diff options
Diffstat (limited to 'ext/resindvd/rsnaudiomunge.c')
| -rw-r--r-- | ext/resindvd/rsnaudiomunge.c | 378 |
1 files changed, 378 insertions, 0 deletions
diff --git a/ext/resindvd/rsnaudiomunge.c b/ext/resindvd/rsnaudiomunge.c new file mode 100644 index 00000000..f0429ee9 --- /dev/null +++ b/ext/resindvd/rsnaudiomunge.c @@ -0,0 +1,378 @@ +/* GStreamer + * Copyright (C) 2008 Jan Schmidt <thaytan@noraisin.net> + * + * 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 along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ +#ifdef HAVE_CONFIG_H +# include <config.h> +#endif + +#include <string.h> + +#include <gst/gst.h> + +#include "rsnaudiomunge.h" + +GST_DEBUG_CATEGORY_STATIC (rsn_audiomunge_debug); +#define GST_CAT_DEFAULT rsn_audiomunge_debug + +#define AUDIO_FILL_THRESHOLD (GST_SECOND/5) + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_SILENT +}; + +/* the capabilities of the inputs and outputs. + * + * describe the real formats here. + */ +static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("ANY") + ); + +static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("ANY") + ); + +GST_BOILERPLATE (RsnAudioMunge, rsn_audiomunge, GstElement, GST_TYPE_ELEMENT); + +static void rsn_audiomunge_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void rsn_audiomunge_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static gboolean rsn_audiomunge_set_caps (GstPad * pad, GstCaps * caps); +static GstFlowReturn rsn_audiomunge_chain (GstPad * pad, GstBuffer * buf); +static gboolean rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event); + +static GstStateChangeReturn +rsn_audiomunge_change_state (GstElement * element, GstStateChange transition); + +static void +rsn_audiomunge_base_init (gpointer gclass) +{ + static GstElementDetails element_details = { + "RsnAudioMunge", + "Audio/Filter", + "Resin DVD audio stream regulator", + "Jan Schmidt <thaytan@noraisin.net>" + }; + GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); + + GST_DEBUG_CATEGORY_INIT (rsn_audiomunge_debug, "rsn_audiomunge", + 0, "Resin audio stream regulator"); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_template)); + + gst_element_class_set_details (element_class, &element_details); +} + +static void +rsn_audiomunge_class_init (RsnAudioMungeClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + gobject_class->set_property = rsn_audiomunge_set_property; + gobject_class->get_property = rsn_audiomunge_get_property; + + gstelement_class->change_state = rsn_audiomunge_change_state; +} + +static void +rsn_audiomunge_init (RsnAudioMunge * munge, RsnAudioMungeClass * gclass) +{ + munge->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink"); + gst_pad_set_setcaps_function (munge->sinkpad, + GST_DEBUG_FUNCPTR (rsn_audiomunge_set_caps)); + gst_pad_set_getcaps_function (munge->sinkpad, + GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps)); + gst_pad_set_chain_function (munge->sinkpad, + GST_DEBUG_FUNCPTR (rsn_audiomunge_chain)); + gst_pad_set_event_function (munge->sinkpad, + GST_DEBUG_FUNCPTR (rsn_audiomunge_sink_event)); + gst_element_add_pad (GST_ELEMENT (munge), munge->sinkpad); + + munge->srcpad = gst_pad_new_from_static_template (&src_template, "src"); + gst_pad_set_getcaps_function (munge->srcpad, + GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps)); + gst_element_add_pad (GST_ELEMENT (munge), munge->srcpad); +} + +static void +rsn_audiomunge_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + //RsnAudioMunge *munge = RSN_AUDIOMUNGE (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +rsn_audiomunge_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + //RsnAudioMunge *munge = RSN_AUDIOMUNGE (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static gboolean +rsn_audiomunge_set_caps (GstPad * pad, GstCaps * caps) +{ + RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad)); + GstPad *otherpad; + gboolean ret; + + g_return_val_if_fail (munge != NULL, FALSE); + + otherpad = (pad == munge->srcpad) ? munge->sinkpad : munge->srcpad; + gst_object_unref (munge); + + ret = gst_pad_set_caps (otherpad, caps); + return ret; +} + +static void +rsn_audiomunge_reset (RsnAudioMunge * munge) +{ + munge->have_audio = FALSE; + munge->in_still = FALSE; + gst_segment_init (&munge->sink_segment, GST_FORMAT_TIME); +} + +static GstFlowReturn +rsn_audiomunge_chain (GstPad * pad, GstBuffer * buf) +{ + RsnAudioMunge *munge = RSN_AUDIOMUNGE (GST_OBJECT_PARENT (pad)); + + if (!munge->have_audio) { + g_print ("First audio after flush has TS %" GST_TIME_FORMAT "\n", + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); + } + + munge->have_audio = TRUE; + + /* just push out the incoming buffer without touching it */ + return gst_pad_push (munge->srcpad, buf); +} + +/* Create and send a silence buffer downstream */ +static GstFlowReturn +rsn_audiomunge_make_audio (RsnAudioMunge * munge, + GstClockTime start, GstClockTime fill_time) +{ + GstFlowReturn ret; + GstBuffer *audio_buf; + GstCaps *caps; + guint buf_size; + + /* Just generate a 48khz stereo buffer for now */ +#if 0 + caps = + gst_caps_from_string + ("audio/x-raw-int,rate=48000,channels=2,width=16,depth=16,signed=(boolean)true,endianness=1234"); + buf_size = 4 * (48000 * fill_time / GST_SECOND); +#else + caps = gst_caps_from_string ("audio/x-raw-float, endianness=(int)1234," + "width=(int)32, channels=(int)2, rate=(int)48000"); + buf_size = 2 * 4 * (48000 * fill_time / GST_SECOND); +#endif + + audio_buf = gst_buffer_new_and_alloc (buf_size); + + gst_buffer_set_caps (audio_buf, caps); + gst_caps_unref (caps); + + GST_BUFFER_TIMESTAMP (audio_buf) = start; + GST_BUFFER_DURATION (audio_buf) = fill_time; + GST_BUFFER_FLAG_SET (audio_buf, GST_BUFFER_FLAG_DISCONT); + + memset (GST_BUFFER_DATA (audio_buf), 0, buf_size); + + g_print ("Sending %u bytes (%" GST_TIME_FORMAT ") of audio data " + "with TS %" GST_TIME_FORMAT "\n", + buf_size, GST_TIME_ARGS (fill_time), GST_TIME_ARGS (start)); + + ret = gst_pad_push (munge->srcpad, audio_buf); + + return ret; +} + +static void +rsn_audiomunge_handle_dvd_event (RsnAudioMunge * munge, GstEvent * event) +{ + const GstStructure *s; + const gchar *event_type; + + s = gst_event_get_structure (event); + event_type = gst_structure_get_string (s, "event"); + if (event_type == NULL) + return; + + if (strcmp (event_type, "dvd-still") == 0) { + gboolean in_still; + + if (!gst_structure_get_boolean (s, "still-state", &in_still)) + return; + + /* Remember the still-frame state, so we can generate a pre-roll buffer + * when a new-segment arrives */ + munge->in_still = in_still; + + g_print ("**** AUDIO MUNGE: still-state now %d\n", munge->in_still); + } +} + +static gboolean +rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event) +{ + gboolean ret = FALSE; + RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + rsn_audiomunge_reset (munge); + + g_print ("*********** AUDIO MUNGE: FLUSH\n"); + ret = gst_pad_push_event (munge->srcpad, event); + break; + case GST_EVENT_NEWSEGMENT: + { + GstSegment *segment; + gboolean update; + GstFormat format; + gdouble rate, arate; + gint64 start, stop, time; + + gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, + &start, &stop, &time); + + /* we need TIME format */ + if (format != GST_FORMAT_TIME) + goto newseg_wrong_format; + + /* now configure the values */ + segment = &munge->sink_segment; + + gst_segment_set_newsegment_full (segment, update, + rate, arate, format, start, stop, time); + + if (munge->have_audio) { + g_print ("*********** AUDIO MUNGE NEWSEG: start %" GST_TIME_FORMAT + " stop %" GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT + "\n", GST_TIME_ARGS (start), GST_TIME_ARGS (stop), + GST_TIME_ARGS (segment->accum)); + + ret = gst_pad_push_event (munge->srcpad, event); + break; + } + + /* + * FIXME: + * If the accum >= threshold or we're in a still frame and there's been + * no audio received, then we need to generate some audio data. + * If caused by a segment start update (time advancing in a gap) adjust + * the new-segment and send the buffer. + * + * Otherwise, send the buffer before the newsegment, so that it appears + * in the closing segment. + */ + if (segment->accum >= AUDIO_FILL_THRESHOLD || munge->in_still) { + g_print ("*********** Send audio mebbe: accum = %" GST_TIME_FORMAT + " still-state=%d\n", GST_TIME_ARGS (segment->accum), + munge->in_still); + /* Just generate a 100ms silence buffer for now. FIXME: Fill the gap */ + if (rsn_audiomunge_make_audio (munge, segment->start, + GST_SECOND / 10) == GST_FLOW_OK) + munge->have_audio = TRUE; + } else { + g_print ("*********** below thresh: accum = %" GST_TIME_FORMAT + "\n", GST_TIME_ARGS (segment->accum)); + } + + g_print ("*********** AUDIO MUNGE NEWSEG: start %" GST_TIME_FORMAT + " stop %" GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT + "\n", GST_TIME_ARGS (start), GST_TIME_ARGS (stop), + GST_TIME_ARGS (segment->accum)); + + ret = gst_pad_push_event (munge->srcpad, event); + break; + } + case GST_EVENT_CUSTOM_DOWNSTREAM: + { + const GstStructure *s = gst_event_get_structure (event); + if (s && gst_structure_has_name (s, "application/x-gst-dvd")) + rsn_audiomunge_handle_dvd_event (munge, event); + + ret = gst_pad_push_event (munge->srcpad, event); + break; + } + default: + ret = gst_pad_push_event (munge->srcpad, event); + break; + } + + return ret; + +newseg_wrong_format: + + GST_DEBUG_OBJECT (munge, "received non TIME newsegment"); + gst_event_unref (event); + gst_object_unref (munge); + return FALSE; +} + +static GstStateChangeReturn +rsn_audiomunge_change_state (GstElement * element, GstStateChange transition) +{ + RsnAudioMunge *munge = RSN_AUDIOMUNGE (element); + GstStateChangeReturn ret; + + if (transition == GST_STATE_CHANGE_READY_TO_PAUSED) + rsn_audiomunge_reset (munge); + + ret = parent_class->change_state (element, transition); + + return ret; +} |