| /* |
| * Copyright (C) 2009, 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk> |
| * Copyright (C) 2013 Collabora Ltd. |
| * Copyright (C) 2019, 2020 Igalia S.L. |
| * |
| * 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 "WebKitWebSourceGStreamer.h" |
| |
| #if ENABLE(VIDEO) && USE(GSTREAMER) |
| |
| #include "GStreamerCommon.h" |
| #include "HTTPHeaderNames.h" |
| #include "MediaPlayer.h" |
| #include "PlatformMediaResourceLoader.h" |
| #include "PolicyChecker.h" |
| #include "ResourceError.h" |
| #include "ResourceRequest.h" |
| #include "ResourceResponse.h" |
| #include "SecurityOrigin.h" |
| #include <cstdint> |
| #include <wtf/Condition.h> |
| #include <wtf/DataMutex.h> |
| #include <wtf/PrintStream.h> |
| #include <wtf/RunLoop.h> |
| #include <wtf/Scope.h> |
| #include <wtf/glib/WTFGType.h> |
| #include <wtf/text/CString.h> |
| #include <wtf/text/StringToIntegerConversion.h> |
| |
| using namespace WebCore; |
| |
| // Never pause download of media resources smaller than 2MiB. |
| #define SMALL_MEDIA_RESOURCE_MAX_SIZE 2 * 1024 * 1024 |
| |
| // Keep at most 2% of the full, non-small, media resource buffered. When this |
| // threshold is reached, the download task is paused. |
| #define HIGH_QUEUE_FACTOR_THRESHOLD 0.02 |
| |
| // Keep at least 20% of maximum queue size buffered. When this threshold is |
| // reached, the download task resumes. |
| #define LOW_QUEUE_FACTOR_THRESHOLD 0.2 |
| |
| class CachedResourceStreamingClient final : public PlatformMediaResourceClient { |
| WTF_MAKE_FAST_ALLOCATED; |
| WTF_MAKE_NONCOPYABLE(CachedResourceStreamingClient); |
| public: |
| CachedResourceStreamingClient(WebKitWebSrc*, ResourceRequest&&, unsigned requestNumber); |
| virtual ~CachedResourceStreamingClient(); |
| |
| const HashSet<RefPtr<WebCore::SecurityOrigin>>& securityOrigins() const { return m_origins; } |
| |
| void setSourceElement(WebKitWebSrc* src) { m_src = GST_ELEMENT_CAST(src); } |
| |
| private: |
| void checkUpdateBlocksize(unsigned bytesRead); |
| |
| // PlatformMediaResourceClient virtual methods. |
| void responseReceived(PlatformMediaResource&, const ResourceResponse&, CompletionHandler<void(ShouldContinuePolicyCheck)>&&) override; |
| void dataReceived(PlatformMediaResource&, Ref<SharedBuffer>&&) override; |
| void accessControlCheckFailed(PlatformMediaResource&, const ResourceError&) override; |
| void loadFailed(PlatformMediaResource&, const ResourceError&) override; |
| void loadFinished(PlatformMediaResource&, const NetworkLoadMetrics&) override; |
| |
| static constexpr int s_growBlocksizeLimit { 1 }; |
| static constexpr int s_growBlocksizeCount { 2 }; |
| static constexpr int s_growBlocksizeFactor { 2 }; |
| static constexpr float s_reduceBlocksizeLimit { 0.5 }; |
| static constexpr int s_reduceBlocksizeCount { 2 }; |
| static constexpr float s_reduceBlocksizeFactor { 0.5 }; |
| int m_reduceBlocksizeCount { 0 }; |
| int m_increaseBlocksizeCount { 0 }; |
| unsigned m_requestNumber; |
| |
| GRefPtr<GstElement> m_src; |
| ResourceRequest m_request; |
| HashSet<RefPtr<WebCore::SecurityOrigin>> m_origins; |
| }; |
| |
| struct WebKitWebSrcPrivate { |
| // Constants initialized during construction: |
| unsigned minimumBlocksize; |
| |
| // Configuration of the element (properties set by the user of WebKitWebSrc): |
| // They can only change when state < PAUSED. |
| CString originalURI; |
| bool keepAlive; |
| GUniquePtr<GstStructure> extraHeaders; |
| bool compress; |
| GUniquePtr<gchar> httpMethod; |
| |
| struct StreamingMembers { |
| #ifndef NDEBUG |
| ~StreamingMembers() |
| { |
| // By the time we're destroying WebKitWebSrcPrivate unLock() should have been called and therefore resource |
| // should have already been cleared. |
| ASSERT(!resource); |
| } |
| #endif |
| |
| // Properties initially empty, but set once the first HTTP response arrives: |
| bool wasResponseReceived; |
| CString redirectedURI; |
| bool didPassAccessControlCheck; |
| bool haveSize; |
| uint64_t size; |
| bool isSeekable; |
| GRefPtr<GstCaps> pendingCaps; |
| GRefPtr<GstMessage> pendingHttpHeadersMessage; // Set from MT, sent from create(). |
| GRefPtr<GstEvent> pendingHttpHeadersEvent; // Set from MT, sent from create(). |
| |
| // Properties updated with every downloaded data block: |
| WallTime downloadStartTime { WallTime::nan() }; |
| uint64_t totalDownloadedBytes { 0 }; |
| bool doesHaveEOS; // Set both when we reach stopPosition and on errors (including on responseReceived). |
| bool isDownloadSuspended { false }; // Set to true from the network handler when the high water level is reached. |
| |
| // Obtained by means of GstContext queries before making the first HTTP request. |
| // We use it for getting access to WebKit networking objects: the PlatformMediaResourceLoader factory [createResourceLoader()] |
| // and the player HTTP referrer string. |
| WebCore::MediaPlayer* player; |
| |
| // MediaPlayer referrer cached value. The corresponding method has to be called from the |
| // main thread, so the value needs to be cached before use in non-main thread. |
| String referrer; |
| |
| // Properties used for GStreamer data-flow in create(). |
| bool isFlushing { false }; |
| Condition responseCondition; // Must be signaled after any updates on HTTP requests, and when flushing. |
| GRefPtr<GstAdapter> adapter; |
| bool isDurationSet; |
| uint64_t readPosition; |
| |
| // Properties only set during seek. |
| // basesrc ensures they can't change during a create() call by taking the STREAMING_LOCK. |
| // (An initial seek is also guaranteed by basesrc.) |
| unsigned requestNumber { 1 }; |
| uint64_t requestedPosition { 0 }; |
| uint64_t stopPosition { UINT64_MAX }; |
| |
| bool isRequestPending { true }; |
| |
| RefPtr<PlatformMediaResourceLoader> loader; |
| RefPtr<PlatformMediaResource> resource; |
| }; |
| DataMutex<StreamingMembers> dataMutex; |
| }; |
| |
| enum { |
| PROP_0, |
| PROP_LOCATION, |
| PROP_RESOLVED_LOCATION, |
| PROP_KEEP_ALIVE, |
| PROP_EXTRA_HEADERS, |
| PROP_COMPRESS, |
| PROP_METHOD |
| }; |
| |
| static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS_ANY); |
| |
| GST_DEBUG_CATEGORY_STATIC(webkit_web_src_debug); |
| #define GST_CAT_DEFAULT webkit_web_src_debug |
| |
| static void webKitWebSrcUriHandlerInit(gpointer gIface, gpointer ifaceData); |
| |
| static void webKitWebSrcConstructed(GObject*); |
| static void webKitWebSrcSetProperty(GObject*, guint propertyID, const GValue*, GParamSpec*); |
| static void webKitWebSrcGetProperty(GObject*, guint propertyID, GValue*, GParamSpec*); |
| static GstFlowReturn webKitWebSrcCreate(GstPushSrc*, GstBuffer**); |
| static void webKitWebSrcMakeRequest(WebKitWebSrc*, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>&); |
| static gboolean webKitWebSrcStart(GstBaseSrc*); |
| static gboolean webKitWebSrcStop(GstBaseSrc*); |
| static gboolean webKitWebSrcGetSize(GstBaseSrc*, guint64* size); |
| static gboolean webKitWebSrcIsSeekable(GstBaseSrc*); |
| static gboolean webKitWebSrcDoSeek(GstBaseSrc*, GstSegment*); |
| static gboolean webKitWebSrcQuery(GstBaseSrc*, GstQuery*); |
| static gboolean webKitWebSrcUnLock(GstBaseSrc*); |
| static gboolean webKitWebSrcUnLockStop(GstBaseSrc*); |
| static void webKitWebSrcSetContext(GstElement*, GstContext*); |
| static void restartLoaderIfNeeded(WebKitWebSrc*, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>&); |
| static void stopLoaderIfNeeded(WebKitWebSrc*, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>&); |
| |
| #define webkit_web_src_parent_class parent_class |
| WEBKIT_DEFINE_TYPE_WITH_CODE(WebKitWebSrc, webkit_web_src, GST_TYPE_PUSH_SRC, |
| G_IMPLEMENT_INTERFACE(GST_TYPE_URI_HANDLER, webKitWebSrcUriHandlerInit); |
| GST_DEBUG_CATEGORY_INIT(webkit_web_src_debug, "webkitwebsrc", 0, "websrc element"); |
| ) |
| |
| static void webkit_web_src_class_init(WebKitWebSrcClass* klass) |
| { |
| GObjectClass* oklass = G_OBJECT_CLASS(klass); |
| |
| oklass->constructed = webKitWebSrcConstructed; |
| oklass->set_property = webKitWebSrcSetProperty; |
| oklass->get_property = webKitWebSrcGetProperty; |
| |
| GstElementClass* eklass = GST_ELEMENT_CLASS(klass); |
| gst_element_class_add_static_pad_template(eklass, &srcTemplate); |
| |
| gst_element_class_set_metadata(eklass, "WebKit Web source element", "Source/Network", "Handles HTTP/HTTPS uris", |
| "Philippe Normand <philn@igalia.com>"); |
| |
| /* Allows setting the uri using the 'location' property, which is used |
| * for example by gst_element_make_from_uri() */ |
| g_object_class_install_property(oklass, PROP_LOCATION, |
| g_param_spec_string("location", "location", "Location to read from", |
| nullptr, static_cast<GParamFlags>(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); |
| |
| g_object_class_install_property(oklass, PROP_RESOLVED_LOCATION, |
| g_param_spec_string("resolved-location", "Resolved location", "The location resolved by the server", |
| nullptr, static_cast<GParamFlags>(G_PARAM_READABLE | G_PARAM_STATIC_STRINGS))); |
| |
| g_object_class_install_property(oklass, PROP_KEEP_ALIVE, |
| g_param_spec_boolean("keep-alive", "keep-alive", "Use HTTP persistent connections", |
| FALSE, static_cast<GParamFlags>(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); |
| |
| g_object_class_install_property(oklass, PROP_EXTRA_HEADERS, |
| g_param_spec_boxed("extra-headers", "Extra Headers", "Extra headers to append to the HTTP request", |
| GST_TYPE_STRUCTURE, static_cast<GParamFlags>(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); |
| |
| g_object_class_install_property(oklass, PROP_COMPRESS, |
| g_param_spec_boolean("compress", "Compress", "Allow compressed content encodings", |
| FALSE, static_cast<GParamFlags>(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); |
| |
| g_object_class_install_property(oklass, PROP_METHOD, |
| g_param_spec_string("method", "method", "The HTTP method to use (default: GET)", |
| nullptr, static_cast<GParamFlags>(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); |
| |
| eklass->set_context = GST_DEBUG_FUNCPTR(webKitWebSrcSetContext); |
| |
| GstBaseSrcClass* baseSrcClass = GST_BASE_SRC_CLASS(klass); |
| baseSrcClass->start = GST_DEBUG_FUNCPTR(webKitWebSrcStart); |
| baseSrcClass->stop = GST_DEBUG_FUNCPTR(webKitWebSrcStop); |
| baseSrcClass->unlock = GST_DEBUG_FUNCPTR(webKitWebSrcUnLock); |
| baseSrcClass->unlock_stop = GST_DEBUG_FUNCPTR(webKitWebSrcUnLockStop); |
| baseSrcClass->get_size = GST_DEBUG_FUNCPTR(webKitWebSrcGetSize); |
| baseSrcClass->is_seekable = GST_DEBUG_FUNCPTR(webKitWebSrcIsSeekable); |
| baseSrcClass->do_seek = GST_DEBUG_FUNCPTR(webKitWebSrcDoSeek); |
| baseSrcClass->query = GST_DEBUG_FUNCPTR(webKitWebSrcQuery); |
| |
| GstPushSrcClass* pushSrcClass = GST_PUSH_SRC_CLASS(klass); |
| pushSrcClass->create = GST_DEBUG_FUNCPTR(webKitWebSrcCreate); |
| } |
| |
| enum class ResetType { |
| Soft, |
| Hard |
| }; |
| |
| static void webkitWebSrcReset(WebKitWebSrc* src, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>& members, ResetType resetType) |
| { |
| GST_DEBUG_OBJECT(src, "Resetting internal state"); |
| gst_adapter_clear(members->adapter.get()); |
| members->isRequestPending = true; |
| |
| // Reset request state. Any previous request has been cancelled at this point. |
| members->wasResponseReceived = false; |
| members->doesHaveEOS = false; |
| members->downloadStartTime = WallTime::nan(); |
| members->totalDownloadedBytes = 0; // Resetted for each request, used to estimate download speed. |
| members->pendingHttpHeadersMessage = nullptr; |
| members->pendingHttpHeadersEvent = nullptr; |
| |
| // After a flush, we have to emit a segment again. |
| members->isDurationSet = false; |
| |
| // Hard reset is done during initialization and state transitions. |
| // Soft reset is done during flushes. In these, we preserve the seek target. |
| if (resetType == ResetType::Hard) { |
| members->didPassAccessControlCheck = false; |
| members->redirectedURI = CString(); |
| members->isSeekable = false; |
| members->haveSize = false; |
| members->size = 0; |
| members->requestedPosition = 0; |
| members->stopPosition = UINT64_MAX; |
| members->readPosition = members->requestedPosition; |
| } |
| } |
| |
| static void webKitWebSrcConstructed(GObject* object) |
| { |
| GST_CALL_PARENT(G_OBJECT_CLASS, constructed, (object)); |
| |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(object); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| priv->minimumBlocksize = gst_base_src_get_blocksize(GST_BASE_SRC_CAST(src)); |
| |
| DataMutexLocker members { priv->dataMutex }; |
| members->adapter = adoptGRef(gst_adapter_new()); |
| webkitWebSrcReset(src, members, ResetType::Hard); |
| |
| gst_base_src_set_automatic_eos(GST_BASE_SRC_CAST(src), FALSE); |
| } |
| |
| static void webKitWebSrcSetProperty(GObject* object, guint propID, const GValue* value, GParamSpec* pspec) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(object); |
| |
| switch (propID) { |
| case PROP_LOCATION: |
| gst_uri_handler_set_uri(reinterpret_cast<GstURIHandler*>(src), g_value_get_string(value), nullptr); |
| break; |
| case PROP_KEEP_ALIVE: |
| src->priv->keepAlive = g_value_get_boolean(value); |
| break; |
| case PROP_EXTRA_HEADERS: { |
| const GstStructure* s = gst_value_get_structure(value); |
| src->priv->extraHeaders.reset(s ? gst_structure_copy(s) : nullptr); |
| break; |
| } |
| case PROP_COMPRESS: |
| src->priv->compress = g_value_get_boolean(value); |
| break; |
| case PROP_METHOD: |
| src->priv->httpMethod.reset(g_value_dup_string(value)); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propID, pspec); |
| break; |
| } |
| } |
| |
| static void webKitWebSrcGetProperty(GObject* object, guint propID, GValue* value, GParamSpec* pspec) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(object); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| switch (propID) { |
| case PROP_LOCATION: |
| g_value_set_string(value, priv->originalURI.data()); |
| break; |
| case PROP_RESOLVED_LOCATION: { |
| DataMutexLocker members { priv->dataMutex }; |
| g_value_set_string(value, members->redirectedURI.isNull() ? priv->originalURI.data() : members->redirectedURI.data()); |
| break; |
| } |
| case PROP_KEEP_ALIVE: |
| g_value_set_boolean(value, priv->keepAlive); |
| break; |
| case PROP_EXTRA_HEADERS: |
| gst_value_set_structure(value, priv->extraHeaders.get()); |
| break; |
| case PROP_COMPRESS: |
| g_value_set_boolean(value, priv->compress); |
| break; |
| case PROP_METHOD: |
| g_value_set_string(value, priv->httpMethod.get()); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propID, pspec); |
| break; |
| } |
| } |
| |
| static void webKitWebSrcSetContext(GstElement* element, GstContext* context) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(element); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| GST_DEBUG_OBJECT(src, "context type: %s", gst_context_get_context_type(context)); |
| if (gst_context_has_context_type(context, WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME)) { |
| const GValue* value = gst_structure_get_value(gst_context_get_structure(context), "player"); |
| DataMutexLocker members { priv->dataMutex }; |
| members->player = reinterpret_cast<MediaPlayer*>(g_value_get_pointer(value)); |
| } |
| GST_ELEMENT_CLASS(parent_class)->set_context(element, context); |
| } |
| |
| static void restartLoaderIfNeeded(WebKitWebSrc* src, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>& members) |
| { |
| if (!members->isDownloadSuspended) { |
| GST_TRACE_OBJECT(src, "download already active"); |
| return; |
| } |
| |
| GST_TRACE_OBJECT(src, "is download suspended %s, does have EOS %s, does have size %s, is seekable %s, size %" G_GUINT64_FORMAT |
| " (min %u)", boolForPrinting(members->isDownloadSuspended), boolForPrinting(members->doesHaveEOS), boolForPrinting(members->haveSize) |
| , boolForPrinting(members->isSeekable), members->size, SMALL_MEDIA_RESOURCE_MAX_SIZE); |
| if (members->doesHaveEOS || !members->haveSize || !members->isSeekable || members->size <= SMALL_MEDIA_RESOURCE_MAX_SIZE) { |
| GST_TRACE_OBJECT(src, "download cannot be stopped/restarted"); |
| return; |
| } |
| GST_TRACE_OBJECT(src, "read position %" G_GUINT64_FORMAT ", state %s", members->readPosition, gst_element_state_get_name(GST_STATE(src))); |
| if (!members->readPosition || members->readPosition == members->size || GST_STATE(src) < GST_STATE_PAUSED) { |
| GST_TRACE_OBJECT(src, "can't restart download"); |
| return; |
| } |
| |
| size_t queueSize = gst_adapter_available(members->adapter.get()); |
| GST_TRACE_OBJECT(src, "queue size %zu (min %1.0f)", queueSize |
| , members->size * HIGH_QUEUE_FACTOR_THRESHOLD * LOW_QUEUE_FACTOR_THRESHOLD); |
| |
| if (queueSize >= members->size * HIGH_QUEUE_FACTOR_THRESHOLD * LOW_QUEUE_FACTOR_THRESHOLD) { |
| GST_TRACE_OBJECT(src, "queue size above low watermark, not restarting download"); |
| return; |
| } |
| |
| GST_DEBUG_OBJECT(src, "restarting download"); |
| members->isDownloadSuspended = false; |
| members->requestNumber++; |
| members->requestedPosition = members->readPosition; |
| webKitWebSrcMakeRequest(src, members); |
| } |
| |
| |
| static void stopLoaderIfNeeded(WebKitWebSrc* src, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>& members) |
| { |
| ASSERT(isMainThread()); |
| |
| if (members->isDownloadSuspended) { |
| GST_TRACE_OBJECT(src, "download already suspended"); |
| return; |
| } |
| |
| GST_TRACE_OBJECT(src, "is download suspended %s, does have size %s, is seekable %s, size %" G_GUINT64_FORMAT " (min %u)" |
| , boolForPrinting(members->isDownloadSuspended), boolForPrinting(members->haveSize), boolForPrinting(members->isSeekable), members->size |
| , SMALL_MEDIA_RESOURCE_MAX_SIZE); |
| if (!members->isSeekable || members->size <= SMALL_MEDIA_RESOURCE_MAX_SIZE) { |
| GST_TRACE_OBJECT(src, "download cannot be stopped/restarted"); |
| return; |
| } |
| |
| size_t queueSize = gst_adapter_available(members->adapter.get()); |
| GST_TRACE_OBJECT(src, "queue size %zu (max %1.0f)", queueSize, members->size * HIGH_QUEUE_FACTOR_THRESHOLD); |
| if (queueSize <= members->size * HIGH_QUEUE_FACTOR_THRESHOLD) { |
| GST_TRACE_OBJECT(src, "queue size under high watermark, not stopping download"); |
| return; |
| } |
| |
| if (members->readPosition == members->size) { |
| GST_TRACE_OBJECT(src, "just downloaded the last chunk in the file, loadFinished() is about to be called"); |
| return; |
| } |
| |
| GST_DEBUG_OBJECT(src, "R%u: stopping download", members->requestNumber); |
| members->isDownloadSuspended = true; |
| members->resource->stop(); |
| } |
| |
| static GstFlowReturn webKitWebSrcCreate(GstPushSrc* pushSrc, GstBuffer** buffer) |
| { |
| ASSERT(!isMainThread()); |
| GstBaseSrc* baseSrc = GST_BASE_SRC_CAST(pushSrc); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| WebKitWebSrcPrivate* priv = src->priv; |
| DataMutexLocker members { priv->dataMutex }; |
| |
| // We need members->player to make requests. There are two mechanisms for this. |
| // |
| // 1) webKitWebSrcSetMediaPlayer() is called by MediaPlayerPrivateGStreamer by means of hooking playbin's |
| // "source-setup" event. This doesn't work for additional WebKitWebSrc elements created by adaptivedemux. |
| // |
| // 2) A GstContext query made here. |
| if (!members->player) { |
| members.runUnlocked([src, baseSrc]() { |
| GRefPtr<GstQuery> query = adoptGRef(gst_query_new_context(WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME)); |
| if (gst_pad_peer_query(GST_BASE_SRC_PAD(baseSrc), query.get())) { |
| GstContext* context; |
| |
| gst_query_parse_context(query.get(), &context); |
| gst_element_set_context(GST_ELEMENT_CAST(src), context); |
| } else |
| gst_element_post_message(GST_ELEMENT_CAST(src), gst_message_new_need_context(GST_OBJECT_CAST(src), WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME)); |
| }); |
| if (members->isFlushing) |
| return GST_FLOW_FLUSHING; |
| } |
| if (!members->player) { |
| GST_ERROR_OBJECT(src, "Couldn't obtain WebKitWebSrcPlayerContext, which is necessary to make network requests"); |
| return GST_FLOW_ERROR; |
| } |
| |
| GST_TRACE_OBJECT(src, "readPosition = %" G_GUINT64_FORMAT " requestedPosition = %" G_GUINT64_FORMAT, members->readPosition, members->requestedPosition); |
| |
| if (members->isRequestPending) { |
| members->isRequestPending = false; |
| webKitWebSrcMakeRequest(src, members); |
| } |
| |
| // Wait for the response headers. |
| members->responseCondition.wait(members.mutex(), [&] () { |
| return members->wasResponseReceived || members->isFlushing; |
| }); |
| |
| if (members->isFlushing) |
| return GST_FLOW_FLUSHING; |
| |
| if (members->pendingCaps) { |
| GST_DEBUG_OBJECT(src, "Setting caps: %" GST_PTR_FORMAT, members->pendingCaps.get()); |
| members.runUnlocked([baseSrc, caps = members->pendingCaps.leakRef()]() { |
| gst_base_src_set_caps(baseSrc, caps); |
| }); |
| if (members->isFlushing) |
| return GST_FLOW_FLUSHING; |
| } |
| |
| if (members->haveSize && !members->isDurationSet) { |
| GST_DEBUG_OBJECT(src, "Setting duration to %" G_GUINT64_FORMAT, members->size); |
| baseSrc->segment.duration = members->size; |
| members->isDurationSet = true; |
| gst_element_post_message(GST_ELEMENT_CAST(src), gst_message_new_duration_changed(GST_OBJECT_CAST(src))); |
| } |
| |
| if (members->pendingHttpHeadersMessage) |
| gst_element_post_message(GST_ELEMENT(src), members->pendingHttpHeadersMessage.leakRef()); |
| if (members->pendingHttpHeadersEvent) |
| gst_pad_push_event(GST_BASE_SRC_PAD(baseSrc), members->pendingHttpHeadersEvent.leakRef()); |
| |
| restartLoaderIfNeeded(src, members); |
| |
| // We don't use the GstAdapter methods marked as fast anymore because sometimes it was slower. The reason why this was slower is that we can be |
| // waiting for more "fast" (that could be retrieved with the _fast API) buffers to be available even in cases where the queue is not empty. These |
| // other buffers can be retrieved with the "non _fast" API. |
| GST_TRACE_OBJECT(src, "doesHaveEOS: %s, isDownloadSuspended: %s", boolForPrinting(members->doesHaveEOS), boolForPrinting(members->isDownloadSuspended)); |
| |
| unsigned size = gst_base_src_get_blocksize(baseSrc); |
| size_t queueSize = gst_adapter_available(members->adapter.get()); |
| GST_TRACE_OBJECT(src, "available bytes %" G_GSIZE_FORMAT ", block size %u", queueSize, size); |
| if (!queueSize) { |
| GST_TRACE_OBJECT(src, "let's wait for data or EOS"); |
| members->responseCondition.wait(members.mutex(), [&] { |
| return members->isFlushing || gst_adapter_available(members->adapter.get()) || members->doesHaveEOS; |
| }); |
| if (members->isFlushing) |
| return GST_FLOW_FLUSHING; |
| |
| queueSize = gst_adapter_available(members->adapter.get()); |
| GST_TRACE_OBJECT(src, "available %" G_GSIZE_FORMAT, queueSize); |
| } |
| |
| if (queueSize) { |
| if (queueSize < size) { |
| GST_TRACE_OBJECT(src, "Did not get the %u blocksize bytes, let's push the %" G_GSIZE_FORMAT " bytes we got", size, queueSize); |
| size = queueSize; |
| } else |
| GST_TRACE_OBJECT(src, "Taking %u bytes from adapter", size); |
| |
| *buffer = gst_adapter_take_buffer(members->adapter.get(), size); |
| RELEASE_ASSERT(*buffer); |
| |
| GST_BUFFER_OFFSET(*buffer) = baseSrc->segment.position; |
| GST_BUFFER_OFFSET_END(*buffer) = GST_BUFFER_OFFSET(*buffer) + size; |
| GST_TRACE_OBJECT(src, "Buffer bounds set to %" G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT, GST_BUFFER_OFFSET(*buffer), GST_BUFFER_OFFSET_END(*buffer)); |
| GST_TRACE_OBJECT(src, "buffer size: %u, total content size: %" G_GUINT64_FORMAT, size, members->size); |
| |
| restartLoaderIfNeeded(src, members); |
| return GST_FLOW_OK; |
| } |
| |
| // If the queue is empty reached this point, the only other option is that we are in EOS. |
| ASSERT(members->doesHaveEOS); |
| GST_DEBUG_OBJECT(src, "Reached the end of the response, signalling EOS"); |
| |
| gst_element_post_message(GST_ELEMENT_CAST(src), gst_message_new_element(GST_OBJECT_CAST(src), |
| gst_structure_new_empty("webkit-web-src-has-eos"))); |
| |
| return GST_FLOW_EOS; |
| } |
| |
| static bool webKitWebSrcSetExtraHeader(GQuark fieldId, const GValue* value, gpointer userData) |
| { |
| GUniquePtr<gchar> fieldContent; |
| |
| if (G_VALUE_HOLDS_STRING(value)) |
| fieldContent.reset(g_value_dup_string(value)); |
| else { |
| GValue dest = G_VALUE_INIT; |
| |
| g_value_init(&dest, G_TYPE_STRING); |
| if (g_value_transform(value, &dest)) |
| fieldContent.reset(g_value_dup_string(&dest)); |
| } |
| |
| const gchar* fieldName = g_quark_to_string(fieldId); |
| if (!fieldContent.get()) { |
| GST_ERROR("extra-headers field '%s' contains no value or can't be converted to a string", fieldName); |
| return false; |
| } |
| |
| GST_DEBUG("Appending extra header: \"%s: %s\"", fieldName, fieldContent.get()); |
| ResourceRequest* request = static_cast<ResourceRequest*>(userData); |
| request->setHTTPHeaderField(fieldName, fieldContent.get()); |
| return true; |
| } |
| |
| static gboolean webKitWebSrcProcessExtraHeaders(GQuark fieldId, const GValue* value, gpointer userData) |
| { |
| if (G_VALUE_TYPE(value) == GST_TYPE_ARRAY) { |
| unsigned size = gst_value_array_get_size(value); |
| |
| for (unsigned i = 0; i < size; i++) { |
| if (!webKitWebSrcSetExtraHeader(fieldId, gst_value_array_get_value(value, i), userData)) |
| return FALSE; |
| } |
| return TRUE; |
| } |
| |
| if (G_VALUE_TYPE(value) == GST_TYPE_LIST) { |
| unsigned size = gst_value_list_get_size(value); |
| |
| for (unsigned i = 0; i < size; i++) { |
| if (!webKitWebSrcSetExtraHeader(fieldId, gst_value_list_get_value(value, i), userData)) |
| return FALSE; |
| } |
| return TRUE; |
| } |
| |
| return webKitWebSrcSetExtraHeader(fieldId, value, userData); |
| } |
| |
| static gboolean webKitWebSrcStart(GstBaseSrc* baseSrc) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| |
| if (src->priv->originalURI.isNull()) { |
| GST_ERROR_OBJECT(src, "No URI provided"); |
| return FALSE; |
| } |
| return TRUE; |
| } |
| |
| static void webKitWebSrcMakeRequest(WebKitWebSrc* src, DataMutexLocker<WebKitWebSrcPrivate::StreamingMembers>& members) |
| { |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| ASSERT(!priv->originalURI.isNull()); |
| ASSERT(members->requestedPosition != members->stopPosition); |
| |
| GST_DEBUG_OBJECT(src, "Posting task to request R%u %s requestedPosition=%" G_GUINT64_FORMAT " stopPosition=%" G_GUINT64_FORMAT, members->requestNumber, priv->originalURI.data(), members->requestedPosition, members->stopPosition); |
| URL url = URL(URL(), priv->originalURI.data()); |
| |
| ResourceRequest request(url); |
| request.setAllowCookies(true); |
| request.setFirstPartyForCookies(url); |
| |
| request.setHTTPReferrer(members->referrer); |
| |
| if (priv->httpMethod.get()) |
| request.setHTTPMethod(priv->httpMethod.get()); |
| |
| #if USE(SOUP) |
| // By default, HTTP Accept-Encoding is disabled here as we don't |
| // want the received response to be encoded in any way as we need |
| // to rely on the proper size of the returned data on |
| // didReceiveResponse. |
| // If Accept-Encoding is used, the server may send the data in encoded format and |
| // request.expectedContentLength() will have the "wrong" size (the size of the |
| // compressed data), even though the data received in didReceiveData is uncompressed. |
| // This is however useful to enable for adaptive streaming |
| // scenarios, when the demuxer needs to download playlists. |
| if (!priv->compress) |
| request.setAcceptEncoding(false); |
| #endif |
| |
| // Let Apple web servers know we want to access their nice movie trailers. |
| if (!g_ascii_strcasecmp("movies.apple.com", url.host().utf8().data()) |
| || !g_ascii_strcasecmp("trailers.apple.com", url.host().utf8().data())) |
| request.setHTTPUserAgent("Quicktime/7.6.6"); |
| |
| if (members->requestedPosition) { |
| GUniquePtr<char> formatedRange(g_strdup_printf("bytes=%" G_GUINT64_FORMAT "-", members->requestedPosition)); |
| GST_DEBUG_OBJECT(src, "Range request: %s", formatedRange.get()); |
| request.setHTTPHeaderField(HTTPHeaderName::Range, formatedRange.get()); |
| } |
| ASSERT(members->readPosition == members->requestedPosition); |
| |
| GST_DEBUG_OBJECT(src, "Persistent connection support %s", priv->keepAlive ? "enabled" : "disabled"); |
| if (!priv->keepAlive) |
| request.setHTTPHeaderField(HTTPHeaderName::Connection, "close"); |
| |
| if (priv->extraHeaders) |
| gst_structure_foreach(priv->extraHeaders.get(), webKitWebSrcProcessExtraHeaders, &request); |
| |
| // We always request Icecast/Shoutcast metadata, just in case ... |
| request.setHTTPHeaderField(HTTPHeaderName::IcyMetadata, "1"); |
| |
| ASSERT(!isMainThread()); |
| RunLoop::main().dispatch([protector = WTF::ensureGRef(src), request = WTFMove(request), requestNumber = members->requestNumber] { |
| WebKitWebSrcPrivate* priv = protector->priv; |
| DataMutexLocker members { priv->dataMutex }; |
| // Ignore this task (not making any HTTP request) if by now WebKitWebSrc streaming thread is already waiting |
| // for a different request. There is no point anymore in sending this one. |
| if (members->requestNumber != requestNumber) { |
| GST_DEBUG_OBJECT(protector.get(), "Skipping R%u, current request number is %u", requestNumber, members->requestNumber); |
| return; |
| } |
| |
| if (!members->loader) |
| members->loader = members->player->createResourceLoader(); |
| |
| PlatformMediaResourceLoader::LoadOptions loadOptions = 0; |
| members->resource = members->loader->requestResource(ResourceRequest(request), loadOptions); |
| if (members->resource) { |
| members->resource->setClient(adoptRef(*new CachedResourceStreamingClient(protector.get(), ResourceRequest(request), requestNumber))); |
| GST_DEBUG_OBJECT(protector.get(), "Started request R%u", requestNumber); |
| } else { |
| GST_ERROR_OBJECT(protector.get(), "Failed to setup streaming client to handle R%u", requestNumber); |
| members->loader = nullptr; |
| } |
| }); |
| } |
| |
| static gboolean webKitWebSrcStop(GstBaseSrc* baseSrc) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| // basesrc will always call unLock() and unLockStop() before calling this. See gst_base_src_stop(). |
| DataMutexLocker members { src->priv->dataMutex }; |
| webkitWebSrcReset(src, members, ResetType::Hard); |
| GST_DEBUG_OBJECT(src, "Stopped WebKitWebSrc"); |
| return TRUE; |
| } |
| |
| static gboolean webKitWebSrcGetSize(GstBaseSrc* baseSrc, guint64* size) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| DataMutexLocker members { src->priv->dataMutex }; |
| |
| GST_DEBUG_OBJECT(src, "haveSize: %s, size: %" G_GUINT64_FORMAT, boolForPrinting(members->haveSize), members->size); |
| if (members->haveSize) { |
| *size = members->size; |
| return TRUE; |
| } |
| |
| return FALSE; |
| } |
| |
| static gboolean webKitWebSrcIsSeekable(GstBaseSrc* baseSrc) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| DataMutexLocker members { src->priv->dataMutex }; |
| GST_DEBUG_OBJECT(src, "isSeekable: %s", boolForPrinting(members->isSeekable)); |
| return members->isSeekable; |
| } |
| |
| static gboolean webKitWebSrcDoSeek(GstBaseSrc* baseSrc, GstSegment* segment) |
| { |
| // This function is mutually exclusive with create(). It's only called when we're transitioning to >=PAUSED and |
| // between flushes. In any case, basesrc holds the STREAM_LOCK, so we know create() is not running. |
| // Also, both webKitWebSrcUnLock() and webKitWebSrcUnLockStop() are guaranteed to be called *before* this function. |
| // [See gst_base_src_perform_seek()]. |
| ASSERT(GST_ELEMENT(baseSrc)->current_state < GST_STATE_PAUSED || GST_PAD_IS_FLUSHING(baseSrc->srcpad)); |
| |
| // Except for the initial seek, this function is only called if isSeekable() returns true. |
| ASSERT(GST_ELEMENT(baseSrc)->current_state < GST_STATE_PAUSED || webKitWebSrcIsSeekable(baseSrc)); |
| |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| DataMutexLocker members { src->priv->dataMutex }; |
| |
| GST_DEBUG_OBJECT(src, "Seek segment: (%" G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT ") Position previous to seek: %" G_GUINT64_FORMAT, segment->start, segment->stop, members->readPosition); |
| |
| // Before attempting to seek, basesrc will call isSeekable(). If isSeekable() is true, a flush will be made and |
| // this function will be called. basesrc still gives us the chance here to return FALSE and cancel the seek. |
| // We cannot afford to return FALSE in this function though unless we're going to fail on purpose, since at this |
| // point we have already been flushed and cancelled the HTTP request that was feeding us data. |
| |
| if (segment->rate < 0.0 || segment->format != GST_FORMAT_BYTES) { |
| GST_ERROR_OBJECT(src, "Invalid seek segment"); |
| return FALSE; |
| } |
| |
| if (members->haveSize && segment->start >= members->size) |
| GST_WARNING_OBJECT(src, "Potentially seeking behind end of file, might EOS immediately"); |
| |
| members->requestedPosition = members->readPosition = segment->start; |
| members->stopPosition = segment->stop; |
| return TRUE; |
| } |
| |
| static gboolean webKitWebSrcQuery(GstBaseSrc* baseSrc, GstQuery* query) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| WebKitWebSrcPrivate* priv = src->priv; |
| gboolean result = FALSE; |
| |
| if (GST_QUERY_TYPE(query) == GST_QUERY_URI) { |
| gst_query_set_uri(query, priv->originalURI.data()); |
| DataMutexLocker members { src->priv->dataMutex }; |
| if (!members->redirectedURI.isNull()) |
| gst_query_set_uri_redirection(query, members->redirectedURI.data()); |
| result = TRUE; |
| } |
| |
| if (!result) |
| result = GST_BASE_SRC_CLASS(parent_class)->query(baseSrc, query); |
| |
| if (GST_QUERY_TYPE(query) == GST_QUERY_SCHEDULING) { |
| GstSchedulingFlags flags; |
| int minSize, maxSize, align; |
| |
| gst_query_parse_scheduling(query, &flags, &minSize, &maxSize, &align); |
| gst_query_set_scheduling(query, static_cast<GstSchedulingFlags>(flags | GST_SCHEDULING_FLAG_BANDWIDTH_LIMITED), minSize, maxSize, align); |
| } |
| |
| return result; |
| } |
| |
| static gboolean webKitWebSrcUnLock(GstBaseSrc* baseSrc) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| DataMutexLocker members { src->priv->dataMutex }; |
| |
| GST_DEBUG_OBJECT(src, "Unlock"); |
| members->isFlushing = true; |
| |
| // If we have a network resource request open, we ask the main thread to close it. |
| if (members->resource) { |
| GST_DEBUG_OBJECT(src, "Resource request R%u will be stopped", members->requestNumber); |
| RunLoop::main().dispatch([protector = WTF::ensureGRef(src), resource = WTFMove(members->resource), requestNumber = members->requestNumber] { |
| GST_DEBUG_OBJECT(protector.get(), "Stopping resource request R%u", requestNumber); |
| resource->stop(); |
| resource->setClient(nullptr); |
| }); |
| } |
| ASSERT(!members->resource); |
| |
| if (!src->priv->keepAlive) |
| members->loader = nullptr; |
| |
| // Ensure all network callbacks from the old request don't feed data to WebKitWebSrc anymore. |
| members->requestNumber++; |
| |
| // Wake up streaming thread. |
| members->responseCondition.notifyOne(); |
| |
| return TRUE; |
| } |
| |
| static gboolean webKitWebSrcUnLockStop(GstBaseSrc* baseSrc) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(baseSrc); |
| DataMutexLocker members { src->priv->dataMutex }; |
| GST_DEBUG_OBJECT(src, "Unlock stop"); |
| members->isFlushing = false; |
| webkitWebSrcReset(src, members, ResetType::Soft); |
| |
| return TRUE; |
| } |
| |
| static bool urlHasSupportedProtocol(const URL& url) |
| { |
| return url.isValid() && (url.protocolIsInHTTPFamily() || url.protocolIsBlob()); |
| } |
| |
| // uri handler interface |
| |
| static GstURIType webKitWebSrcUriGetType(GType) |
| { |
| return GST_URI_SRC; |
| } |
| |
| const gchar* const* webKitWebSrcGetProtocols(GType) |
| { |
| static const char* protocols[4]; |
| protocols[0] = "http"; |
| protocols[1] = "https"; |
| protocols[2] = "blob"; |
| protocols[3] = nullptr; |
| return protocols; |
| } |
| |
| static URL convertPlaybinURI(const char* uriString) |
| { |
| URL url(URL(), uriString); |
| return url; |
| } |
| |
| static gchar* webKitWebSrcGetUri(GstURIHandler* handler) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(handler); |
| gchar* ret = g_strdup(src->priv->originalURI.data()); |
| return ret; |
| } |
| |
| static gboolean webKitWebSrcSetUri(GstURIHandler* handler, const gchar* uri, GError** error) |
| { |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(handler); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| if (GST_STATE(src) >= GST_STATE_PAUSED) { |
| GST_ERROR_OBJECT(src, "URI can only be set in states < PAUSED"); |
| return FALSE; |
| } |
| |
| priv->originalURI = CString(); |
| if (!uri) |
| return TRUE; |
| |
| if (priv->originalURI.length()) { |
| GST_ERROR_OBJECT(src, "URI can only be set in states < PAUSED"); |
| return FALSE; |
| } |
| |
| URL url = convertPlaybinURI(uri); |
| |
| if (!urlHasSupportedProtocol(url)) { |
| g_set_error(error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Invalid URI '%s'", uri); |
| return FALSE; |
| } |
| |
| priv->originalURI = url.string().utf8(); |
| return TRUE; |
| } |
| |
| static void webKitWebSrcUriHandlerInit(gpointer gIface, gpointer) |
| { |
| GstURIHandlerInterface* iface = (GstURIHandlerInterface *) gIface; |
| |
| iface->get_type = webKitWebSrcUriGetType; |
| iface->get_protocols = webKitWebSrcGetProtocols; |
| iface->get_uri = webKitWebSrcGetUri; |
| iface->set_uri = webKitWebSrcSetUri; |
| } |
| |
| void webKitWebSrcSetMediaPlayer(WebKitWebSrc* src, WebCore::MediaPlayer* player, const String& referrer) |
| { |
| ASSERT(player); |
| DataMutexLocker members { src->priv->dataMutex }; |
| members->player = player; |
| members->referrer = referrer; |
| } |
| |
| bool webKitSrcPassedCORSAccessCheck(WebKitWebSrc* src) |
| { |
| DataMutexLocker members { src->priv->dataMutex }; |
| return members->didPassAccessControlCheck; |
| } |
| |
| CachedResourceStreamingClient::CachedResourceStreamingClient(WebKitWebSrc* src, ResourceRequest&& request, unsigned requestNumber) |
| : m_requestNumber(requestNumber) |
| , m_src(GST_ELEMENT(src)) |
| , m_request(WTFMove(request)) |
| { |
| } |
| |
| CachedResourceStreamingClient::~CachedResourceStreamingClient() = default; |
| |
| void CachedResourceStreamingClient::checkUpdateBlocksize(unsigned bytesRead) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| GstBaseSrc* baseSrc = GST_BASE_SRC_CAST(src); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| unsigned blocksize = gst_base_src_get_blocksize(baseSrc); |
| GST_LOG_OBJECT(src, "Checking to update blocksize. Read: %u, current blocksize: %u", bytesRead, blocksize); |
| |
| if (bytesRead > blocksize * s_growBlocksizeLimit) { |
| m_reduceBlocksizeCount = 0; |
| m_increaseBlocksizeCount++; |
| |
| if (m_increaseBlocksizeCount >= s_growBlocksizeCount) { |
| blocksize *= s_growBlocksizeFactor; |
| GST_DEBUG_OBJECT(src, "Increased blocksize to %u", blocksize); |
| gst_base_src_set_blocksize(baseSrc, blocksize); |
| m_increaseBlocksizeCount = 0; |
| } |
| } else if (bytesRead < blocksize * s_reduceBlocksizeLimit) { |
| m_reduceBlocksizeCount++; |
| m_increaseBlocksizeCount = 0; |
| |
| if (m_reduceBlocksizeCount >= s_reduceBlocksizeCount) { |
| blocksize *= s_reduceBlocksizeFactor; |
| blocksize = std::max(blocksize, priv->minimumBlocksize); |
| GST_DEBUG_OBJECT(src, "Decreased blocksize to %u", blocksize); |
| gst_base_src_set_blocksize(baseSrc, blocksize); |
| m_reduceBlocksizeCount = 0; |
| } |
| } else { |
| m_reduceBlocksizeCount = 0; |
| m_increaseBlocksizeCount = 0; |
| } |
| } |
| |
| void CachedResourceStreamingClient::responseReceived(PlatformMediaResource&, const ResourceResponse& response, CompletionHandler<void(ShouldContinuePolicyCheck)>&& completionHandler) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| WebKitWebSrcPrivate* priv = src->priv; |
| DataMutexLocker members { priv->dataMutex }; |
| if (members->requestNumber != m_requestNumber) { |
| completionHandler(ShouldContinuePolicyCheck::No); |
| return; |
| } |
| |
| GST_DEBUG_OBJECT(src, "R%u: Received response: %d", m_requestNumber, response.httpStatusCode()); |
| |
| members->didPassAccessControlCheck = members->resource->didPassAccessControlCheck(); |
| m_origins.add(SecurityOrigin::create(response.url())); |
| |
| auto responseURI = response.url().string().utf8(); |
| if (priv->originalURI != responseURI) |
| members->redirectedURI = WTFMove(responseURI); |
| |
| // length will be zero (unknown) if no Content-Length is provided or the response is compressed with Content-Encoding. |
| uint64_t length = !response.httpHeaderFields().contains(HTTPHeaderName::ContentEncoding) ? response.expectedContentLength() : 0; |
| if (length > 0 && members->requestedPosition && response.httpStatusCode() == 206) |
| length += members->requestedPosition; |
| |
| GUniquePtr<GstStructure> httpHeaders(gst_structure_new_empty("http-headers")); |
| |
| gst_structure_set(httpHeaders.get(), "uri", G_TYPE_STRING, priv->originalURI.data(), |
| "http-status-code", G_TYPE_UINT, response.httpStatusCode(), nullptr); |
| if (!members->redirectedURI.isNull()) |
| gst_structure_set(httpHeaders.get(), "redirection-uri", G_TYPE_STRING, members->redirectedURI.data(), nullptr); |
| |
| // Pack request headers in the http-headers structure. |
| GUniquePtr<GstStructure> headers(gst_structure_new_empty("request-headers")); |
| for (const auto& header : m_request.httpHeaderFields()) |
| gst_structure_set(headers.get(), header.key.utf8().data(), G_TYPE_STRING, header.value.utf8().data(), nullptr); |
| GST_DEBUG_OBJECT(src, "R%u: Request headers going downstream: %" GST_PTR_FORMAT, m_requestNumber, headers.get()); |
| gst_structure_set(httpHeaders.get(), "request-headers", GST_TYPE_STRUCTURE, headers.get(), nullptr); |
| |
| // Pack response headers in the http-headers structure. |
| headers.reset(gst_structure_new_empty("response-headers")); |
| for (const auto& header : response.httpHeaderFields()) { |
| if (auto convertedValue = parseIntegerAllowingTrailingJunk<uint64_t>(header.value)) |
| gst_structure_set(headers.get(), header.key.utf8().data(), G_TYPE_UINT64, *convertedValue, nullptr); |
| else |
| gst_structure_set(headers.get(), header.key.utf8().data(), G_TYPE_STRING, header.value.utf8().data(), nullptr); |
| } |
| GST_DEBUG_OBJECT(src, "R%u: Response headers going downstream: %" GST_PTR_FORMAT, m_requestNumber, headers.get()); |
| gst_structure_set(httpHeaders.get(), "response-headers", GST_TYPE_STRUCTURE, headers.get(), nullptr); |
| |
| members->pendingHttpHeadersMessage = adoptGRef(gst_message_new_element(GST_OBJECT_CAST(src), gst_structure_copy(httpHeaders.get()))); |
| members->pendingHttpHeadersEvent = adoptGRef(gst_event_new_custom(GST_EVENT_CUSTOM_DOWNSTREAM_STICKY, httpHeaders.release())); |
| |
| if (response.httpStatusCode() >= 400) { |
| GST_ELEMENT_ERROR(src, RESOURCE, READ, ("R%u: Received %d HTTP error code", m_requestNumber, response.httpStatusCode()), (nullptr)); |
| members->doesHaveEOS = true; |
| members->responseCondition.notifyOne(); |
| completionHandler(ShouldContinuePolicyCheck::No); |
| return; |
| } |
| |
| if (members->requestedPosition) { |
| // Seeking ... we expect a 206 == PARTIAL_CONTENT |
| if (response.httpStatusCode() != 206) { |
| // Range request completely failed. |
| GST_ELEMENT_ERROR(src, RESOURCE, READ, ("R%u: Received unexpected %d HTTP status code for range request", m_requestNumber, response.httpStatusCode()), (nullptr)); |
| members->doesHaveEOS = true; |
| members->responseCondition.notifyOne(); |
| completionHandler(ShouldContinuePolicyCheck::No); |
| return; |
| } |
| GST_DEBUG_OBJECT(src, "R%u: Range request succeeded", m_requestNumber); |
| } |
| |
| members->isSeekable = length > 0 && g_ascii_strcasecmp("none", response.httpHeaderField(HTTPHeaderName::AcceptRanges).utf8().data()); |
| |
| GST_DEBUG_OBJECT(src, "R%u: Size: %" G_GUINT64_FORMAT ", isSeekable: %s", m_requestNumber, length, boolForPrinting(members->isSeekable)); |
| if (length > 0) { |
| if (!members->haveSize || members->size != length) { |
| members->haveSize = true; |
| members->size = length; |
| } |
| } else |
| members->haveSize = false; |
| |
| // Signal to downstream if this is an Icecast stream. |
| GRefPtr<GstCaps> caps; |
| if (auto metadataInterval = parseIntegerAllowingTrailingJunk<int>(response.httpHeaderField(HTTPHeaderName::IcyMetaInt)); metadataInterval && *metadataInterval > 0) { |
| caps = adoptGRef(gst_caps_new_simple("application/x-icy", "metadata-interval", G_TYPE_INT, *metadataInterval, nullptr)); |
| |
| String contentType = response.httpHeaderField(HTTPHeaderName::ContentType); |
| GST_DEBUG_OBJECT(src, "R%u: Response ContentType: %s", m_requestNumber, contentType.utf8().data()); |
| gst_caps_set_simple(caps.get(), "content-type", G_TYPE_STRING, contentType.utf8().data(), nullptr); |
| } |
| if (caps) { |
| GST_DEBUG_OBJECT(src, "R%u: Set caps to %" GST_PTR_FORMAT, m_requestNumber, caps.get()); |
| members->pendingCaps = WTFMove(caps); |
| } |
| |
| members->wasResponseReceived = true; |
| members->responseCondition.notifyOne(); |
| |
| completionHandler(ShouldContinuePolicyCheck::Yes); |
| } |
| |
| void CachedResourceStreamingClient::dataReceived(PlatformMediaResource&, Ref<SharedBuffer>&& buffer) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| WebKitWebSrcPrivate* priv = src->priv; |
| |
| DataMutexLocker members { priv->dataMutex }; |
| if (members->requestNumber != m_requestNumber) |
| return; |
| |
| // Rough bandwidth calculation. We ignore here the first data package because we would have to reset the counters when we issue the request and |
| // that first package delivery would include the time of sending out the request and getting the data back. Since we can't distinguish the |
| // sending time from the receiving time, it is better to ignore it. |
| if (!std::isnan(members->downloadStartTime)) { |
| members->totalDownloadedBytes += buffer->size(); |
| double timeSinceStart = (WallTime::now() - members->downloadStartTime).seconds(); |
| GST_TRACE_OBJECT(src, "R%u: downloaded %" G_GUINT64_FORMAT " bytes in %f seconds =~ %1.0f bytes/second", m_requestNumber, members->totalDownloadedBytes, timeSinceStart |
| , timeSinceStart ? members->totalDownloadedBytes / timeSinceStart : 0); |
| } else { |
| members->downloadStartTime = WallTime::now(); |
| } |
| |
| buffer->forEachSegment([&](auto& segment) { |
| int length = segment.size(); |
| GST_LOG_OBJECT(src, "R%u: Have %d bytes of data", m_requestNumber, length); |
| |
| members->readPosition += length; |
| ASSERT(!members->haveSize || members->readPosition <= members->size); |
| |
| gst_element_post_message(GST_ELEMENT_CAST(src), gst_message_new_element(GST_OBJECT_CAST(src), |
| gst_structure_new("webkit-network-statistics", "read-position", G_TYPE_UINT64, members->readPosition, "size", G_TYPE_UINT64, members->size, nullptr))); |
| |
| checkUpdateBlocksize(length); |
| GstBuffer* buffer = gstBufferNewWrappedFast(fastMemDup(segment.data(), length), length); |
| gst_adapter_push(members->adapter.get(), buffer); |
| }); |
| stopLoaderIfNeeded(src, members); |
| members->responseCondition.notifyOne(); |
| } |
| |
| void CachedResourceStreamingClient::accessControlCheckFailed(PlatformMediaResource&, const ResourceError& error) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| DataMutexLocker members { src->priv->dataMutex }; |
| if (members->requestNumber != m_requestNumber) |
| return; |
| |
| GST_ELEMENT_ERROR(src, RESOURCE, READ, ("R%u: %s", m_requestNumber, error.localizedDescription().utf8().data()), (nullptr)); |
| members->doesHaveEOS = true; |
| members->responseCondition.notifyOne(); |
| } |
| |
| void CachedResourceStreamingClient::loadFailed(PlatformMediaResource&, const ResourceError& error) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| DataMutexLocker members { src->priv->dataMutex }; |
| if (members->requestNumber != m_requestNumber) |
| return; |
| |
| if (!error.isCancellation()) { |
| GST_ERROR_OBJECT(src, "R%u: Have failure: %s", m_requestNumber, error.localizedDescription().utf8().data()); |
| GST_ELEMENT_ERROR(src, RESOURCE, FAILED, ("R%u: %s", m_requestNumber, error.localizedDescription().utf8().data()), (nullptr)); |
| } else |
| GST_LOG_OBJECT(src, "R%u: Request cancelled: %s", m_requestNumber, error.localizedDescription().utf8().data()); |
| |
| members->doesHaveEOS = true; |
| members->responseCondition.notifyOne(); |
| } |
| |
| void CachedResourceStreamingClient::loadFinished(PlatformMediaResource&, const NetworkLoadMetrics&) |
| { |
| ASSERT(isMainThread()); |
| WebKitWebSrc* src = WEBKIT_WEB_SRC(m_src.get()); |
| DataMutexLocker members { src->priv->dataMutex }; |
| if (members->requestNumber != m_requestNumber) |
| return; |
| |
| GST_LOG_OBJECT(src, "R%u: Load finished. Read position: %" G_GUINT64_FORMAT, m_requestNumber, members->readPosition); |
| |
| members->doesHaveEOS = true; |
| members->responseCondition.notifyOne(); |
| } |
| |
| bool webKitSrcWouldTaintOrigin(WebKitWebSrc* src, const SecurityOrigin& origin) |
| { |
| DataMutexLocker members { src->priv->dataMutex }; |
| |
| auto* cachedResourceStreamingClient = reinterpret_cast<CachedResourceStreamingClient*>(members->resource->client()); |
| for (auto& responseOrigin : cachedResourceStreamingClient->securityOrigins()) { |
| if (!origin.isSameOriginDomain(*responseOrigin)) |
| return true; |
| } |
| return false; |
| } |
| |
| #endif // ENABLE(VIDEO) && USE(GSTREAMER) |