ScriptState* script_state,
const RtpContributingSourceCache::RTCRtpSources* rtp_sources) {
LocalDOMWindow* window = LocalDOMWindow::From(script_state);
DocumentLoadTiming& time_converter =
window->GetFrame()->Loader().GetDocumentLoader()->GetTiming();
HeapVector<Member<RTCRtpContributingSource>> contributing_sources;
if (!rtp_sources)
return contributing_sources;
for (const auto& rtp_source : *rtp_sources) {
if (rtp_source->SourceType() != RTCRtpSource::Type::kCSRC)
continue;
RTCRtpContributingSource* contributing_source =
MakeGarbageCollected<RTCRtpContributingSource>();
contributing_source->setTimestamp(
time_converter.MonotonicTimeToPseudoWallTime(rtp_source->Timestamp())
.InMilliseconds());
contributing_source->setSource(rtp_source->Source());
if (rtp_source->AudioLevel().has_value()) {
contributing_source->setAudioLevel(rtp_source->AudioLevel().value());
}
if (rtp_source->CaptureTimestamp().has_value()) {
contributing_source->setCaptureTimestamp(
rtp_source->CaptureTimestamp().value());
}
if (rtp_source->SenderCaptureTimeOffset().has_value()) {
contributing_source->setSenderCaptureTimeOffset(
rtp_source->SenderCaptureTimeOffset().value());
}
contributing_source->setRtpTimestamp(rtp_source->RtpTimestamp());
contributing_sources.push_back(contributing_source);
}
return contributing_sources;
}
} // namespace
RtpContributingSourceCache::RtpContributingSourceCache(
RTCPeerConnection* pc,
scoped_refptr<base::SingleThreadTaskRunner> worker_thread_runner)
: pc_(pc), worker_thread_runner_(worker_thread_runner) {
DCHECK(pc_);
DCHECK(worker_thread_runner_);
}
void RtpContributingSourceCache::Shutdown() {
weak_factory_.InvalidateWeakPtrs();
}
HeapVector<Member<RTCRtpSynchronizationSource>>
RtpContributingSourceCache::getSynchronizationSources(
ScriptState* script_state,
ExceptionState& exception_state,
RTCRtpReceiver* receiver) {
if (!script_state->ContextIsValid()) {
exception_state.ThrowDOMException(DOMExceptionCode::kInvalidStateError,
"Window is detached");
return HeapVector<Member<RTCRtpSynchronizationSource>>();
}
MaybeUpdateRtpSources(script_state, receiver);
return RTCRtpSynchronizationSourcesFromRTCRtpSources(script_state,
GetRtpSources(receiver));
}
HeapVector<Member<RTCRtpContributingSource>>
RtpContributingSourceCache::getContributingSources(
ScriptState* script_state,
ExceptionState& exception_state,
RTCRtpReceiver* receiver) {
if (!script_state->ContextIsValid()) {
exception_state.ThrowDOMException(DOMExceptionCode::kInvalidStateError,
"Window is detached");
return HeapVector<Member<RTCRtpContributingSource>>();
}
MaybeUpdateRtpSources(script_state, receiver);
return RTCRtpContributingSourcesFromRTCRtpSources(script_state,
GetRtpSources(receiver));
}
void RtpContributingSourceCache::MaybeUpdateRtpSources(
ScriptState* script_state,
RTCRtpReceiver* requesting_receiver) {
if (!pc_) {
return;
}
HashMap<RTCRtpReceiverPlatform*, RTCRtpSources>* cached_sources_by_receiver;
switch (requesting_receiver->kind()) {
case RTCRtpReceiver::MediaKind::kAudio:
cached_sources_by_receiver = &cached_sources_by_audio_receiver_;
break;
case RTCRtpReceiver::MediaKind::kVideo:
cached_sources_by_receiver = &cached_sources_by_video_receiver_;
break;
}
if (cached_sources_by_receiver->find(
requesting_receiver->platform_receiver()) !=
cached_sources_by_receiver->end()) {
// The sources are already cached for this receiver, no action needed.
return;
}
// Receivers whose cache to update.
Vector<RTCRtpReceiverPlatform*> receivers;
if (cached_sources_by_receiver->empty()) {
// If the cache is empty then we only update the cache for this one
// receiver. This avoids updating the cache for all receivers in cases where
// the app is only interested in a single receiver per kind.
receivers.push_back(requesting_receiver->platform_receiver());
} else {
// If the cache is not empty, the app is interested in multiple
// RTCRtpReceiver objects. In this case, pay the cost up-front to update the
// cache for all receivers of this kind under the assumption that the app
// will be interested in all receivers of this kind. This heuristic limits
// the number of block-invoke in common use cases, but may increase overhead
// in edge cases where a subset of receivers are polled per microtask.
for (const Member<RTCRtpReceiver>& receiver : pc_->getReceivers()) {
if (receiver->kind() != requesting_receiver->kind())
continue;
receivers.push_back(receiver->platform_receiver());
}
}
base::WaitableEvent event;
// Unretained is safe because we're waiting for the operation to complete.
PostCrossThreadTask(
*worker_thread_runner_, FROM_HERE,
WTF::CrossThreadBindOnce(
&RtpContributingSourceCache::UpdateRtpSourcesOnWorkerThread,
WTF::CrossThreadUnretained(this),
WTF::CrossThreadUnretained(&receivers),
WTF::CrossThreadUnretained(cached_sources_by_receiver),
WTF::CrossThreadUnretained(&event)));
event.Wait();
ExecutionContext::From(script_state)
->GetAgent()
->event_loop()
->EnqueueMicrotask(WTF::BindOnce(&RtpContributingSourceCache::ClearCache,
weak_factory_.GetWeakPtr()));
}
void RtpContributingSourceCache::UpdateRtpSourcesOnWorkerThread(
Vector<RTCRtpReceiverPlatform*>* receivers,
HashMap<RTCRtpReceiverPlatform*, RTCRtpSources>* cached_sources_by_receiver,
base::WaitableEvent* event) {
// Calling GetSources() while on the worker thread avoids a per-receiver
// block-invoke inside the webrtc::RtpReceiverInterface PROXY.
for (RTCRtpReceiverPlatform* receiver : *receivers) {
if (cached_sources_by_receiver->find(receiver) ==
cached_sources_by_receiver->end()) {
cached_sources_by_receiver->insert(receiver, receiver->GetSources());
}
}
event->Signal();
}
void RtpContributingSourceCache::ClearCache() {
cached_sources_by_audio_receiver_.clear();
cached_sources_by_video_receiver_.clear();
}
const RtpContributingSourceCache::RTCRtpSources*
RtpContributingSourceCache::GetRtpSources(RTCRtpReceiver* receiver) const {
const HashMap<RTCRtpReceiverPlatform*, RTCRtpSources>*
cached_sources_by_receiver =
receiver->kind() == RTCRtpReceiver::MediaKind::kAudio
? &cached_sources_by_audio_receiver_
: &cached_sources_by_video_receiver_;
auto it = cached_sources_by_receiver->find(receiver->platform_receiver());
if (it == cached_sources_by_receiver->end())
return nullptr;
return &it->value;
}
} // namespace blink
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