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/*
* Copyright (C) 2019-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "GStreamerDataChannelHandler.h"
#if USE(GSTREAMER_WEBRTC)
#include "EventNames.h"
#include "GStreamerWebRTCUtils.h"
#include "RTCDataChannel.h"
#include "RTCDataChannelEvent.h"
#include "RTCError.h"
#include "RTCPriorityType.h"
#include <wtf/MainThread.h>
#include <wtf/TZoneMallocInlines.h>
GST_DEBUG_CATEGORY(webkit_webrtc_data_channel_debug);
#define GST_CAT_DEFAULT webkit_webrtc_data_channel_debug
#if GST_CHECK_VERSION(1, 22, 0)
#define DC_DEBUG(...) GST_DEBUG_ID(m_channelId.ascii().data(), __VA_ARGS__)
#define DC_TRACE(...) GST_TRACE_ID(m_channelId.ascii().data(), __VA_ARGS__)
#define DC_WARNING(...) GST_WARNING_ID(m_channelId.ascii().data(), __VA_ARGS__)
#define DC_MEMDUMP(...) GST_MEMDUMP_ID(m_channelId.ascii().data(), __VA_ARGS__)
#else
#define DC_DEBUG(...) GST_DEBUG(__VA_ARGS__)
#define DC_TRACE(...) GST_TRACE(__VA_ARGS__)
#define DC_WARNING(...) GST_WARNING(__VA_ARGS__)
#define DC_MEMDUMP(...) GST_MEMDUMP(__VA_ARGS__)
#endif
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(GStreamerDataChannelHandler);
GUniquePtr<GstStructure> GStreamerDataChannelHandler::fromRTCDataChannelInit(const RTCDataChannelInit& options)
{
GUniquePtr<GstStructure> init(gst_structure_new("options", "protocol", G_TYPE_STRING, options.protocol.utf8().data(), nullptr));
if (options.ordered)
gst_structure_set(init.get(), "ordered", G_TYPE_BOOLEAN, *options.ordered, nullptr);
if (options.maxPacketLifeTime)
gst_structure_set(init.get(), "max-packet-lifetime", G_TYPE_INT, *options.maxPacketLifeTime, nullptr);
if (options.maxRetransmits)
gst_structure_set(init.get(), "max-retransmits", G_TYPE_INT, *options.maxRetransmits, nullptr);
if (options.negotiated)
gst_structure_set(init.get(), "negotiated", G_TYPE_BOOLEAN, *options.negotiated, nullptr);
if (options.id)
gst_structure_set(init.get(), "id", G_TYPE_INT, *options.id, nullptr);
GstWebRTCPriorityType priorityType;
switch (options.priority) {
case RTCPriorityType::VeryLow:
priorityType = GST_WEBRTC_PRIORITY_TYPE_VERY_LOW;
break;
case RTCPriorityType::Low:
priorityType = GST_WEBRTC_PRIORITY_TYPE_LOW;
break;
case RTCPriorityType::Medium:
priorityType = GST_WEBRTC_PRIORITY_TYPE_MEDIUM;
break;
case RTCPriorityType::High:
priorityType = GST_WEBRTC_PRIORITY_TYPE_HIGH;
break;
}
gst_structure_set(init.get(), "priority", GST_TYPE_WEBRTC_PRIORITY_TYPE, priorityType, nullptr);
return init;
}
GStreamerDataChannelHandler::GStreamerDataChannelHandler(GRefPtr<GstWebRTCDataChannel>&& channel)
: m_channel(WTFMove(channel))
{
static Atomic<uint64_t> nChannel = 0;
m_channelId = makeString("webkit-webrtc-data-channel-"_s, nChannel.exchangeAdd(1));
ASSERT(m_channel);
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_data_channel_debug, "webkitwebrtcdatachannel", 0, "WebKit WebRTC data-channel");
});
DC_DEBUG("New GStreamerDataChannelHandler for channel %p", m_channel.get());
{
Locker locker { m_clientLock };
checkState();
}
g_signal_connect_swapped(m_channel.get(), "notify::ready-state", G_CALLBACK(+[](GStreamerDataChannelHandler* handler) {
handler->readyStateChanged();
}), this);
g_signal_connect_swapped(m_channel.get(), "notify::buffered-amount", G_CALLBACK(+[](GStreamerDataChannelHandler* handler) {
handler->bufferedAmountChanged();
}), this);
g_signal_connect_swapped(m_channel.get(), "on-message-data", G_CALLBACK(+[](GStreamerDataChannelHandler* handler, GBytes* bytes) {
handler->onMessageData(bytes);
}), this);
g_signal_connect_swapped(m_channel.get(), "on-message-string", G_CALLBACK(+[](GStreamerDataChannelHandler* handler, const char* message) {
handler->onMessageString(message);
}), this);
g_signal_connect_swapped(m_channel.get(), "on-error", G_CALLBACK(+[](GStreamerDataChannelHandler* handler, GError* error) {
handler->onError(error);
}), this);
g_signal_connect_swapped(m_channel.get(), "on-close", G_CALLBACK(+[](GStreamerDataChannelHandler* handler) {
handler->onClose();
}), this);
}
GStreamerDataChannelHandler::~GStreamerDataChannelHandler()
{
DC_DEBUG("Deleting GStreamerDataChannelHandler for channel %p", m_channel.get());
if (m_channel)
g_signal_handlers_disconnect_by_data(m_channel.get(), this);
}
RTCDataChannelInit GStreamerDataChannelHandler::dataChannelInit() const
{
GUniqueOutPtr<char> protocol;
gboolean ordered, negotiated;
gint maxPacketLifeTime, maxRetransmits, id;
g_object_get(m_channel.get(), "ordered", &ordered, "max-packet-lifetime", &maxPacketLifeTime, "max-retransmits", &maxRetransmits,
"protocol", &protocol.outPtr(), "negotiated", &negotiated, "id", &id, nullptr);
RTCDataChannelInit init;
init.ordered = ordered;
init.maxPacketLifeTime = maxPacketLifeTime;
init.maxRetransmits = maxRetransmits;
init.protocol = String::fromLatin1(protocol.get());
init.negotiated = negotiated;
init.id = id;
return init;
}
String GStreamerDataChannelHandler::label() const
{
GUniqueOutPtr<char> label;
g_object_get(m_channel.get(), "label", &label.outPtr(), nullptr);
return String::fromUTF8(label.get());
}
void GStreamerDataChannelHandler::setClient(RTCDataChannelHandlerClient& client, std::optional<ScriptExecutionContextIdentifier> contextIdentifier)
{
Locker locker { m_clientLock };
ASSERT(!m_client);
DC_DEBUG("Setting client on channel %p", m_channel.get());
m_client = client;
m_contextIdentifier = contextIdentifier;
auto readyStateDispatched = checkState();
auto messages = WTFMove(m_pendingMessages);
for (auto& message : messages) {
switchOn(message, [&](Ref<FragmentedSharedBuffer>& data) {
DC_DEBUG("Notifying queued raw data (size: %zu)", data->size());
Ref contiguousData = data->makeContiguous();
auto span = contiguousData->span();
DC_MEMDUMP("Notifying raw data", span.data(), span.size());
client.didReceiveRawData(span);
}, [&](String& text) {
DC_DEBUG("Notifying queued string of size %d", text.sizeInBytes());
DC_TRACE("Notifying queued string %s", text.ascii().data());
client.didReceiveStringData(text);
}, [&](StateChange stateChange) {
if (stateChange.error) {
if (auto rtcError = toRTCError(*stateChange.error))
client.didDetectError(rtcError.releaseNonNull());
return;
}
if (!readyStateDispatched) {
DC_DEBUG("Dispatching state change to %d on channel %p", static_cast<int>(stateChange.state), m_channel.get());
client.didChangeReadyState(stateChange.state);
}
});
}
}
bool GStreamerDataChannelHandler::sendStringData(const CString& text)
{
DC_DEBUG("Sending string of length: %zu", text.length());
DC_TRACE("Sending string %s", text.data());
#if GST_CHECK_VERSION(1, 22, 0)
GUniqueOutPtr<GError> error;
// https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958
gst_webrtc_data_channel_send_string_full(GST_WEBRTC_DATA_CHANNEL(m_channel.get()), text.data(), &error.outPtr());
if (error)
DC_WARNING("Unable to send string, error: %s", error->message);
return !error;
#else
g_signal_emit_by_name(m_channel.get(), "send-string", text.data());
return true;
#endif
}
bool GStreamerDataChannelHandler::sendRawData(std::span<const uint8_t> data)
{
DC_DEBUG("Sending raw data of length: %zu", data.size());
DC_MEMDUMP("Sending raw data", data.data(), data.size());
auto bytes = adoptGRef(g_bytes_new(data.data(), data.size()));
#if GST_CHECK_VERSION(1, 22, 0)
GUniqueOutPtr<GError> error;
// https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958
gst_webrtc_data_channel_send_data_full(GST_WEBRTC_DATA_CHANNEL(m_channel.get()), bytes.get(), &error.outPtr());
if (error)
DC_WARNING("Unable to send raw data, error: %s", error->message);
return !error;
#else
g_signal_emit_by_name(m_channel.get(), "send-data", bytes.get());
return true;
#endif
}
void GStreamerDataChannelHandler::close()
{
DC_DEBUG("Closing channel %p", m_channel.get());
m_closing = true;
GstWebRTCDataChannelState channelState;
g_object_get(m_channel.get(), "ready-state", &channelState, nullptr);
if (channelState == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN)
g_signal_emit_by_name(m_channel.get(), "close");
}
std::optional<unsigned short> GStreamerDataChannelHandler::id() const
{
int id;
g_object_get(m_channel.get(), "id", &id, nullptr);
return id != -1 ? std::make_optional(id) : std::nullopt;
}
bool GStreamerDataChannelHandler::checkState()
{
ASSERT(m_clientLock.isHeld());
DC_DEBUG("Checking state.");
if (!m_channel) {
DC_DEBUG("No channel.");
return false;
}
GstWebRTCDataChannelState channelState;
g_object_get(m_channel.get(), "ready-state", &channelState, nullptr);
if (!channelState) {
DC_DEBUG("Data-channel ready-state hasn't been set yet.");
return false;
}
RTCDataChannelState state;
switch (channelState) {
#if !GST_CHECK_VERSION(1, 22, 0)
// Removed in https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2099. In
// GStreamer < 1.22 GST_WEBRTC_DATA_CHANNEL_STATE_NEW had the 0 value. We keep this case only to
// avoid adding a default case.
case GST_WEBRTC_DATA_CHANNEL_STATE_NEW:
break;
#endif
case GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING:
state = RTCDataChannelState::Connecting;
break;
case GST_WEBRTC_DATA_CHANNEL_STATE_OPEN:
state = RTCDataChannelState::Open;
break;
case GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING:
state = RTCDataChannelState::Closing;
break;
case GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED:
state = RTCDataChannelState::Closed;
break;
}
if (!m_client) {
DC_DEBUG("No client yet on channel %p, queueing state", m_channel.get());
m_pendingMessages.append(StateChange { state, { } });
return false;
}
if (channelState == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN && m_closing) {
DC_DEBUG("Ignoring open state notification on channel %p because it was pending to be closed", m_channel.get());
return false;
}
if (!*m_client) {
DC_DEBUG("Client is empty.");
return false;
}
#ifndef GST_DISABLE_GST_DEBUG
GUniquePtr<char> stateString(g_enum_to_string(GST_TYPE_WEBRTC_DATA_CHANNEL_STATE, channelState));
DC_DEBUG("Dispatching state change to %s on channel %p", stateString.get(), m_channel.get());
#endif
postTask([client = m_client, state] {
if (!*client) {
GST_DEBUG("No client");
return;
}
client.value()->didChangeReadyState(state);
});
return true;
}
void GStreamerDataChannelHandler::readyStateChanged()
{
Locker locker { m_clientLock };
checkState();
}
void GStreamerDataChannelHandler::bufferedAmountChanged()
{
Locker locker { m_clientLock };
uint64_t currentBufferedAmount;
g_object_get(m_channel.get(), "buffered-amount", ¤tBufferedAmount, nullptr);
auto bufferedAmount = static_cast<size_t>(currentBufferedAmount);
DC_DEBUG("New buffered amount on channel %p: %" G_GSIZE_FORMAT " old: %" G_GSIZE_FORMAT, m_channel.get(), bufferedAmount, m_cachedBufferedAmount ? *m_cachedBufferedAmount : -1);
if (m_cachedBufferedAmount && (*m_cachedBufferedAmount >= bufferedAmount)) {
DC_DEBUG("Buffered amount getting low on channel %p", m_channel.get());
if (!m_client) {
DC_DEBUG("No client yet on channel %p", m_channel.get());
return;
}
postTask([client = m_client, amount = *m_cachedBufferedAmount - bufferedAmount] {
if (!client)
return;
size_t clientAmount = client.value()->bufferedAmount();
size_t clampedAmount = amount > clientAmount ? clientAmount : amount;
client.value()->bufferedAmountIsDecreasing(clampedAmount);
});
}
m_cachedBufferedAmount = bufferedAmount;
}
void GStreamerDataChannelHandler::onMessageData(GBytes* bytes)
{
DC_DEBUG("Incoming data of size: %zu", g_bytes_get_size(bytes));
Locker locker { m_clientLock };
auto buffer = SharedBuffer::create(bytes);
if (!m_client) {
m_pendingMessages.append(WTFMove(buffer));
return;
}
if (!*m_client)
return;
postTask([this, client = m_client, buffer = WTFMove(buffer)] {
UNUSED_VARIABLE(this); // Conditionally used in DC_MEMDUMP.
if (!*client)
return;
auto span = buffer->span();
DC_MEMDUMP("Incoming raw data", span.data(), span.size());
client.value()->didReceiveRawData(span);
});
}
void GStreamerDataChannelHandler::onMessageString(const char* message)
{
Locker locker { m_clientLock };
DC_TRACE("Incoming string: %s", message);
if (!m_client) {
DC_DEBUG("No client yet, keeping as buffered message");
m_pendingMessages.append(String::fromUTF8(message));
return;
}
if (!*m_client)
return;
DC_DEBUG("Dispatching string of size %zu", strlen(message));
postTask([client = m_client, string = String::fromUTF8(message)] {
if (!*client)
return;
client.value()->didReceiveStringData(string);
});
}
void GStreamerDataChannelHandler::onError(GError* error)
{
Locker locker { m_clientLock };
if (!m_client)
return;
DC_WARNING("Got data-channel error %s", error->message);
GUniquePtr<GError> errorCopy(g_error_copy(error));
postTask([client = m_client, error = WTFMove(errorCopy)] {
if (!client || !error)
return;
auto rtcError = toRTCError(error.get());
if (!rtcError)
rtcError = RTCError::create(RTCError::Init { RTCErrorDetailType::DataChannelFailure, { }, { }, { }, { } }, { });
client.value()->didDetectError(rtcError.releaseNonNull());
});
}
void GStreamerDataChannelHandler::onClose()
{
Locker locker { m_clientLock };
DC_DEBUG("Channel %p closed!", m_channel.get());
checkState();
}
void GStreamerDataChannelHandler::postTask(Function<void()>&& function)
{
ASSERT(m_clientLock.isHeld());
if (!m_contextIdentifier) {
callOnMainThread(WTFMove(function));
return;
}
ScriptExecutionContext::postTaskTo(*m_contextIdentifier, WTFMove(function));
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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