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/*
* Copyright (C) 2020 Igalia S.L
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* aint with this library; see the file COPYING.LIB. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include "config.h"
#include "GStreamerAudioMixer.h"
#if USE(GSTREAMER)
#include "GStreamerCommon.h"
#include <wtf/NeverDestroyed.h>
namespace WebCore {
GST_DEBUG_CATEGORY_STATIC(webkit_media_gst_audio_mixer_debug);
#define GST_CAT_DEFAULT webkit_media_gst_audio_mixer_debug
bool GStreamerAudioMixer::isAvailable()
{
return isGStreamerPluginAvailable("inter") && isGStreamerPluginAvailable("audiomixer");
}
GStreamerAudioMixer& GStreamerAudioMixer::singleton()
{
static NeverDestroyed<GStreamerAudioMixer> sharedInstance;
return sharedInstance;
}
GStreamerAudioMixer::GStreamerAudioMixer()
{
GST_DEBUG_CATEGORY_INIT(webkit_media_gst_audio_mixer_debug, "webkitaudiomixer", 0, "WebKit GStreamer audio mixer");
m_pipeline = gst_element_factory_make("pipeline", "webkitaudiomixer");
registerActivePipeline(m_pipeline);
connectSimpleBusMessageCallback(m_pipeline.get());
m_mixer = makeGStreamerElement("audiomixer", nullptr);
auto* audioSink = createAutoAudioSink({ });
gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), m_mixer.get(), audioSink, nullptr);
gst_element_link(m_mixer.get(), audioSink);
gst_element_set_state(m_pipeline.get(), GST_STATE_READY);
}
void GStreamerAudioMixer::ensureState(GstStateChange stateChange)
{
GST_DEBUG_OBJECT(m_pipeline.get(), "Handling %s transition (%u mixer pads)", gst_state_change_get_name(stateChange), m_mixer->numsinkpads);
switch (stateChange) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_set_state(m_pipeline.get(), GST_STATE_PAUSED);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (m_mixer->numsinkpads == 1)
gst_element_set_state(m_pipeline.get(), GST_STATE_PAUSED);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (m_mixer->numsinkpads == 1)
gst_element_set_state(m_pipeline.get(), GST_STATE_READY);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (m_mixer->numsinkpads == 1) {
unregisterPipeline(m_pipeline);
gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
}
break;
default:
break;
}
}
GRefPtr<GstPad> GStreamerAudioMixer::registerProducer(GstElement* interaudioSink)
{
GstElement* src = makeGStreamerElement("interaudiosrc", nullptr);
g_object_set(src, "channel", GST_ELEMENT_NAME(interaudioSink), nullptr);
g_object_set(interaudioSink, "channel", GST_ELEMENT_NAME(interaudioSink), nullptr);
auto bin = gst_bin_new(nullptr);
auto audioResample = makeGStreamerElement("audioresample", nullptr);
auto audioConvert = makeGStreamerElement("audioconvert", nullptr);
gst_bin_add_many(GST_BIN_CAST(bin), audioResample, audioConvert, nullptr);
gst_element_link(audioConvert, audioResample);
if (auto pad = adoptGRef(gst_bin_find_unlinked_pad(GST_BIN_CAST(bin), GST_PAD_SRC)))
gst_element_add_pad(GST_ELEMENT_CAST(bin), gst_ghost_pad_new("src", pad.get()));
if (auto pad = adoptGRef(gst_bin_find_unlinked_pad(GST_BIN_CAST(bin), GST_PAD_SINK)))
gst_element_add_pad(GST_ELEMENT_CAST(bin), gst_ghost_pad_new("sink", pad.get()));
gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), src, bin, nullptr);
gst_element_link(src, bin);
bool shouldStart = !m_mixer->numsinkpads;
auto mixerPad = adoptGRef(gst_element_request_pad_simple(m_mixer.get(), "sink_%u"));
auto srcPad = adoptGRef(gst_element_get_static_pad(bin, "src"));
gst_pad_link(srcPad.get(), mixerPad.get());
if (shouldStart)
gst_element_set_state(m_pipeline.get(), GST_STATE_READY);
else
gst_bin_sync_children_states(GST_BIN_CAST(m_pipeline.get()));
GST_DEBUG_OBJECT(m_pipeline.get(), "Registered audio producer %" GST_PTR_FORMAT, mixerPad.get());
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_ALL, "audio-mixer-after-producer-registration");
return mixerPad;
}
void GStreamerAudioMixer::unregisterProducer(const GRefPtr<GstPad>& mixerPad)
{
GST_DEBUG_OBJECT(m_pipeline.get(), "Unregistering audio producer %" GST_PTR_FORMAT, mixerPad.get());
auto peer = adoptGRef(gst_pad_get_peer(mixerPad.get()));
auto bin = adoptGRef(gst_pad_get_parent_element(peer.get()));
auto sinkPad = adoptGRef(gst_element_get_static_pad(bin.get(), "sink"));
auto srcPad = adoptGRef(gst_pad_get_peer(sinkPad.get()));
auto interaudioSrc = adoptGRef(gst_pad_get_parent_element(srcPad.get()));
GST_LOG_OBJECT(m_pipeline.get(), "interaudiosrc: %" GST_PTR_FORMAT, interaudioSrc.get());
gst_element_set_locked_state(interaudioSrc.get(), true);
gst_element_set_state(interaudioSrc.get(), GST_STATE_NULL);
gst_element_set_state(bin.get(), GST_STATE_NULL);
gst_pad_unlink(peer.get(), mixerPad.get());
gst_element_unlink(interaudioSrc.get(), bin.get());
gst_element_release_request_pad(m_mixer.get(), mixerPad.get());
gst_bin_remove_many(GST_BIN_CAST(m_pipeline.get()), interaudioSrc.get(), bin.get(), nullptr);
if (!m_mixer->numsinkpads)
gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_ALL, "audio-mixer-after-producer-unregistration");
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif
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