webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 108 insertions, 114 deletions

diff --git a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
index ff672f371..445c9793c 100644
--- a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
@@ -1,5 +1,6 @@
 /*
  *  Copyright (C) 2011, 2012 Igalia S.L
+ *  Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.com>
  *
  *  This library is free software; you can redistribute it and/or
  *  modify it under the terms of the GNU Lesser General Public
@@ -26,9 +27,10 @@
 #include "AudioIOCallback.h"
 #include "GRefPtrGStreamer.h"
 #include "GStreamerUtilities.h"
-#include <gst/audio/audio.h>
-#include <gst/pbutils/pbutils.h>
-#include <wtf/gobject/GUniquePtr.h>
+#include <gst/app/gstappsrc.h>
+#include <gst/audio/audio-info.h>
+#include <gst/pbutils/missing-plugins.h>
+#include <wtf/glib/GUniquePtr.h>
 
 using namespace WebCore;
 
@@ -51,18 +53,22 @@ struct _WebKitWebAudioSourcePrivate {
     AudioBus* bus;
     AudioIOCallback* provider;
     guint framesToPull;
+    guint bufferSize;
 
     GRefPtr<GstElement> interleave;
-    GRefPtr<GstElement> wavEncoder;
 
     GRefPtr<GstTask> task;
     GRecMutex mutex;
 
-    GSList* pads; // List of queue sink pads. One queue for each planar audio channel.
-    GstPad* sourcePad; // src pad of the element, interleaved wav data is pushed to it.
+    // List of appsrc. One appsrc for each planar audio channel.
+    Vector<GRefPtr<GstElement>> sources;
 
-    bool newStreamEventPending;
-    GstSegment segment;
+    // src pad of the element, interleaved wav data is pushed to it.
+    GstPad* sourcePad;
+
+    guint64 numberOfSamples;
+
+    GRefPtr<GstBufferPool> pool;
 };
 
 enum {
@@ -73,9 +79,9 @@ enum {
 };
 
 static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src",
-                                                                  GST_PAD_SRC,
-                                                                  GST_PAD_ALWAYS,
-                                                                  GST_STATIC_CAPS("audio/x-wav"));
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS(GST_AUDIO_CAPS_MAKE(GST_AUDIO_NE(F32))));
 
 GST_DEBUG_CATEGORY_STATIC(webkit_web_audio_src_debug);
 #define GST_CAT_DEFAULT webkit_web_audio_src_debug
@@ -91,8 +97,8 @@ static GstCaps* getGStreamerMonoAudioCaps(float sampleRate)
 {
     return gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(sampleRate),
         "channels", G_TYPE_INT, 1,
-        "format", G_TYPE_STRING, gst_audio_format_to_string(GST_AUDIO_FORMAT_F32),
-        "layout", G_TYPE_STRING, "non-interleaved", NULL);
+        "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
+        "layout", G_TYPE_STRING, "interleaved", nullptr);
 }
 
 static GstAudioChannelPosition webKitWebAudioGStreamerChannelPosition(int channelIndex)
@@ -178,17 +184,14 @@ static void webkit_web_audio_src_init(WebKitWebAudioSrc* src)
     src->priv = priv;
     new (priv) WebKitWebAudioSourcePrivate();
 
-    priv->sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", 0);
+    priv->sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", nullptr);
     gst_element_add_pad(GST_ELEMENT(src), priv->sourcePad);
 
-    priv->provider = 0;
-    priv->bus = 0;
-
-    priv->newStreamEventPending = true;
-    gst_segment_init(&priv->segment, GST_FORMAT_TIME);
+    priv->provider = nullptr;
+    priv->bus = nullptr;
 
     g_rec_mutex_init(&priv->mutex);
-    priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0);
+    priv->task = adoptGRef(gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, nullptr));
 
     gst_task_set_lock(priv->task.get(), &priv->mutex);
 }
@@ -202,54 +205,40 @@ static void webKitWebAudioSrcConstructed(GObject* object)
     ASSERT(priv->provider);
     ASSERT(priv->sampleRate);
 
-    priv->interleave = gst_element_factory_make("interleave", 0);
-    priv->wavEncoder = gst_element_factory_make("wavenc", 0);
+    priv->interleave = gst_element_factory_make("interleave", nullptr);
 
     if (!priv->interleave) {
         GST_ERROR_OBJECT(src, "Failed to create interleave");
         return;
     }
 
-    if (!priv->wavEncoder) {
-        GST_ERROR_OBJECT(src, "Failed to create wavenc");
-        return;
-    }
-
-    gst_bin_add_many(GST_BIN(src), priv->interleave.get(), priv->wavEncoder.get(), NULL);
-    gst_element_link_pads_full(priv->interleave.get(), "src", priv->wavEncoder.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_bin_add(GST_BIN(src), priv->interleave.get());
 
     // For each channel of the bus create a new upstream branch for interleave, like:
-    // queue ! capsfilter ! audioconvert. which is plugged to a new interleave request sinkpad.
+    // appsrc ! . which is plugged to a new interleave request sinkpad.
     for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
-        GUniquePtr<gchar> queueName(g_strdup_printf("webaudioQueue%u", channelIndex));
-        GstElement* queue = gst_element_factory_make("queue", queueName.get());
-        GstElement* capsfilter = gst_element_factory_make("capsfilter", 0);
-        GstElement* audioconvert = gst_element_factory_make("audioconvert", 0);
-
+        GUniquePtr<gchar> appsrcName(g_strdup_printf("webaudioSrc%u", channelIndex));
+        GRefPtr<GstElement> appsrc = gst_element_factory_make("appsrc", appsrcName.get());
         GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
 
         GstAudioInfo info;
         gst_audio_info_from_caps(&info, monoCaps.get());
         GST_AUDIO_INFO_POSITION(&info, 0) = webKitWebAudioGStreamerChannelPosition(channelIndex);
         GRefPtr<GstCaps> caps = adoptGRef(gst_audio_info_to_caps(&info));
-        g_object_set(capsfilter, "caps", caps.get(), NULL);
-
-        // Configure the queue for minimal latency.
-        g_object_set(queue, "max-size-buffers", static_cast<guint>(1), NULL);
 
-        GstPad* pad = gst_element_get_static_pad(queue, "sink");
-        priv->pads = g_slist_prepend(priv->pads, pad);
+        // Configure the appsrc for minimal latency.
+        g_object_set(appsrc.get(), "max-bytes", static_cast<guint64>(2 * priv->bufferSize), "block", TRUE,
+            "blocksize", priv->bufferSize,
+            "format", GST_FORMAT_TIME, "caps", caps.get(), nullptr);
 
-        gst_bin_add_many(GST_BIN(src), queue, capsfilter, audioconvert, NULL);
-        gst_element_link_pads_full(queue, "src", capsfilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
-        gst_element_link_pads_full(capsfilter, "src", audioconvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
-        gst_element_link_pads_full(audioconvert, "src", priv->interleave.get(), 0, GST_PAD_LINK_CHECK_NOTHING);
+        priv->sources.append(appsrc);
 
+        gst_bin_add(GST_BIN(src), appsrc.get());
+        gst_element_link_pads_full(appsrc.get(), "src", priv->interleave.get(), "sink_%u", GST_PAD_LINK_CHECK_NOTHING);
     }
-    priv->pads = g_slist_reverse(priv->pads);
 
-    // wavenc's src pad is the only visible pad of our element.
-    GRefPtr<GstPad> targetPad = adoptGRef(gst_element_get_static_pad(priv->wavEncoder.get(), "src"));
+    // interleave's src pad is the only visible pad of our element.
+    GRefPtr<GstPad> targetPad = adoptGRef(gst_element_get_static_pad(priv->interleave.get(), "src"));
     gst_ghost_pad_set_target(GST_GHOST_PAD(priv->sourcePad), targetPad.get());
 }
 
@@ -260,8 +249,6 @@ static void webKitWebAudioSrcFinalize(GObject* object)
 
     g_rec_mutex_clear(&priv->mutex);
 
-    g_slist_free_full(priv->pads, reinterpret_cast<GDestroyNotify>(gst_object_unref));
-
     priv->~WebKitWebAudioSourcePrivate();
     GST_CALL_PARENT(G_OBJECT_CLASS, finalize, ((GObject* )(src)));
 }
@@ -283,6 +270,7 @@ static void webKitWebAudioSrcSetProperty(GObject* object, guint propertyId, cons
         break;
     case PROP_FRAMES:
         priv->framesToPull = g_value_get_uint(value);
+        priv->bufferSize = sizeof(float) * priv->framesToPull;
         break;
     default:
         G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propertyId, pspec);
@@ -320,68 +308,65 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
 
     ASSERT(priv->bus);
     ASSERT(priv->provider);
-    if (!priv->provider || !priv->bus)
+    if (!priv->provider || !priv->bus) {
+        GST_ELEMENT_ERROR(src, CORE, FAILED, ("Internal WebAudioSrc error"), ("Can't start without provider or bus"));
+        gst_task_stop(src->priv->task.get());
         return;
-
-    GSList* channelBufferList = 0;
-    register int i;
-    unsigned bufferSize = priv->framesToPull * sizeof(float);
-    for (i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
-        GstBuffer* channelBuffer = gst_buffer_new_and_alloc(bufferSize);
-        ASSERT(channelBuffer);
-        channelBufferList = g_slist_prepend(channelBufferList, channelBuffer);
-        GstMapInfo info;
-        gst_buffer_map(channelBuffer, &info, GST_MAP_READ);
-        priv->bus->setChannelMemory(i, reinterpret_cast<float*>(info.data), priv->framesToPull);
-        gst_buffer_unmap(channelBuffer, &info);
     }
 
-    // FIXME: Add support for local/live audio input.
-    priv->provider->render(0, priv->bus, priv->framesToPull);
-
-    GSList* padsIt = priv->pads;
-    GSList* buffersIt = channelBufferList;
-
-#if GST_CHECK_VERSION(1, 2, 0)
-    guint groupId = 0;
-    if (priv->newStreamEventPending)
-        groupId = gst_util_group_id_next();
-#endif
-
-    for (i = 0; padsIt && buffersIt; padsIt = g_slist_next(padsIt), buffersIt = g_slist_next(buffersIt), ++i) {
-        GstPad* pad = static_cast<GstPad*>(padsIt->data);
-        GstBuffer* channelBuffer = static_cast<GstBuffer*>(buffersIt->data);
-
-        // Send stream-start, segment and caps events downstream, along with the first buffer.
-        if (priv->newStreamEventPending) {
-            GRefPtr<GstElement> queue = adoptGRef(gst_pad_get_parent_element(pad));
-            GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue.get(), "sink"));
-            GUniquePtr<gchar> queueName(gst_element_get_name(queue.get()));
-            GUniquePtr<gchar> streamId(g_strdup_printf("webaudio/%s", queueName.get()));
-            GstEvent* streamStartEvent = gst_event_new_stream_start(streamId.get());
-#if GST_CHECK_VERSION(1, 2, 0)
-            gst_event_set_group_id(streamStartEvent, groupId);
-#endif
-            gst_pad_send_event(sinkPad.get(), streamStartEvent);
-
-            GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
-            GstAudioInfo info;
-            gst_audio_info_from_caps(&info, monoCaps.get());
-            GST_AUDIO_INFO_POSITION(&info, 0) = webKitWebAudioGStreamerChannelPosition(i);
-            GRefPtr<GstCaps> capsWithChannelPosition = adoptGRef(gst_audio_info_to_caps(&info));
-            gst_pad_send_event(sinkPad.get(), gst_event_new_caps(capsWithChannelPosition.get()));
-
-            gst_pad_send_event(sinkPad.get(), gst_event_new_segment(&priv->segment));
+    ASSERT(priv->pool);
+    GstClockTime timestamp = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate);
+    priv->numberOfSamples += priv->framesToPull;
+    GstClockTime duration = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate) - timestamp;
+
+    Vector<GRefPtr<GstBuffer>> channelBufferList;
+    channelBufferList.reserveInitialCapacity(priv->sources.size());
+    for (unsigned i = 0; i < priv->sources.size(); ++i) {
+        GRefPtr<GstBuffer> buffer;
+        GstFlowReturn ret = gst_buffer_pool_acquire_buffer(priv->pool.get(), &buffer.outPtr(), nullptr);
+        if (ret != GST_FLOW_OK) {
+            for (auto& buffer : channelBufferList)
+                unmapGstBuffer(buffer.get());
+
+            // FLUSHING and EOS are not errors.
+            if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
+                GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to allocate buffer for flow: %s", gst_flow_get_name(ret)));
+            gst_task_stop(src->priv->task.get());
+            return;
         }
 
-        GstFlowReturn ret = gst_pad_chain(pad, channelBuffer);
-        if (ret != GST_FLOW_OK)
-            GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s:%s flow: %s", GST_DEBUG_PAD_NAME(pad), gst_flow_get_name(ret)));
+        ASSERT(buffer);
+        GST_BUFFER_TIMESTAMP(buffer.get()) = timestamp;
+        GST_BUFFER_DURATION(buffer.get()) = duration;
+        mapGstBuffer(buffer.get(), GST_MAP_READWRITE);
+        priv->bus->setChannelMemory(i, reinterpret_cast<float*>(getGstBufferDataPointer(buffer.get())), priv->framesToPull);
+        channelBufferList.uncheckedAppend(WTFMove(buffer));
     }
 
-    priv->newStreamEventPending = false;
-
-    g_slist_free(channelBufferList);
+    // FIXME: Add support for local/live audio input.
+    priv->provider->render(nullptr, priv->bus, priv->framesToPull);
+
+    ASSERT(channelBufferList.size() == priv->sources.size());
+    bool failed = false;
+    for (unsigned i = 0; i < priv->sources.size(); ++i) {
+        // Unmap before passing on the buffer.
+        auto& buffer = channelBufferList[i];
+        unmapGstBuffer(buffer.get());
+
+        if (failed)
+            continue;
+
+        auto& appsrc = priv->sources[i];
+        // Leak the buffer ref, because gst_app_src_push_buffer steals it.
+        GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc.get()), buffer.leakRef());
+        if (ret != GST_FLOW_OK) {
+            // FLUSHING and EOS are not errors.
+            if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
+                GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc.get()), gst_flow_get_name(ret)));
+            gst_task_stop(src->priv->task.get());
+            failed = true;
+        }
+    }
 }
 
 static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, GstStateChange transition)
@@ -393,14 +378,10 @@ static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, Gs
     case GST_STATE_CHANGE_NULL_TO_READY:
         if (!src->priv->interleave) {
             gst_element_post_message(element, gst_missing_element_message_new(element, "interleave"));
-            GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (0), ("no interleave"));
-            return GST_STATE_CHANGE_FAILURE;
-        }
-        if (!src->priv->wavEncoder) {
-            gst_element_post_message(element, gst_missing_element_message_new(element, "wavenc"));
-            GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (0), ("no wavenc"));
+            GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (nullptr), ("no interleave"));
             return GST_STATE_CHANGE_FAILURE;
         }
+        src->priv->numberOfSamples = 0;
         break;
     default:
         break;
@@ -413,16 +394,29 @@ static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, Gs
     }
 
     switch (transition) {
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    case GST_STATE_CHANGE_READY_TO_PAUSED: {
         GST_DEBUG_OBJECT(src, "READY->PAUSED");
-        if (!gst_task_start(src->priv->task.get()))
+
+        src->priv->pool = gst_buffer_pool_new();
+        GstStructure* config = gst_buffer_pool_get_config(src->priv->pool.get());
+        gst_buffer_pool_config_set_params(config, nullptr, src->priv->bufferSize, 0, 0);
+        gst_buffer_pool_set_config(src->priv->pool.get(), config);
+        if (!gst_buffer_pool_set_active(src->priv->pool.get(), TRUE))
+            returnValue = GST_STATE_CHANGE_FAILURE;
+        else if (!gst_task_start(src->priv->task.get()))
             returnValue = GST_STATE_CHANGE_FAILURE;
         break;
+    }
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-        src->priv->newStreamEventPending = true;
         GST_DEBUG_OBJECT(src, "PAUSED->READY");
+
+#if GST_CHECK_VERSION(1, 4, 0)
+        gst_buffer_pool_set_flushing(src->priv->pool.get(), TRUE);
+#endif
         if (!gst_task_join(src->priv->task.get()))
             returnValue = GST_STATE_CHANGE_FAILURE;
+        gst_buffer_pool_set_active(src->priv->pool.get(), FALSE);
+        src->priv->pool = nullptr;
         break;
     default:
         break;