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/*
* 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
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "WebKitWebAudioSourceGStreamer.h"
#if ENABLE(WEB_AUDIO) && USE(GSTREAMER)
#include "AudioBus.h"
#include "AudioIOCallback.h"
#include "GStreamerCommon.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;
// List of appsrc. One appsrc for each planar audio channel.
Vector<GRefPtr<GstElement>> sources;
// src pad of the element, interleaved wav data is pushed to it.
GstPad* sourcePad;
guint64 numberOfSamples;
GRefPtr<GstBufferPool> pool;
bool enableGapBufferSupport;
};
enum {
PROP_RATE = 1,
PROP_BUS,
PROP_PROVIDER,
PROP_FRAMES
};
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", nullptr);
gst_element_add_pad(GST_ELEMENT(src), priv->sourcePad);
priv->provider = nullptr;
priv->bus = nullptr;
g_rec_mutex_init(&priv->mutex);
priv->task = adoptGRef(gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, nullptr));
// GAP buffer support is enabled only for GStreamer 1.12.5 because of a
// memory leak that was fixed in that version.
// https://bugzilla.gnome.org/show_bug.cgi?id=793067
priv->enableGapBufferSupport = webkitGstCheckVersion(1, 12, 5);
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 each channel of the bus create a new upstream branch for interleave, like:
// appsrc ! . which is plugged to a new interleave request sinkpad.
for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
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));
// 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);
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);
}
// 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());
}
static void webKitWebAudioSrcFinalize(GObject* object)
{
WebKitWebAudioSrc* src = WEBKIT_WEB_AUDIO_SRC(object);
WebKitWebAudioSourcePrivate* priv = src->priv;
g_rec_mutex_clear(&priv->mutex);
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 Optional<Vector<GRefPtr<GstBuffer>>> webKitWebAudioSrcAllocateBuffersAndRenderAudio(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 WTF::nullopt;
}
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());
Vector<RefPtr<GstMappedBuffer>> mappedBuffers;
mappedBuffers.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) {
// 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)));
return WTF::nullopt;
}
ASSERT(buffer);
GST_BUFFER_TIMESTAMP(buffer.get()) = timestamp;
GST_BUFFER_DURATION(buffer.get()) = duration;
auto mappedBuffer = GstMappedBuffer::create(buffer.get(), GST_MAP_READWRITE);
ASSERT(mappedBuffer);
mappedBuffers.uncheckedAppend(WTFMove(mappedBuffer));
priv->bus->setChannelMemory(i, reinterpret_cast<float*>(mappedBuffers[i]->data()), priv->framesToPull);
channelBufferList.uncheckedAppend(WTFMove(buffer));
}
// FIXME: Add support for local/live audio input.
priv->provider->render(nullptr, priv->bus, priv->framesToPull);
return makeOptional(channelBufferList);
}
static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
{
WebKitWebAudioSourcePrivate* priv = src->priv;
Optional<Vector<GRefPtr<GstBuffer>>> channelBufferList = webKitWebAudioSrcAllocateBuffersAndRenderAudio(src);
if (!channelBufferList) {
gst_task_stop(src->priv->task.get());
return;
}
ASSERT(channelBufferList->size() == priv->sources.size());
bool failed = false;
for (unsigned i = 0; i < priv->sources.size(); ++i) {
auto& buffer = channelBufferList.value()[i];
if (priv->enableGapBufferSupport && priv->bus->channel(i)->isSilent())
GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_GAP);
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)
{
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, (nullptr), ("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.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:
GST_DEBUG_OBJECT(src, "PAUSED->READY");
gst_buffer_pool_set_flushing(src->priv->pool.get(), TRUE);
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;
}
return returnValue;
}
#endif // ENABLE(WEB_AUDIO) && USE(GSTREAMER)