WebKitWebAudioSourceGStreamer.cpp [plain text]
#include "config.h"
#include "WebKitWebAudioSourceGStreamer.h"
#if ENABLE(WEB_AUDIO) && USE(GSTREAMER)
#include "AudioBus.h"
#include "AudioIOCallback.h"
#include "GRefPtrGStreamer.h"
#include "GStreamerUtilities.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;
typedef struct _WebKitWebAudioSrcClass WebKitWebAudioSrcClass;
typedef struct _WebKitWebAudioSourcePrivate WebKitWebAudioSourcePrivate;
struct _WebKitWebAudioSrc {
GstBin parent;
WebKitWebAudioSourcePrivate* priv;
};
struct _WebKitWebAudioSrcClass {
GstBinClass parentClass;
};
#define WEBKIT_WEB_AUDIO_SRC_GET_PRIVATE(obj) (G_TYPE_INSTANCE_GET_PRIVATE((obj), WEBKIT_TYPE_WEBAUDIO_SRC, WebKitWebAudioSourcePrivate))
struct _WebKitWebAudioSourcePrivate {
gfloat sampleRate;
AudioBus* bus;
AudioIOCallback* provider;
guint framesToPull;
guint bufferSize;
GRefPtr<GstElement> interleave;
GRefPtr<GstTask> task;
GRecMutex mutex;
GSList* sources; GstPad* sourcePad;
guint64 numberOfSamples;
GstBufferPool* pool;
};
enum {
PROP_RATE = 1,
PROP_BUS,
PROP_PROVIDER,
PROP_FRAMES
};
typedef struct {
GstBuffer* buffer;
GstMapInfo info;
} AudioSrcBuffer;
static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src",
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
static void webKitWebAudioSrcConstructed(GObject*);
static void webKitWebAudioSrcFinalize(GObject*);
static void webKitWebAudioSrcSetProperty(GObject*, guint propertyId, const GValue*, GParamSpec*);
static void webKitWebAudioSrcGetProperty(GObject*, guint propertyId, GValue*, GParamSpec*);
static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement*, GstStateChange);
static void webKitWebAudioSrcLoop(WebKitWebAudioSrc*);
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_NE(F32),
"layout", G_TYPE_STRING, "interleaved", nullptr);
}
static GstAudioChannelPosition webKitWebAudioGStreamerChannelPosition(int channelIndex)
{
GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_NONE;
switch (channelIndex) {
case AudioBus::ChannelLeft:
position = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case AudioBus::ChannelRight:
position = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case AudioBus::ChannelCenter:
position = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case AudioBus::ChannelLFE:
position = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
case AudioBus::ChannelSurroundLeft:
position = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case AudioBus::ChannelSurroundRight:
position = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
default:
break;
};
return position;
}
#define webkit_web_audio_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE(WebKitWebAudioSrc, webkit_web_audio_src, GST_TYPE_BIN, GST_DEBUG_CATEGORY_INIT(webkit_web_audio_src_debug, \
"webkitwebaudiosrc", \
0, \
"webaudiosrc element"));
static void webkit_web_audio_src_class_init(WebKitWebAudioSrcClass* webKitWebAudioSrcClass)
{
GObjectClass* objectClass = G_OBJECT_CLASS(webKitWebAudioSrcClass);
GstElementClass* elementClass = GST_ELEMENT_CLASS(webKitWebAudioSrcClass);
gst_element_class_add_pad_template(elementClass, gst_static_pad_template_get(&srcTemplate));
gst_element_class_set_metadata(elementClass, "WebKit WebAudio source element", "Source", "Handles WebAudio data from WebCore", "Philippe Normand <pnormand@igalia.com>");
objectClass->constructed = webKitWebAudioSrcConstructed;
objectClass->finalize = webKitWebAudioSrcFinalize;
elementClass->change_state = webKitWebAudioSrcChangeState;
objectClass->set_property = webKitWebAudioSrcSetProperty;
objectClass->get_property = webKitWebAudioSrcGetProperty;
GParamFlags flags = static_cast<GParamFlags>(G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE);
g_object_class_install_property(objectClass,
PROP_RATE,
g_param_spec_float("rate", "rate",
"Sample rate", G_MINDOUBLE, G_MAXDOUBLE,
44100.0, flags));
g_object_class_install_property(objectClass,
PROP_BUS,
g_param_spec_pointer("bus", "bus",
"Bus", flags));
g_object_class_install_property(objectClass,
PROP_PROVIDER,
g_param_spec_pointer("provider", "provider",
"Provider", flags));
g_object_class_install_property(objectClass,
PROP_FRAMES,
g_param_spec_uint("frames", "frames",
"Number of audio frames to pull at each iteration",
0, G_MAXUINT8, 128, flags));
g_type_class_add_private(webKitWebAudioSrcClass, sizeof(WebKitWebAudioSourcePrivate));
}
static void webkit_web_audio_src_init(WebKitWebAudioSrc* src)
{
WebKitWebAudioSourcePrivate* priv = G_TYPE_INSTANCE_GET_PRIVATE(src, WEBKIT_TYPE_WEB_AUDIO_SRC, WebKitWebAudioSourcePrivate);
src->priv = priv;
new (priv) WebKitWebAudioSourcePrivate();
priv->sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", 0);
gst_element_add_pad(GST_ELEMENT(src), priv->sourcePad);
priv->provider = 0;
priv->bus = 0;
g_rec_mutex_init(&priv->mutex);
priv->task = adoptGRef(gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0));
gst_task_set_lock(priv->task.get(), &priv->mutex);
}
static void webKitWebAudioSrcConstructed(GObject* object)
{
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(object);
WebKitWebAudioSourcePrivate* priv = src->priv;
ASSERT(priv->bus);
ASSERT(priv->provider);
ASSERT(priv->sampleRate);
priv->interleave = gst_element_factory_make("interleave", nullptr);
if (!priv->interleave) {
GST_ERROR_OBJECT(src, "Failed to create interleave");
return;
}
gst_bin_add(GST_BIN(src), priv->interleave.get());
for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
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(appsrc, "max-bytes", static_cast<guint64>(2 * priv->bufferSize), "block", TRUE,
"blocksize", priv->bufferSize,
"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);
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());
}
static void webKitWebAudioSrcFinalize(GObject* object)
{
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(object);
WebKitWebAudioSourcePrivate* priv = src->priv;
g_rec_mutex_clear(&priv->mutex);
g_slist_free_full(priv->sources, reinterpret_cast<GDestroyNotify>(gst_object_unref));
priv->~WebKitWebAudioSourcePrivate();
GST_CALL_PARENT(G_OBJECT_CLASS, finalize, ((GObject* )(src)));
}
static void webKitWebAudioSrcSetProperty(GObject* object, guint propertyId, const GValue* value, GParamSpec* pspec)
{
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(object);
WebKitWebAudioSourcePrivate* priv = src->priv;
switch (propertyId) {
case PROP_RATE:
priv->sampleRate = g_value_get_float(value);
break;
case PROP_BUS:
priv->bus = static_cast<AudioBus*>(g_value_get_pointer(value));
break;
case PROP_PROVIDER:
priv->provider = static_cast<AudioIOCallback*>(g_value_get_pointer(value));
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);
break;
}
}
static void webKitWebAudioSrcGetProperty(GObject* object, guint propertyId, GValue* value, GParamSpec* pspec)
{
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(object);
WebKitWebAudioSourcePrivate* priv = src->priv;
switch (propertyId) {
case PROP_RATE:
g_value_set_float(value, priv->sampleRate);
break;
case PROP_BUS:
g_value_set_pointer(value, priv->bus);
break;
case PROP_PROVIDER:
g_value_set_pointer(value, priv->provider);
break;
case PROP_FRAMES:
g_value_set_uint(value, priv->framesToPull);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propertyId, pspec);
break;
}
}
static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
{
WebKitWebAudioSourcePrivate* priv = src->priv;
ASSERT(priv->bus);
ASSERT(priv->provider);
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;
}
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;
GSList* channelBufferList = 0;
for (int i = g_slist_length(priv->sources) - 1; i >= 0; i--) {
AudioSrcBuffer* buffer = g_new(AudioSrcBuffer, 1);
GstBuffer* channelBuffer;
GstFlowReturn ret = gst_buffer_pool_acquire_buffer(priv->pool, &channelBuffer, nullptr);
if (ret != GST_FLOW_OK) {
g_free(buffer);
while (channelBufferList) {
buffer = static_cast<AudioSrcBuffer*>(channelBufferList->data);
gst_buffer_unmap(buffer->buffer, &buffer->info);
gst_buffer_unref(buffer->buffer);
g_free(buffer);
channelBufferList = g_slist_delete_link(channelBufferList, channelBufferList);
}
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;
}
ASSERT(channelBuffer);
buffer->buffer = channelBuffer;
GST_BUFFER_TIMESTAMP(channelBuffer) = timestamp;
GST_BUFFER_DURATION(channelBuffer) = duration;
gst_buffer_map(channelBuffer, &buffer->info, (GstMapFlags) GST_MAP_READWRITE);
priv->bus->setChannelMemory(i, reinterpret_cast<float*>(buffer->info.data), priv->framesToPull);
channelBufferList = g_slist_prepend(channelBufferList, buffer);
}
priv->provider->render(0, priv->bus, priv->framesToPull);
GSList* sourcesIt = priv->sources;
GSList* buffersIt = channelBufferList;
GstFlowReturn ret = GST_FLOW_OK;
for (int 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;
gst_buffer_unmap(channelBuffer, &buffer->info);
g_free(buffer);
if (priv->bus->channel(i)->isSilent())
GST_BUFFER_FLAG_SET(channelBuffer, GST_BUFFER_FLAG_GAP);
if (ret == GST_FLOW_OK) {
ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc), channelBuffer);
if (ret != GST_FLOW_OK) {
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), gst_flow_get_name(ret)));
gst_task_stop(src->priv->task.get());
}
} else
gst_buffer_unref(channelBuffer);
}
g_slist_free(channelBufferList);
}
static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, GstStateChange transition)
{
GstStateChangeReturn returnValue = GST_STATE_CHANGE_SUCCESS;
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(element);
switch (transition) {
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;
}
src->priv->numberOfSamples = 0;
break;
default:
break;
}
returnValue = GST_ELEMENT_CLASS(parent_class)->change_state(element, transition);
if (UNLIKELY(returnValue == GST_STATE_CHANGE_FAILURE)) {
GST_DEBUG_OBJECT(src, "State change failed");
return returnValue;
}
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED: {
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->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;
else if (!gst_task_start(src->priv->task.get()))
returnValue = GST_STATE_CHANGE_FAILURE;
break;
}
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT(src, "PAUSED->READY");
#if GST_CHECK_VERSION(1, 4, 0)
gst_buffer_pool_set_flushing(src->priv->pool, TRUE);
#endif
if (!gst_task_join(src->priv->task.get()))
returnValue = GST_STATE_CHANGE_FAILURE;
gst_buffer_pool_set_active(src->priv->pool, FALSE);
gst_object_unref(src->priv->pool);
src->priv->pool = nullptr;
break;
default:
break;
}
return returnValue;
}
#endif // ENABLE(WEB_AUDIO) && USE(GSTREAMER)