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gstreamer/gst/adder/gstadder.c
Mathieu Duponchelle 6954d2167c adder: rework the logic to check if eos has to be sent. 
Checking the size available was incorrect, and the infos
for per-pad EOS are available.

Same logic as audiomixer.

fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727025
2014-04-08 13:48:27 +02:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
*
* adder.c: Adder element, N in, one out, samples are added
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-adder
*
* The adder allows to mix several streams into one by adding the data.
* Mixed data is clamped to the min/max values of the data format.
*
* The adder currently mixes all data received on the sinkpads as soon as
* possible without trying to synchronize the streams.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
* ]| This pipeline produces two sine waves mixed together.
* </refsect2>
*
* Last reviewed on 2006-05-09 (0.10.7)
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstadder.h"
#include <gst/audio/audio.h>
#include <string.h> /* strcmp */
#include "gstadderorc.h"
#define GST_CAT_DEFAULT gst_adder_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define DEFAULT_PAD_VOLUME (1.0)
#define DEFAULT_PAD_MUTE (FALSE)
/* some defines for audio processing */
/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
* we map 1.0 to VOLUME_UNITY_INT*
*/
#define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */
#define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */
#define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */
#define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
#define VOLUME_UNITY_INT32_BIT_SHIFT 27
enum
{
PROP_PAD_0,
PROP_PAD_VOLUME,
PROP_PAD_MUTE
};
G_DEFINE_TYPE (GstAdderPad, gst_adder_pad, GST_TYPE_PAD);
static void
gst_adder_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAdderPad *pad = GST_ADDER_PAD (object);
switch (prop_id) {
case PROP_PAD_VOLUME:
g_value_set_double (value, pad->volume);
break;
case PROP_PAD_MUTE:
g_value_set_boolean (value, pad->mute);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_adder_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAdderPad *pad = GST_ADDER_PAD (object);
switch (prop_id) {
case PROP_PAD_VOLUME:
GST_OBJECT_LOCK (pad);
pad->volume = g_value_get_double (value);
pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8;
pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16;
pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32;
GST_OBJECT_UNLOCK (pad);
break;
case PROP_PAD_MUTE:
GST_OBJECT_LOCK (pad);
pad->mute = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_adder_pad_class_init (GstAdderPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_adder_pad_set_property;
gobject_class->get_property = gst_adder_pad_get_property;
g_object_class_install_property (gobject_class, PROP_PAD_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume of this pad",
0.0, 10.0, DEFAULT_PAD_VOLUME,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PAD_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute this pad",
DEFAULT_PAD_MUTE,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_adder_pad_init (GstAdderPad * pad)
{
pad->volume = DEFAULT_PAD_VOLUME;
pad->mute = DEFAULT_PAD_MUTE;
}
enum
{
PROP_0,
PROP_FILTER_CAPS
};
/* elementfactory information */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
", layout = (string) { interleaved, non-interleaved }"
#else
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
", layout = (string) { interleaved, non-interleaved }"
#endif
static GstStaticPadTemplate gst_adder_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS)
);
static GstStaticPadTemplate gst_adder_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS (CAPS)
);
static void gst_adder_child_proxy_init (gpointer g_iface, gpointer iface_data);
#define gst_adder_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAdder, gst_adder, GST_TYPE_ELEMENT,
G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_adder_child_proxy_init));
static void gst_adder_dispose (GObject * object);
static void gst_adder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_adder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_adder_setcaps (GstAdder * adder, GstPad * pad,
GstCaps * caps);
static gboolean gst_adder_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_adder_sink_query (GstCollectPads * pads,
GstCollectData * pad, GstQuery * query, gpointer user_data);
static gboolean gst_adder_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_adder_sink_event (GstCollectPads * pads,
GstCollectData * pad, GstEvent * event, gpointer user_data);
static GstPad *gst_adder_request_new_pad (GstElement * element,
GstPadTemplate * temp, const gchar * unused, const GstCaps * caps);
static void gst_adder_release_pad (GstElement * element, GstPad * pad);
static GstStateChangeReturn gst_adder_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_adder_do_clip (GstCollectPads * pads,
GstCollectData * data, GstBuffer * buffer, GstBuffer ** out,
gpointer user_data);
static GstFlowReturn gst_adder_collected (GstCollectPads * pads,
gpointer user_data);
/* we can only accept caps that we and downstream can handle.
* if we have filtercaps set, use those to constrain the target caps.
*/
static GstCaps *
gst_adder_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstAdder *adder;
GstCaps *result, *peercaps, *current_caps, *filter_caps;
GstStructure *s;
gint i, n;
adder = GST_ADDER (GST_PAD_PARENT (pad));
GST_OBJECT_LOCK (adder);
/* take filter */
if ((filter_caps = adder->filter_caps)) {
if (filter)
filter_caps =
gst_caps_intersect_full (filter, filter_caps,
GST_CAPS_INTERSECT_FIRST);
else
gst_caps_ref (filter_caps);
} else {
filter_caps = filter ? gst_caps_ref (filter) : NULL;
}
GST_OBJECT_UNLOCK (adder);
if (filter_caps && gst_caps_is_empty (filter_caps)) {
GST_WARNING_OBJECT (pad, "Empty filter caps");
return filter_caps;
}
/* get the downstream possible caps */
peercaps = gst_pad_peer_query_caps (adder->srcpad, filter_caps);
/* get the allowed caps on this sinkpad */
GST_OBJECT_LOCK (adder);
current_caps =
adder->current_caps ? gst_caps_ref (adder->current_caps) : NULL;
if (current_caps == NULL) {
current_caps = gst_pad_get_pad_template_caps (pad);
if (!current_caps)
current_caps = gst_caps_new_any ();
}
GST_OBJECT_UNLOCK (adder);
if (peercaps) {
/* if the peer has caps, intersect */
GST_DEBUG_OBJECT (adder, "intersecting peer and our caps");
result =
gst_caps_intersect_full (peercaps, current_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
gst_caps_unref (current_caps);
} else {
/* the peer has no caps (or there is no peer), just use the allowed caps
* of this sinkpad. */
/* restrict with filter-caps if any */
if (filter_caps) {
GST_DEBUG_OBJECT (adder, "no peer caps, using filtered caps");
result =
gst_caps_intersect_full (filter_caps, current_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (current_caps);
} else {
GST_DEBUG_OBJECT (adder, "no peer caps, using our caps");
result = current_caps;
}
}
result = gst_caps_make_writable (result);
n = gst_caps_get_size (result);
for (i = 0; i < n; i++) {
GstStructure *sref;
s = gst_caps_get_structure (result, i);
sref = gst_structure_copy (s);
gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL);
if (gst_structure_is_subset (s, sref)) {
/* This field is irrelevant when in mono or stereo */
gst_structure_remove_field (s, "channel-mask");
}
gst_structure_free (sref);
}
if (filter_caps)
gst_caps_unref (filter_caps);
GST_LOG_OBJECT (adder, "getting caps on pad %p,%s to %" GST_PTR_FORMAT, pad,
GST_PAD_NAME (pad), result);
return result;
}
static gboolean
gst_adder_sink_query (GstCollectPads * pads, GstCollectData * pad,
GstQuery * query, gpointer user_data)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_adder_sink_getcaps (pad->pad, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
res = gst_collect_pads_query_default (pads, pad, query, FALSE);
break;
}
return res;
}
/* the first caps we receive on any of the sinkpads will define the caps for all
* the other sinkpads because we can only mix streams with the same caps.
*/
static gboolean
gst_adder_setcaps (GstAdder * adder, GstPad * pad, GstCaps * orig_caps)
{
GstCaps *caps;
GstAudioInfo info;
GstStructure *s;
gint channels;
caps = gst_caps_copy (orig_caps);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels))
if (channels <= 2)
gst_structure_remove_field (s, "channel-mask");
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_format;
GST_OBJECT_LOCK (adder);
/* don't allow reconfiguration for now; there's still a race between the
* different upstream threads doing query_caps + accept_caps + sending
* (possibly different) CAPS events, but there's not much we can do about
* that, upstream needs to deal with it. */
if (adder->current_caps != NULL) {
if (gst_audio_info_is_equal (&info, &adder->info)) {
GST_OBJECT_UNLOCK (adder);
gst_caps_unref (caps);
return TRUE;
} else {
GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
"current caps are %" GST_PTR_FORMAT, caps, adder->current_caps);
GST_OBJECT_UNLOCK (adder);
gst_pad_push_event (pad, gst_event_new_reconfigure ());
gst_caps_unref (caps);
return FALSE;
}
}
GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps);
adder->current_caps = gst_caps_ref (caps);
memcpy (&adder->info, &info, sizeof (info));
GST_OBJECT_UNLOCK (adder);
/* send caps event later, after stream-start event */
GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
return TRUE;
/* ERRORS */
invalid_format:
{
gst_caps_unref (caps);
GST_WARNING_OBJECT (adder, "invalid format set as caps");
return FALSE;
}
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_adder_query_duration (GstAdder * adder, GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (adder, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_adder_query_latency (GstAdder * adder, GstQuery * query)
{
GstClockTime min, max;
gboolean live;
gboolean res;
GstIterator *it;
gboolean done;
GValue item = { 0, };
res = TRUE;
done = FALSE;
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
/* Take maximum of all latency values */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
GstQuery *peerquery;
GstClockTime min_cur, max_cur;
gboolean live_cur;
peerquery = gst_query_new_latency ();
/* Ask peer for latency */
res &= gst_pad_peer_query (pad, peerquery);
/* take max from all valid return values */
if (res) {
gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
if (min_cur > min)
min = min_cur;
if (max_cur != GST_CLOCK_TIME_NONE &&
((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
(max == GST_CLOCK_TIME_NONE)))
max = max_cur;
live = live || live_cur;
}
gst_query_unref (peerquery);
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* store the results */
GST_DEBUG_OBJECT (adder, "Calculated total latency: live %s, min %"
GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
(live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
return res;
}
static gboolean
gst_adder_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAdder *adder = GST_ADDER (parent);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
/* FIXME, bring to stream time, might be tricky */
gst_query_set_position (query, format, adder->segment.position);
res = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, adder->offset);
res = TRUE;
break;
default:
break;
}
break;
}
case GST_QUERY_DURATION:
res = gst_adder_query_duration (adder, query);
break;
case GST_QUERY_LATENCY:
res = gst_adder_query_latency (adder, query);
break;
default:
/* FIXME, needs a custom query handler because we have multiple
* sinkpads */
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
/* event handling */
typedef struct
{
GstEvent *event;
gboolean flush;
} EventData;
static gboolean
forward_event_func (const GValue * val, GValue * ret, EventData * data)
{
GstPad *pad = g_value_get_object (val);
GstEvent *event = data->event;
GstPad *peer;
gst_event_ref (event);
GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
peer = gst_pad_get_peer (pad);
/* collect pad might have been set flushing,
* so bypass core checking that and send directly to peer */
if (!peer || !gst_pad_send_event (peer, event)) {
if (!peer)
gst_event_unref (event);
GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
event, GST_EVENT_TYPE_NAME (event));
/* quick hack to unflush the pads, ideally we need a way to just unflush
* this single collect pad */
if (data->flush)
gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE));
} else {
g_value_set_boolean (ret, TRUE);
GST_LOG_OBJECT (pad, "Sent event %p (%s).",
event, GST_EVENT_TYPE_NAME (event));
}
if (peer)
gst_object_unref (peer);
/* continue on other pads, even if one failed */
return TRUE;
}
/* forwards the event to all sinkpads, takes ownership of the
* event
*
* Returns: TRUE if the event could be forwarded on all
* sinkpads.
*/
static gboolean
forward_event (GstAdder * adder, GstEvent * event, gboolean flush)
{
gboolean ret;
GstIterator *it;
GstIteratorResult ires;
GValue vret = { 0 };
EventData data;
GST_LOG_OBJECT (adder, "Forwarding event %p (%s)", event,
GST_EVENT_TYPE_NAME (event));
data.event = event;
data.flush = flush;
g_value_init (&vret, G_TYPE_BOOLEAN);
g_value_set_boolean (&vret, FALSE);
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (TRUE) {
ires =
gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func,
&vret, &data);
switch (ires) {
case GST_ITERATOR_RESYNC:
GST_WARNING ("resync");
gst_iterator_resync (it);
g_value_set_boolean (&vret, TRUE);
break;
case GST_ITERATOR_OK:
case GST_ITERATOR_DONE:
ret = g_value_get_boolean (&vret);
goto done;
default:
ret = FALSE;
goto done;
}
}
done:
gst_iterator_free (it);
GST_LOG_OBJECT (adder, "Forwarded event %p (%s), ret=%d", event,
GST_EVENT_TYPE_NAME (event), ret);
gst_event_unref (event);
return ret;
}
static gboolean
gst_adder_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAdder *adder;
gboolean result;
adder = GST_ADDER (parent);
GST_DEBUG_OBJECT (pad, "Got %s event on src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
gdouble rate;
GstSeekType start_type, stop_type;
gint64 start, stop;
GstFormat seek_format, dest_format;
gboolean flush;
/* parse the seek parameters */
gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
&start, &stop_type, &stop);
if ((start_type != GST_SEEK_TYPE_NONE)
&& (start_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (adder,
"seeking failed, unhandled seek type for start: %d", start_type);
goto done;
}
if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (adder,
"seeking failed, unhandled seek type for end: %d", stop_type);
goto done;
}
dest_format = adder->segment.format;
if (seek_format != dest_format) {
result = FALSE;
GST_DEBUG_OBJECT (adder,
"seeking failed, unhandled seek format: %d", seek_format);
goto done;
}
flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH;
/* check if we are flushing */
if (flush) {
/* flushing seek, start flush downstream, the flush will be done
* when all pads received a FLUSH_STOP.
* Make sure we accept nothing anymore and return WRONG_STATE.
* We send a flush-start before, to ensure no streaming is done
* as we need to take the stream lock.
*/
gst_pad_push_event (adder->srcpad, gst_event_new_flush_start ());
gst_collect_pads_set_flushing (adder->collect, TRUE);
/* We can't send FLUSH_STOP here since upstream could start pushing data
* after we unlock adder->collect.
* We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
* forwarding the seek upstream or from gst_adder_collected,
* whichever happens first.
*/
GST_COLLECT_PADS_STREAM_LOCK (adder->collect);
adder->flush_stop_pending = TRUE;
GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect);
GST_DEBUG_OBJECT (adder, "mark pending flush stop event");
}
GST_DEBUG_OBJECT (adder, "handling seek event: %" GST_PTR_FORMAT, event);
/* now wait for the collected to be finished and mark a new
* segment. After we have the lock, no collect function is running and no
* new collect function will be called for as long as we're flushing. */
GST_COLLECT_PADS_STREAM_LOCK (adder->collect);
/* clip position and update our segment */
if (adder->segment.stop != -1) {
adder->segment.position = adder->segment.stop;
}
gst_segment_do_seek (&adder->segment, rate, seek_format, flags,
start_type, start, stop_type, stop, NULL);
if (flush) {
/* Yes, we need to call _set_flushing again *WHEN* the streaming threads
* have stopped so that the cookie gets properly updated. */
gst_collect_pads_set_flushing (adder->collect, TRUE);
}
GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect);
GST_DEBUG_OBJECT (adder, "forwarding seek event: %" GST_PTR_FORMAT,
event);
GST_DEBUG_OBJECT (adder, "updated segment: %" GST_SEGMENT_FORMAT,
&adder->segment);
/* we're forwarding seek to all upstream peers and wait for one to reply
* with a newsegment-event before we send a newsegment-event downstream */
g_atomic_int_set (&adder->new_segment_pending, TRUE);
result = forward_event (adder, event, flush);
if (!result) {
/* seek failed. maybe source is a live source. */
GST_DEBUG_OBJECT (adder, "seeking failed");
}
if (g_atomic_int_compare_and_exchange (&adder->flush_stop_pending,
TRUE, FALSE)) {
GST_DEBUG_OBJECT (adder, "pending flush stop");
if (!gst_pad_push_event (adder->srcpad,
gst_event_new_flush_stop (TRUE))) {
GST_WARNING_OBJECT (adder, "Sending flush stop event failed");
}
}
break;
}
case GST_EVENT_QOS:
/* QoS might be tricky */
result = FALSE;
gst_event_unref (event);
break;
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
result = FALSE;
gst_event_unref (event);
break;
default:
/* just forward the rest for now */
GST_DEBUG_OBJECT (adder, "forward unhandled event: %s",
GST_EVENT_TYPE_NAME (event));
result = forward_event (adder, event, FALSE);
break;
}
done:
return result;
}
static gboolean
gst_adder_sink_event (GstCollectPads * pads, GstCollectData * pad,
GstEvent * event, gpointer user_data)
{
GstAdder *adder = GST_ADDER (user_data);
gboolean res = TRUE, discard = FALSE;
GST_DEBUG_OBJECT (pad->pad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_adder_setcaps (adder, pad->pad, caps);
gst_event_unref (event);
event = NULL;
break;
}
case GST_EVENT_FLUSH_START:
/* ensure that we will send a flush stop */
res = gst_collect_pads_event_default (pads, pad, event, discard);
event = NULL;
GST_COLLECT_PADS_STREAM_LOCK (adder->collect);
adder->flush_stop_pending = TRUE;
GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect);
break;
case GST_EVENT_FLUSH_STOP:
/* we received a flush-stop. We will only forward it when
* flush_stop_pending is set, and we will unset it then.
*/
g_atomic_int_set (&adder->new_segment_pending, TRUE);
GST_COLLECT_PADS_STREAM_LOCK (adder->collect);
if (adder->flush_stop_pending) {
GST_DEBUG_OBJECT (pad->pad, "forwarding flush stop");
res = gst_collect_pads_event_default (pads, pad, event, discard);
adder->flush_stop_pending = FALSE;
event = NULL;
} else {
discard = TRUE;
GST_DEBUG_OBJECT (pad->pad, "eating flush stop");
}
GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect);
/* Clear pending tags */
if (adder->pending_events) {
g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
break;
case GST_EVENT_TAG:
/* collect tags here so we can push them out when we collect data */
adder->pending_events = g_list_append (adder->pending_events, event);
event = NULL;
break;
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->rate != adder->segment.rate) {
GST_ERROR_OBJECT (pad->pad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, adder->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
}
discard = TRUE;
break;
}
default:
break;
}
if (G_LIKELY (event))
return gst_collect_pads_event_default (pads, pad, event, discard);
else
return res;
}
static void
gst_adder_class_init (GstAdderClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_adder_set_property;
gobject_class->get_property = gst_adder_get_property;
gobject_class->dispose = gst_adder_dispose;
g_object_class_install_property (gobject_class, PROP_FILTER_CAPS,
g_param_spec_boxed ("caps", "Target caps",
"Set target format for mixing (NULL means ANY). "
"Setting this property takes a reference to the supplied GstCaps "
"object.", GST_TYPE_CAPS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_sink_template));
gst_element_class_set_static_metadata (gstelement_class, "Adder",
"Generic/Audio",
"Add N audio channels together",
"Thomas Vander Stichele <thomas at apestaart dot org>");
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_adder_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_adder_release_pad);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_adder_change_state);
}
static void
gst_adder_init (GstAdder * adder)
{
GstPadTemplate *template;
template = gst_static_pad_template_get (&gst_adder_src_template);
adder->srcpad = gst_pad_new_from_template (template, "src");
gst_object_unref (template);
gst_pad_set_query_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_src_query));
gst_pad_set_event_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_src_event));
GST_PAD_SET_PROXY_CAPS (adder->srcpad);
gst_element_add_pad (GST_ELEMENT (adder), adder->srcpad);
adder->current_caps = NULL;
gst_audio_info_init (&adder->info);
adder->padcount = 0;
adder->filter_caps = NULL;
/* keep track of the sinkpads requested */
adder->collect = gst_collect_pads_new ();
gst_collect_pads_set_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_collected), adder);
gst_collect_pads_set_clip_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_do_clip), adder);
gst_collect_pads_set_event_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_sink_event), adder);
gst_collect_pads_set_query_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_sink_query), adder);
}
static void
gst_adder_dispose (GObject * object)
{
GstAdder *adder = GST_ADDER (object);
if (adder->collect) {
gst_object_unref (adder->collect);
adder->collect = NULL;
}
gst_caps_replace (&adder->filter_caps, NULL);
gst_caps_replace (&adder->current_caps, NULL);
if (adder->pending_events) {
g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_adder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAdder *adder = GST_ADDER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:{
GstCaps *new_caps = NULL;
GstCaps *old_caps;
const GstCaps *new_caps_val = gst_value_get_caps (value);
if (new_caps_val != NULL) {
new_caps = (GstCaps *) new_caps_val;
gst_caps_ref (new_caps);
}
GST_OBJECT_LOCK (adder);
old_caps = adder->filter_caps;
adder->filter_caps = new_caps;
GST_OBJECT_UNLOCK (adder);
if (old_caps)
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (adder, "set new caps %" GST_PTR_FORMAT, new_caps);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_adder_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAdder *adder = GST_ADDER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:
GST_OBJECT_LOCK (adder);
gst_value_set_caps (value, adder->filter_caps);
GST_OBJECT_UNLOCK (adder);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstPad *
gst_adder_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * unused, const GstCaps * caps)
{
gchar *name;
GstAdder *adder;
GstPad *newpad;
gint padcount;
if (templ->direction != GST_PAD_SINK)
goto not_sink;
adder = GST_ADDER (element);
/* increment pad counter */
padcount = g_atomic_int_add (&adder->padcount, 1);
name = g_strdup_printf ("sink_%u", padcount);
newpad = g_object_new (GST_TYPE_ADDER_PAD, "name", name, "direction",
templ->direction, "template", templ, NULL);
GST_DEBUG_OBJECT (adder, "request new pad %s", name);
g_free (name);
gst_collect_pads_add_pad (adder->collect, newpad, sizeof (GstCollectData),
NULL, TRUE);
/* takes ownership of the pad */
if (!gst_element_add_pad (GST_ELEMENT (adder), newpad))
goto could_not_add;
gst_child_proxy_child_added (GST_CHILD_PROXY (adder), G_OBJECT (newpad),
GST_OBJECT_NAME (newpad));
return newpad;
/* errors */
not_sink:
{
g_warning ("gstadder: request new pad that is not a SINK pad\n");
return NULL;
}
could_not_add:
{
GST_DEBUG_OBJECT (adder, "could not add pad");
gst_collect_pads_remove_pad (adder->collect, newpad);
gst_object_unref (newpad);
return NULL;
}
}
static void
gst_adder_release_pad (GstElement * element, GstPad * pad)
{
GstAdder *adder;
adder = GST_ADDER (element);
GST_DEBUG_OBJECT (adder, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
gst_child_proxy_child_removed (GST_CHILD_PROXY (adder), G_OBJECT (pad),
GST_OBJECT_NAME (pad));
if (adder->collect)
gst_collect_pads_remove_pad (adder->collect, pad);
gst_element_remove_pad (element, pad);
}
static GstFlowReturn
gst_adder_do_clip (GstCollectPads * pads, GstCollectData * data,
GstBuffer * buffer, GstBuffer ** out, gpointer user_data)
{
GstAdder *adder = GST_ADDER (user_data);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&adder->info);
bpf = GST_AUDIO_INFO_BPF (&adder->info);
buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf);
*out = buffer;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_adder_collected (GstCollectPads * pads, gpointer user_data)
{
/*
* combine streams by adding data values
* basic algorithm :
* - this function is called when all pads have a buffer
* - get available bytes on all pads.
* - repeat for each input pad :
* - read available bytes, copy or add to target buffer
* - if there's an EOS event, remove the input channel
* - push out the output buffer
*
* todo:
* - would be nice to have a mixing mode, where instead of adding we mix
* - for float we could downscale after collect loop
* - for int we need to downscale each input to avoid clipping or
* mix into a temp (float) buffer and scale afterwards as well
*/
GstAdder *adder;
GSList *collected, *next = NULL;
GstFlowReturn ret;
GstBuffer *outbuf = NULL, *gapbuf = NULL;
GstMapInfo outmap = { NULL };
guint outsize;
gint64 next_offset;
gint64 next_timestamp;
gint rate, bps, bpf;
gboolean had_mute = FALSE;
gboolean is_eos = TRUE;
adder = GST_ADDER (user_data);
/* this is fatal */
if (G_UNLIKELY (adder->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN))
goto not_negotiated;
if (adder->flush_stop_pending == TRUE) {
GST_INFO_OBJECT (adder->srcpad, "send pending flush stop event");
if (!gst_pad_push_event (adder->srcpad, gst_event_new_flush_stop (TRUE))) {
GST_WARNING_OBJECT (adder->srcpad, "Sending flush stop event failed");
}
adder->flush_stop_pending = FALSE;
}
if (adder->send_stream_start) {
gchar s_id[32];
GstEvent *event;
GST_INFO_OBJECT (adder->srcpad, "send pending stream start event");
/* FIXME: create id based on input ids, we can't use
* gst_pad_create_stream_id() though as that only handles 0..1 sink-pad
*/
g_snprintf (s_id, sizeof (s_id), "adder-%08x", g_random_int ());
event = gst_event_new_stream_start (s_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
if (!gst_pad_push_event (adder->srcpad, event)) {
GST_WARNING_OBJECT (adder->srcpad, "Sending stream start event failed");
}
adder->send_stream_start = FALSE;
}
if (adder->send_caps) {
GstEvent *caps_event;
caps_event = gst_event_new_caps (adder->current_caps);
GST_INFO_OBJECT (adder->srcpad, "send pending caps event %" GST_PTR_FORMAT,
caps_event);
if (!gst_pad_push_event (adder->srcpad, caps_event)) {
GST_WARNING_OBJECT (adder->srcpad, "Sending caps event failed");
}
adder->send_caps = FALSE;
}
rate = GST_AUDIO_INFO_RATE (&adder->info);
bps = GST_AUDIO_INFO_BPS (&adder->info);
bpf = GST_AUDIO_INFO_BPF (&adder->info);
if (g_atomic_int_compare_and_exchange (&adder->new_segment_pending, TRUE,
FALSE)) {
GstEvent *event;
/*
* When seeking we set the start and stop positions as given in the seek
* event. We also adjust offset & timestamp accordingly.
* This basically ignores all newsegments sent by upstream.
*/
event = gst_event_new_segment (&adder->segment);
if (adder->segment.rate > 0.0) {
adder->segment.position = adder->segment.start;
} else {
adder->segment.position = adder->segment.stop;
}
adder->offset = gst_util_uint64_scale (adder->segment.position,
rate, GST_SECOND);
GST_INFO_OBJECT (adder->srcpad, "sending pending new segment event %"
GST_SEGMENT_FORMAT, &adder->segment);
if (event) {
if (!gst_pad_push_event (adder->srcpad, event)) {
GST_WARNING_OBJECT (adder->srcpad, "Sending new segment event failed");
}
} else {
GST_WARNING_OBJECT (adder->srcpad, "Creating new segment event for "
"start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed",
adder->segment.start, adder->segment.stop);
}
}
/* get available bytes for reading, this can be 0 which could mean empty
* buffers or EOS, which we will catch when we loop over the pads. */
outsize = gst_collect_pads_available (pads);
GST_LOG_OBJECT (adder,
"starting to cycle through channels, %d bytes available (bps = %d, bpf = %d)",
outsize, bps, bpf);
for (collected = pads->data; collected; collected = next) {
GstCollectData *collect_data;
GstBuffer *inbuf;
gboolean is_gap;
GstAdderPad *pad;
GstClockTime timestamp, stream_time;
/* take next to see if this is the last collectdata */
next = g_slist_next (collected);
collect_data = (GstCollectData *) collected->data;
pad = GST_ADDER_PAD (collect_data->pad);
/* get a buffer of size bytes, if we get a buffer, it is at least outsize
* bytes big. */
inbuf = gst_collect_pads_take_buffer (pads, collect_data, outsize);
if (!GST_COLLECT_PADS_STATE_IS_SET (collect_data,
GST_COLLECT_PADS_STATE_EOS))
is_eos = FALSE;
/* NULL means EOS or an empty buffer so we still need to flush in
* case of an empty buffer. */
if (inbuf == NULL) {
GST_LOG_OBJECT (adder, "channel %p: no bytes available", collect_data);
continue;
}
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
stream_time =
gst_segment_to_stream_time (&collect_data->segment, GST_FORMAT_TIME,
timestamp);
/* sync object properties on stream time */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time);
GST_OBJECT_LOCK (pad);
if (pad->mute || pad->volume < G_MINDOUBLE) {
had_mute = TRUE;
GST_DEBUG_OBJECT (adder, "channel %p: skipping muted pad", collect_data);
gst_buffer_unref (inbuf);
GST_OBJECT_UNLOCK (pad);
continue;
}
is_gap = GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP);
/* Try to make an output buffer */
if (outbuf == NULL) {
/* if this is a gap buffer but we have some more pads to check, skip it.
* If we are at the last buffer, take it, regardless if it is a GAP
* buffer or not. */
if (is_gap && next) {
GST_DEBUG_OBJECT (adder, "skipping, non-last GAP buffer");
/* we keep the GAP buffer, if we don't have anymore buffers (all pads
* EOS, we can use this one as the output buffer. */
if (gapbuf == NULL)
gapbuf = inbuf;
else
gst_buffer_unref (inbuf);
GST_OBJECT_UNLOCK (pad);
continue;
}
GST_LOG_OBJECT (adder, "channel %p: preparing output buffer of %d bytes",
collect_data, outsize);
/* make data and metadata writable, can simply return the inbuf when we
* are the only one referencing this buffer. If this is the last (and
* only) GAP buffer, it will automatically copy the GAP flag. */
outbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
if (pad->volume != 1.0) {
switch (adder->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
adder_orc_volume_u8 ((gpointer) outmap.data, pad->volume_i8,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_S8:
adder_orc_volume_s8 ((gpointer) outmap.data, pad->volume_i8,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_U16:
adder_orc_volume_u16 ((gpointer) outmap.data, pad->volume_i16,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_S16:
adder_orc_volume_s16 ((gpointer) outmap.data, pad->volume_i16,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_U32:
adder_orc_volume_u32 ((gpointer) outmap.data, pad->volume_i32,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_S32:
adder_orc_volume_s32 ((gpointer) outmap.data, pad->volume_i32,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_F32:
adder_orc_volume_f32 ((gpointer) outmap.data, pad->volume,
outmap.size / bps);
break;
case GST_AUDIO_FORMAT_F64:
adder_orc_volume_f64 ((gpointer) outmap.data, pad->volume,
outmap.size / bps);
break;
default:
g_assert_not_reached ();
break;
}
}
} else {
if (!is_gap) {
/* we had a previous output buffer, mix this non-GAP buffer */
GstMapInfo inmap;
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
/* all buffers should have outsize, there are no short buffers because we
* asked for the max size above */
g_assert (inmap.size == outmap.size);
GST_LOG_OBJECT (adder, "channel %p: mixing %" G_GSIZE_FORMAT " bytes"
" from data %p", collect_data, inmap.size, inmap.data);
/* further buffers, need to add them */
if (pad->volume == 1.0) {
switch (adder->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
adder_orc_add_u8 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S8:
adder_orc_add_s8 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_U16:
adder_orc_add_u16 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S16:
adder_orc_add_s16 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_U32:
adder_orc_add_u32 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S32:
adder_orc_add_s32 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_F32:
adder_orc_add_f32 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_F64:
adder_orc_add_f64 ((gpointer) outmap.data,
(gpointer) inmap.data, inmap.size / bps);
break;
default:
g_assert_not_reached ();
break;
}
} else {
switch (adder->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
adder_orc_add_volume_u8 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i8, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S8:
adder_orc_add_volume_s8 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i8, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_U16:
adder_orc_add_volume_u16 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i16, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S16:
adder_orc_add_volume_s16 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i16, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_U32:
adder_orc_add_volume_u32 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i32, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_S32:
adder_orc_add_volume_s32 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume_i32, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_F32:
adder_orc_add_volume_f32 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume, inmap.size / bps);
break;
case GST_AUDIO_FORMAT_F64:
adder_orc_add_volume_f64 ((gpointer) outmap.data,
(gpointer) inmap.data, pad->volume, inmap.size / bps);
break;
default:
g_assert_not_reached ();
break;
}
}
gst_buffer_unmap (inbuf, &inmap);
} else {
/* skip gap buffer */
GST_LOG_OBJECT (adder, "channel %p: skipping GAP buffer", collect_data);
}
gst_buffer_unref (inbuf);
}
GST_OBJECT_UNLOCK (pad);
}
if (outbuf)
gst_buffer_unmap (outbuf, &outmap);
if (is_eos)
goto eos;
if (outbuf == NULL) {
/* no output buffer, reuse one of the GAP buffers then if we have one */
if (gapbuf) {
GST_LOG_OBJECT (adder, "reusing GAP buffer %p", gapbuf);
outbuf = gapbuf;
} else if (had_mute) {
GstMapInfo map;
/* Means we had all pads muted, create some silence */
outbuf = gst_buffer_new_allocate (NULL, outsize, NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
gst_audio_format_fill_silence (adder->info.finfo, map.data, outsize);
gst_buffer_unmap (outbuf, &map);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
} else {
/* assume EOS otherwise, this should not happen, really */
goto eos;
}
} else if (gapbuf) {
/* we had an output buffer, unref the gapbuffer we kept */
gst_buffer_unref (gapbuf);
}
if (G_UNLIKELY (adder->pending_events)) {
GList *tmp = adder->pending_events;
while (tmp) {
GstEvent *ev = (GstEvent *) tmp->data;
gst_pad_push_event (adder->srcpad, ev);
tmp = g_list_next (tmp);
}
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
if (adder->segment.rate > 0.0) {
next_offset = adder->offset + outsize / bpf;
} else {
next_offset = adder->offset - outsize / bpf;
}
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
/* set timestamps on the output buffer */
GST_BUFFER_DTS (outbuf) = GST_CLOCK_TIME_NONE;
if (adder->segment.rate > 0.0) {
GST_BUFFER_PTS (outbuf) = adder->segment.position;
GST_BUFFER_OFFSET (outbuf) = adder->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
GST_BUFFER_DURATION (outbuf) = next_timestamp - adder->segment.position;
} else {
GST_BUFFER_PTS (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = adder->offset;
GST_BUFFER_DURATION (outbuf) = adder->segment.position - next_timestamp;
}
adder->offset = next_offset;
adder->segment.position = next_timestamp;
/* send it out */
GST_LOG_OBJECT (adder, "pushing outbuf %p, timestamp %" GST_TIME_FORMAT
" offset %" G_GINT64_FORMAT, outbuf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_OFFSET (outbuf));
ret = gst_pad_push (adder->srcpad, outbuf);
GST_LOG_OBJECT (adder, "pushed outbuf, result = %s", gst_flow_get_name (ret));
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
eos:
{
GST_DEBUG_OBJECT (adder, "no data available, must be EOS");
gst_pad_push_event (adder->srcpad, gst_event_new_eos ());
return GST_FLOW_EOS;
}
}
static GstStateChangeReturn
gst_adder_change_state (GstElement * element, GstStateChange transition)
{
GstAdder *adder;
GstStateChangeReturn ret;
adder = GST_ADDER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
adder->offset = 0;
adder->flush_stop_pending = FALSE;
adder->new_segment_pending = TRUE;
adder->send_stream_start = TRUE;
adder->send_caps = TRUE;
gst_caps_replace (&adder->current_caps, NULL);
gst_segment_init (&adder->segment, GST_FORMAT_TIME);
gst_collect_pads_start (adder->collect);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* need to unblock the collectpads before calling the
* parent change_state so that streaming can finish */
gst_collect_pads_stop (adder->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
/* GstChildProxy implementation */
static GObject *
gst_adder_child_proxy_get_child_by_index (GstChildProxy * child_proxy,
guint index)
{
GstAdder *adder = GST_ADDER (child_proxy);
GObject *obj = NULL;
GST_OBJECT_LOCK (adder);
obj = g_list_nth_data (GST_ELEMENT_CAST (adder)->sinkpads, index);
if (obj)
gst_object_ref (obj);
GST_OBJECT_UNLOCK (adder);
return obj;
}
static guint
gst_adder_child_proxy_get_children_count (GstChildProxy * child_proxy)
{
guint count = 0;
GstAdder *adder = GST_ADDER (child_proxy);
GST_OBJECT_LOCK (adder);
count = GST_ELEMENT_CAST (adder)->numsinkpads;
GST_OBJECT_UNLOCK (adder);
GST_INFO_OBJECT (adder, "Children Count: %d", count);
return count;
}
static void
gst_adder_child_proxy_init (gpointer g_iface, gpointer iface_data)
{
GstChildProxyInterface *iface = g_iface;
GST_INFO ("intializing child proxy interface");
iface->get_child_by_index = gst_adder_child_proxy_get_child_by_index;
iface->get_children_count = gst_adder_child_proxy_get_children_count;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "adder", 0,
"audio channel mixing element");
if (!gst_element_register (plugin, "adder", GST_RANK_NONE, GST_TYPE_ADDER)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
adder,
"Adds multiple streams",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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