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RTCRtpScriptTransformer.cpp
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RTCRtpScriptTransformer.cpp
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/*
* Copyright (C) 2020 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
* OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include "RTCRtpScriptTransformer.h"
#if ENABLE(WEB_RTC)
#include "DedicatedWorkerGlobalScope.h"
#include "EventLoop.h"
#include "FrameRateMonitor.h"
#include "JSDOMPromiseDeferred.h"
#include "JSRTCEncodedAudioFrame.h"
#include "JSRTCEncodedVideoFrame.h"
#include "Logging.h"
#include "MessageWithMessagePorts.h"
#include "RTCRtpTransformableFrame.h"
#include "ReadableStream.h"
#include "ReadableStreamSource.h"
#include "SerializedScriptValue.h"
#include "WorkerThread.h"
#include "WritableStream.h"
#include "WritableStreamSink.h"
namespace WebCore {
ExceptionOr<Ref<RTCRtpScriptTransformer>> RTCRtpScriptTransformer::create(ScriptExecutionContext& context, MessageWithMessagePorts&& options)
{
if (!context.globalObject())
return Exception { ExceptionCode::InvalidStateError };
auto& globalObject = *JSC::jsCast<JSDOMGlobalObject*>(context.globalObject());
JSC::JSLockHolder lock(globalObject.vm());
auto readableSource = SimpleReadableStreamSource::create();
auto readable = ReadableStream::create(globalObject, readableSource.copyRef());
if (readable.hasException())
return readable.releaseException();
if (!options.message)
return Exception { ExceptionCode::InvalidStateError };
auto ports = MessagePort::entanglePorts(context, WTFMove(options.transferredPorts));
auto transformer = adoptRef(*new RTCRtpScriptTransformer(context, options.message.releaseNonNull(), WTFMove(ports), readable.releaseReturnValue(), WTFMove(readableSource)));
transformer->suspendIfNeeded();
return transformer;
}
RTCRtpScriptTransformer::RTCRtpScriptTransformer(ScriptExecutionContext& context, Ref<SerializedScriptValue>&& options, Vector<RefPtr<MessagePort>>&& ports, Ref<ReadableStream>&& readable, Ref<SimpleReadableStreamSource>&& readableSource)
: ActiveDOMObject(&context)
, m_options(WTFMove(options))
, m_ports(WTFMove(ports))
, m_readableSource(WTFMove(readableSource))
, m_readable(WTFMove(readable))
#if !RELEASE_LOG_DISABLED
, m_enableAdditionalLogging(context.settingsValues().webRTCMediaPipelineAdditionalLoggingEnabled)
, m_identifier(RTCRtpScriptTransformerIdentifier::generate())
#endif
{
}
RTCRtpScriptTransformer::~RTCRtpScriptTransformer() = default;
ReadableStream& RTCRtpScriptTransformer::readable()
{
return m_readable.get();
}
ExceptionOr<Ref<WritableStream>> RTCRtpScriptTransformer::writable()
{
if (!m_writable) {
RefPtr context = downcast<WorkerGlobalScope>(scriptExecutionContext());
if (!context || !context->globalObject())
return Exception { ExceptionCode::InvalidStateError };
auto& globalObject = *JSC::jsCast<JSDOMGlobalObject*>(context->globalObject());
auto writableOrException = WritableStream::create(globalObject, SimpleWritableStreamSink::create([transformer = Ref { *this }](auto& context, auto value) -> ExceptionOr<void> {
if (!transformer->m_backend)
return Exception { ExceptionCode::InvalidStateError };
auto& globalObject = *context.globalObject();
auto scope = DECLARE_THROW_SCOPE(globalObject.vm());
auto frame = convert<IDLUnion<IDLInterface<RTCEncodedAudioFrame>, IDLInterface<RTCEncodedVideoFrame>>>(globalObject, value);
if (scope.exception())
return Exception { ExceptionCode::ExistingExceptionError };
auto rtcFrame = WTF::switchOn(frame, [&](RefPtr<RTCEncodedAudioFrame>& value) {
return value->rtcFrame();
}, [&](RefPtr<RTCEncodedVideoFrame>& value) {
return value->rtcFrame();
});
// If no data, skip the frame since there is nothing to packetize or decode.
if (rtcFrame->data().data()) {
#if !RELEASE_LOG_DISABLED
if (transformer->m_enableAdditionalLogging && transformer->m_backend->mediaType() == RTCRtpTransformBackend::MediaType::Video) {
if (!transformer->m_writableFrameRateMonitor) {
transformer->m_writableFrameRateMonitor = makeUnique<FrameRateMonitor>([identifier = transformer->m_identifier](auto info) {
RELEASE_LOG(WebRTC, "RTCRtpScriptTransformer writable %" PRIu64 ", frame at %f, previous frame was at %f, observed frame rate is %f, delay since last frame is %f ms, frame count is %lu", identifier.toUInt64(), info.frameTime.secondsSinceEpoch().value(), info.lastFrameTime.secondsSinceEpoch().value(), info.observedFrameRate, ((info.frameTime - info.lastFrameTime) * 1000).value(), info.frameCount);
});
}
transformer->m_writableFrameRateMonitor->update();
}
#endif
transformer->m_backend->processTransformedFrame(rtcFrame.get());
}
return { };
}));
if (writableOrException.hasException())
return writableOrException;
m_writable = writableOrException.releaseReturnValue();
}
return Ref { *m_writable };
}
void RTCRtpScriptTransformer::start(Ref<RTCRtpTransformBackend>&& backend)
{
m_backend = WTFMove(backend);
auto& context = downcast<WorkerGlobalScope>(*scriptExecutionContext());
m_backend->setTransformableFrameCallback([weakThis = WeakPtr { *this }, thread = Ref { context.thread() }](Ref<RTCRtpTransformableFrame>&& frame) mutable {
thread->runLoop().postTaskForMode([weakThis, frame = WTFMove(frame)](auto& context) mutable {
if (weakThis)
weakThis->enqueueFrame(context, WTFMove(frame));
}, WorkerRunLoop::defaultMode());
});
}
void RTCRtpScriptTransformer::clear(ClearCallback clearCallback)
{
if (m_backend && clearCallback == ClearCallback::Yes)
m_backend->clearTransformableFrameCallback();
m_backend = nullptr;
stopPendingActivity();
}
void RTCRtpScriptTransformer::enqueueFrame(ScriptExecutionContext& context, Ref<RTCRtpTransformableFrame>&& frame)
{
if (!m_backend)
return;
auto* globalObject = JSC::jsCast<JSDOMGlobalObject*>(context.globalObject());
if (!globalObject)
return;
auto& vm = globalObject->vm();
JSC::JSLockHolder lock(vm);
bool isVideo = m_backend->mediaType() == RTCRtpTransformBackend::MediaType::Video;
if (isVideo && !m_pendingKeyFramePromises.isEmpty() && frame->isKeyFrame()) {
for (auto& promise : std::exchange(m_pendingKeyFramePromises, { }))
promise->resolve();
}
#if !RELEASE_LOG_DISABLED
if (m_enableAdditionalLogging && isVideo) {
if (!m_readableFrameRateMonitor) {
m_readableFrameRateMonitor = makeUnique<FrameRateMonitor>([identifier = m_identifier](auto info) {
RELEASE_LOG(WebRTC, "RTCRtpScriptTransformer readable %" PRIu64 ", frame at %f, previous frame was at %f, observed frame rate is %f, delay since last frame is %f ms, frame count is %lu", identifier.toUInt64(), info.frameTime.secondsSinceEpoch().value(), info.lastFrameTime.secondsSinceEpoch().value(), info.observedFrameRate, ((info.frameTime - info.lastFrameTime) * 1000).value(), info.frameCount);
});
}
m_readableFrameRateMonitor->update();
}
#endif
auto value = isVideo ? toJS(globalObject, globalObject, RTCEncodedVideoFrame::create(WTFMove(frame))) : toJS(globalObject, globalObject, RTCEncodedAudioFrame::create(WTFMove(frame)));
m_readableSource->enqueue(value);
}
void RTCRtpScriptTransformer::generateKeyFrame(Ref<DeferredPromise>&& promise)
{
RefPtr context = scriptExecutionContext();
if (!context || !m_backend || m_backend->side() != RTCRtpTransformBackend::Side::Sender || m_backend->mediaType() != RTCRtpTransformBackend::MediaType::Video) {
promise->reject(Exception { ExceptionCode::InvalidStateError, "Not attached to a valid video sender"_s });
return;
}
bool shouldRequestKeyFrame = m_pendingKeyFramePromises.isEmpty();
m_pendingKeyFramePromises.append(WTFMove(promise));
if (shouldRequestKeyFrame)
m_backend->requestKeyFrame();
}
void RTCRtpScriptTransformer::sendKeyFrameRequest(Ref<DeferredPromise>&& promise)
{
RefPtr context = scriptExecutionContext();
if (!context || !m_backend || m_backend->side() != RTCRtpTransformBackend::Side::Receiver || m_backend->mediaType() != RTCRtpTransformBackend::MediaType::Video) {
promise->reject(Exception { ExceptionCode::InvalidStateError, "Not attached to a valid video receiver"_s });
return;
}
// FIXME: We should be able to know when the FIR request is sent to resolve the promise at this exact time.
m_backend->requestKeyFrame();
context->eventLoop().queueTask(TaskSource::Networking, [promise = WTFMove(promise)]() mutable {
promise->resolve();
});
}
JSC::JSValue RTCRtpScriptTransformer::options(JSC::JSGlobalObject& globalObject)
{
return m_options->deserialize(globalObject, &globalObject, m_ports);
}
} // namespace WebCore
#endif // ENABLE(WEB_RTC)