Changeset 177085 in webkit

Timestamp:

Dec 10, 2014 11:43:00 AM (9 years ago)

Author:

commit-queue@webkit.org

Message:

[GStreamer] Use appsrcs instead of unconnected queues
​https://bugs.webkit.org/show_bug.cgi?id=139490

Patch by Sebastian Dröge <​sebastian@centricular.com> on 2014-12-10
Reviewed by Philippe Normand.

• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:

(webkit_web_audio_src_init):
(webKitWebAudioSrcConstructed):
(webKitWebAudioSrcFinalize):
(webKitWebAudioSrcSetProperty):
(webKitWebAudioSrcLoop):
(webKitWebAudioSrcChangeState):
Previously we directly chained buffers into unconnected queues,
which confused some code inside GStreamer and caused some harmless
warnings. Now we use appsrcs instead, which also allows us to remove
quite some code.

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (modified) (14 diffs)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2014-12-10  Sebastian Dröge  <sebastian@centricular.com>
+
+        [GStreamer] Use appsrcs instead of unconnected queues
+        https://bugs.webkit.org/show_bug.cgi?id=139490
+
+        Reviewed by Philippe Normand.
+
+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+        (webkit_web_audio_src_init):
+        (webKitWebAudioSrcConstructed):
+        (webKitWebAudioSrcFinalize):
+        (webKitWebAudioSrcSetProperty):
+        (webKitWebAudioSrcLoop):
+        (webKitWebAudioSrcChangeState):
+        Previously we directly chained buffers into unconnected queues,
+        which confused some code inside GStreamer and caused some harmless
+        warnings. Now we use appsrcs instead, which also allows us to remove
+        quite some code.
+
 2014-12-10  Enrica Casucci  <enrica@apple.com>
 

trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp

@@ -28 +28 @@
 #include "GRefPtrGStreamer.h"
 #include "GStreamerUtilities.h"
+#include <gst/app/app.h>
 #include <gst/audio/audio.h>
 #include <gst/pbutils/pbutils.h>
@@ -53 +54 @@
     AudioIOCallback* provider;
     guint framesToPull;
+    guint bufferSize;
 
     GRefPtr<GstElement> interleave;
@@ -59 +61 @@
     GRecMutex mutex;
 
-    GSList* pads; // List of queue sink pads. One queue for each planar audio channel.
+    GSList* sources; // List of appsrc. One appsrc for each planar audio channel.
     GstPad* sourcePad; // src pad of the element, interleaved wav data is pushed to it.
 
-    bool newStreamEventPending;
-    GstSegment segment;
     guint64 numberOfSamples;
 
@@ -193 +193 @@
     priv->bus = 0;
 
-    priv->newStreamEventPending = true;
-    gst_segment_init(&priv->segment, GST_FORMAT_TIME);
-
     g_rec_mutex_init(&priv->mutex);
     priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0);
@@ -211 +208 @@
     ASSERT(priv->sampleRate);
 
-    priv->interleave = gst_element_factory_make("interleave", 0);
+    priv->interleave = gst_element_factory_make("interleave", nullptr);
 
     if (!priv->interleave) {
@@ -221 +218 @@
 
     // For each channel of the bus create a new upstream branch for interleave, like:
-    // queue ! capsfilter. 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);
-
+        GUniquePtr<gchar> appsrcName(g_strdup_printf("webaudioSrc%u", channelIndex));
+        GstElement* appsrc = gst_element_factory_make("appsrc", appsrcName.get());
         GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
 
@@ -233 +228 @@
         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);
-
-        gst_bin_add_many(GST_BIN(src), queue, capsfilter, NULL);
-        gst_element_link_pads_full(queue, "src", capsfilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
-        gst_element_link_pads_full(capsfilter, "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.
+
+        // Configure the appsrc for minimal latency.
+        g_object_set(appsrc, "max-bytes", 2 * priv->bufferSize, "block", TRUE,
+            "format", GST_FORMAT_TIME, "caps", caps.get(), nullptr);
+
+        priv->sources = g_slist_prepend(priv->sources, gst_object_ref(appsrc));
+
+        gst_bin_add(GST_BIN(src), appsrc);
+        gst_element_link_pads_full(appsrc, "src", priv->interleave.get(), "sink_%u", GST_PAD_LINK_CHECK_NOTHING);
+    }
+    priv->sources = g_slist_reverse(priv->sources);
+
+    // 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 +252 @@
     g_rec_mutex_clear(&priv->mutex);
 
-    g_slist_free_full(priv->pads, reinterpret_cast<GDestroyNotify>(gst_object_unref));
+    g_slist_free_full(priv->sources, reinterpret_cast<GDestroyNotify>(gst_object_unref));
 
     priv->~WebKitWebAudioSourcePrivate();
@@ -283 +275 @@
     case PROP_FRAMES:
         priv->framesToPull = g_value_get_uint(value);
+        priv->bufferSize = sizeof(float) * priv->framesToPull;
         break;
     default:
@@ -332 +325 @@
     GSList* channelBufferList = 0;
     register int i;
-    for (i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
+    for (i = g_slist_length(priv->sources) - 1; i >= 0; i--) {
         AudioSrcBuffer* buffer = g_new(AudioSrcBuffer, 1);
         GstBuffer* channelBuffer;
@@ -364 +357 @@
     priv->provider->render(0, priv->bus, priv->framesToPull);
 
-    GSList* padsIt = priv->pads;
+    GSList* sourcesIt = priv->sources;
     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);
+    for (i = 0; sourcesIt && buffersIt; sourcesIt = g_slist_next(sourcesIt), buffersIt = g_slist_next(buffersIt), ++i) {
+        GstElement* appsrc = static_cast<GstElement*>(sourcesIt->data);
         AudioSrcBuffer* buffer = static_cast<AudioSrcBuffer*>(buffersIt->data);
         GstBuffer* channelBuffer = buffer->buffer;
@@ -382 +369 @@
         g_free(buffer);
 
-        // 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));
-        }
-
-        GstFlowReturn ret = gst_pad_chain(pad, channelBuffer);
+        GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc), 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)));
+            GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc), gst_flow_get_name(ret)));
             gst_task_pause(src->priv->task.get());
         }
     }
-
-    priv->newStreamEventPending = false;
 
     g_slist_free(channelBufferList);
@@ -445 +408 @@
         src->priv->pool = gst_buffer_pool_new();
         GstStructure* config = gst_buffer_pool_get_config(src->priv->pool);
-        gst_buffer_pool_config_set_params(config, nullptr, src->priv->framesToPull * sizeof(float), 0, 0);
+        gst_buffer_pool_config_set_params(config, nullptr, src->priv->bufferSize, 0, 0);
         gst_buffer_pool_set_config(src->priv->pool, config);
         if (!gst_buffer_pool_set_active(src->priv->pool, TRUE))
@@ -454 +417 @@
     }
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-        src->priv->newStreamEventPending = true;
         GST_DEBUG_OBJECT(src, "PAUSED->READY");
         if (!gst_task_join(src->priv->task.get()))