Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (177084 => 177085)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-10 19:41:37 UTC (rev 177084)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-10 19:43:00 UTC (rev 177085)
@@ -27,6 +27,7 @@
#include "AudioIOCallback.h"
#include "GRefPtrGStreamer.h"
#include "GStreamerUtilities.h"
+#include <gst/app/app.h>
#include <gst/audio/audio.h>
#include <gst/pbutils/pbutils.h>
#include <wtf/gobject/GUniquePtr.h>
@@ -52,17 +53,16 @@
AudioBus* bus;
AudioIOCallback* provider;
guint framesToPull;
+ guint bufferSize;
GRefPtr<GstElement> interleave;
GRefPtr<GstTask> task;
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;
GstBufferPool* pool;
@@ -192,9 +192,6 @@
priv->provider = 0;
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);
@@ -210,7 +207,7 @@
ASSERT(priv->provider);
ASSERT(priv->sampleRate);
- priv->interleave = gst_element_factory_make("interleave", 0);
+ priv->interleave = gst_element_factory_make("interleave", nullptr);
if (!priv->interleave) {
GST_ERROR_OBJECT(src, "Failed to create interleave");
@@ -220,34 +217,29 @@
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. 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));
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);
+ // 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);
- GstPad* pad = gst_element_get_static_pad(queue, "sink");
- priv->pads = g_slist_prepend(priv->pads, pad);
+ priv->sources = g_slist_prepend(priv->sources, gst_object_ref(appsrc));
- 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);
-
+ 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->pads = g_slist_reverse(priv->pads);
+ priv->sources = g_slist_reverse(priv->sources);
- // wavenc's src pad is the only visible pad of our element.
+ // 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());
}
@@ -259,7 +251,7 @@
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();
GST_CALL_PARENT(G_OBJECT_CLASS, finalize, ((GObject* )(src)));
@@ -282,6 +274,7 @@
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);
@@ -331,7 +324,7 @@
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;
@@ -363,17 +356,11 @@
// FIXME: Add support for local/live audio input.
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;
@@ -381,37 +368,13 @@
gst_buffer_unmap(channelBuffer, &buffer->info);
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);
}
@@ -444,7 +407,7 @@
GST_DEBUG_OBJECT(src, "READY->PAUSED");
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))
returnValue = GST_STATE_CHANGE_FAILURE;
@@ -453,7 +416,6 @@
break;
}
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()))
returnValue = GST_STATE_CHANGE_FAILURE;
