Bug 162901

Summary: [GStreamer][MSE] Playback pipeline
Product: WebKit Reporter: Enrique Ocaña <eocanha>
Component: PlatformAssignee: Enrique Ocaña <eocanha>
Status: RESOLVED FIXED    
Severity: Normal CC: calvaris
Priority: P2    
Version: WebKit Local Build   
Hardware: Unspecified   
OS: Unspecified   
Bug Depends on:    
Bug Blocks: 157314    
Attachments:
Description Flags
Patch
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Patch
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Patch none

Enrique Ocaña
Reported 2016-10-04 05:13:36 PDT
Encapsulate the responsibility to interact with the GStreamer playback pipeline and the different streams managed by the WebKitMediaSrc element into a new PlaybackPipeline class.
Attachments
Patch (26.51 KB, patch)
2016-10-04 05:41 PDT, Enrique Ocaña
no flags
Patch (26.03 KB, patch)
2016-10-16 12:07 PDT, Enrique Ocaña
no flags
Patch (24.98 KB, patch)
2016-10-21 14:27 PDT, Enrique Ocaña
no flags
Patch (24.99 KB, patch)
2016-10-21 14:39 PDT, Enrique Ocaña
no flags
Patch (23.04 KB, patch)
2016-10-25 11:04 PDT, Enrique Ocaña
no flags
Patch (23.04 KB, patch)
2016-10-26 01:21 PDT, Enrique Ocaña
no flags
Enrique Ocaña
Comment 1 2016-10-04 05:41:11 PDT
Enrique Ocaña
Comment 2 2016-10-04 07:08:27 PDT
Comment on attachment 290595 [details] Patch Wait until all the patches in 157314 are ready.
Xabier Rodríguez Calvar
Comment 3 2016-10-05 06:12:45 PDT
Comment on attachment 290595 [details] Patch View in context: https://bugs.webkit.org/attachment.cgi?id=290595&action=review > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:40 > +#include <wtf/MainThread.h> > +#include <wtf/RefCounted.h> > +#include <wtf/glib/GMutexLocker.h> > +#include <wtf/glib/GRefPtr.h> > +#include <wtf/glib/GUniquePtr.h> These lines are not sorted > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:72 > + GstBuffer* buffer; > + GstCaps* caps; Move these variables to the first place they are used. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:128 > + GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "State %d", (int)GST_STATE(m_webKitMediaSrc.get())); Do not use C-like casts, please. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:147 > + gst_app_src_set_callbacks(GST_APP_SRC(stream->appsrc), &enabledAppsrcCallbacks, stream->parent, 0); 0 -> nullptr > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:180 > + Deque<Stream*>::iterator streamPosition = priv->streams.begin(); > + > + while (streamPosition != priv->streams.end()) { > + if ((*streamPosition)->sourceBuffer == sourceBufferPrivate.get()) { > + stream = *streamPosition; > + break; > + } > + ++streamPosition; > + } I think it is better to use a for loop and if possible a C++11 one. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:192 > + Stream* stream = nullptr; You don't need to initialize Stream here. Actually you can declare it when you assign it later. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:193 > + unsigned padId = 0; Ditto > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:194 > + const gchar* mediaType = gst_structure_get_name(structure); Move to before it is used. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:207 > + GUniquePtr<gchar> parserBinName(g_strdup_printf("streamparser%u", padId)); Declare this after the GST_DEBUG_OBJECT > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:215 > + GstElement* parser; > + GstElement* capsfilter; > + GstPad* pad = nullptr; > + GRefPtr<GstCaps> filterCaps = adoptGRef(gst_caps_new_simple("video/x-h264", "alignment", G_TYPE_STRING, "au", nullptr)); Declare these when used. pad should be a GRefPtr. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:236 > + GstElement* parser; > + GstElement* capsfilter; > + GstPad* pad = nullptr; > + GRefPtr<GstCaps> filterCaps = adoptGRef(gst_caps_new_simple("video/x-h265", "alignment", G_TYPE_STRING, "au", nullptr)); Declare these when used. pad should be a GRefPtr. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:256 > + GstElement* parser; > + GstPad* pad = nullptr; Declare these when used. pad should be a GRefPtr. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:261 > + if (mpegversion == 1) { > + parser = gst_element_factory_make("mpegaudioparse", nullptr); > + } else if (mpegversion == 2 || mpegversion == 4) You don't need the { } > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:352 > + ASSERT(stream); > + ASSERT(stream->type != Invalid); You don't need the first ASSERT. Either you shortcircuit with stream && ... or you just use the second cause a segmentation fault is going to have almost the same result as the assertion. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:357 > + // gst_app_src_set_caps(GST_APP_SRC(stream->appsrc), caps); Why is this commented out? If it is not useful, then we can remove it. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:364 > +#if (!(LOG_DISABLED || GST_DISABLE_GST_DEBUG)) You don't need LOG_DISABLED here as we don't use LOG_* anymore. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:366 > + gchar* stroldcaps = gst_caps_to_string(oldAppsrcCaps.get()); > + gchar* strnewcaps = gst_caps_to_string(appsrcCaps.get()); GUniquePtr > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:430 > + for (Vector<GstAppSrc*>::iterator iterator = appsrcs.begin(); iterator != appsrcs.end(); ++iterator) > + gst_app_src_end_of_stream(*iterator); Use C++11 for if possible? > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:468 > + for (Vector<RefPtr<MediaSample>>::iterator iterator = samples.begin(); iterator != samples.end(); ++iterator) { Use C++11 for if possible? > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:496 > + GST_TRACE("enqueing sample trackId=%s PTS=%f presentationSize=%.0fx%.0f at %" GST_TIME_FORMAT " duration: %" GST_TIME_FORMAT, trackId.string().utf8().data(), protectedPrSample->presentationTime().toFloat(), protectedPrSample->presentationSize().width(), protectedPrSample->presentationSize().height(), GST_TIME_ARGS(WebCore::toGstClockTime(protectedPrSample->presentationTime().toDouble())), GST_TIME_ARGS(WebCore::toGstClockTime(protectedPrSample->duration().toDouble()))); In WebKit we accept long times, but this is probably too much :) > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:515 > + GstSample* gstSample = gst_sample_ref(sample->sample()); GRefPtr > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.h:32 > + Remove extra line > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.h:65 > + // From MediaSourceGStreamer > + void markEndOfStream(MediaSourcePrivate::EndOfStreamStatus); > + > + // From SourceBufferPrivateGStreamer Periods.
Enrique Ocaña
Comment 4 2016-10-16 12:07:19 PDT
Zan Dobersek
Comment 5 2016-10-21 00:56:16 PDT
Comment on attachment 291764 [details] Patch View in context: https://bugs.webkit.org/attachment.cgi?id=291764&action=review > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:94 > +PassRefPtr<PlaybackPipeline> PlaybackPipeline::create() > +{ > + return adoptRef(new PlaybackPipeline()); > +} Return Ref<>. This is usually implemented in the header. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:99 > +PlaybackPipeline::PlaybackPipeline() > + : RefCounted<PlaybackPipeline>() > +{ > +} The RefCounted<> initialization is always implicit. This constructor definition can be defaulted: PlaybackPipeline::PlaybackPipeline() = default; > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:103 > +PlaybackPipeline::~PlaybackPipeline() > +{ > +} Destructor can be defaulted as well. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:191 > + GST_OBJECT_LOCK(webKitMediaSrc); > + Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get()); > + GST_OBJECT_UNLOCK(webKitMediaSrc); I don't think scoped locks are used much with GstObject locks in the current GStreamer code, but IMO they should be. Something for a follow-up patch. > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:314 > + if (g_str_has_prefix(mediaType, "audio")) { > + GST_OBJECT_LOCK(webKitMediaSrc); > + stream->type = Audio; > + stream->parent->priv->numberOfAudioStreams++; > + GST_OBJECT_UNLOCK(webKitMediaSrc); > + signal = SIGNAL_AUDIO_CHANGED; > + > + stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get())); > + } else if (g_str_has_prefix(mediaType, "video")) { > + GST_OBJECT_LOCK(webKitMediaSrc); > + stream->type = Video; > + stream->parent->priv->numberOfVideoStreams++; > + GST_OBJECT_UNLOCK(webKitMediaSrc); > + signal = SIGNAL_VIDEO_CHANGED; > + > + stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get())); > + } else if (g_str_has_prefix(mediaType, "text")) { > + GST_OBJECT_LOCK(webKitMediaSrc); > + stream->type = Text; > + stream->parent->priv->numberOfTextStreams++; > + GST_OBJECT_UNLOCK(webKitMediaSrc); > + signal = SIGNAL_TEXT_CHANGED; > + > + // FIXME: Support text tracks. > + } Can the webKitMediaSrc object lock be held for this whole block? > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:350 > + GRefPtr<GstCaps> oldAppsrcCaps = adoptGRef(gst_app_src_get_caps(GST_APP_SRC(stream->appsrc))); > + // The caps change is managed by gst_appsrc_push_sample() in enqueueSample() and > + // flushAndEnqueueNonDisplayingSamples(), so the caps aren't set from here. > + GRefPtr<GstCaps> appsrcCaps = adoptGRef(gst_app_src_get_caps(GST_APP_SRC(stream->appsrc))); > + const gchar* mediaType = gst_structure_get_name(gst_caps_get_structure(appsrcCaps.get(), 0)); > + > + if (!gst_caps_is_equal(oldAppsrcCaps.get(), appsrcCaps.get())) { > + GST_DEBUG("Caps have changed, but reconstructing the sequence of elements is not supported yet"); > + > +#ifndef GST_DISABLE_GST_DEBUG > + GUniquePtr<gchar> stroldcaps(gst_caps_to_string(oldAppsrcCaps.get())); > + GUniquePtr<gchar> strnewcaps(gst_caps_to_string(appsrcCaps.get())); > + GST_DEBUG("oldcaps: %s", stroldcaps.get()); > + GST_DEBUG("newcaps: %s", strnewcaps.get()); > +#endif > + } What's the point of this? How can the appsrc caps change between the two gst_app_src_get_caps() calls? > Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp:471 > + RefPtr<MediaSample> protectedPrSample = prSample; The method should be receiving RefPtr<> as an argument.
Enrique Ocaña
Comment 6 2016-10-21 14:27:33 PDT
Enrique Ocaña
Comment 7 2016-10-21 14:39:18 PDT
Enrique Ocaña
Comment 8 2016-10-25 11:04:31 PDT
Enrique Ocaña
Comment 9 2016-10-26 01:21:21 PDT
Enrique Ocaña
Comment 10 2016-10-26 01:45:31 PDT
Comment on attachment 292895 [details] Patch Clearing flags on attachment: 292895 Committed r207882: <http://trac.webkit.org/changeset/207882>
Enrique Ocaña
Comment 11 2016-10-26 01:45:39 PDT
All reviewed patches have been landed. Closing bug.
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