mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-30 21:51:09 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audiomixer/gstaudioaggregator.c
Olivier Crête 86fb628d09 audioaggregator: Register function name 
Otherwise, it sometimes segfaults with debugging enabled
2015-07-22 19:30:19 -04:00

1298 lines
39 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
* 2013 Sebastian Dröge <sebastian@centricular.com>
* 2014 Collabora
* Olivier Crete <olivier.crete@collabora.com>
*
* gstaudioaggregator.c:
*
* 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: gstaudioaggregator
* @short_description: manages a set of pads with the purpose of
* aggregating their buffers for raw audio
* @see_also: #GstAggregator
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstaudioaggregator.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
#define GST_CAT_DEFAULT audio_aggregator_debug
struct _GstAudioAggregatorPadPrivate
{
/* All members are protected by the pad object lock */
GstBuffer *buffer; /* current buffer we're mixing,
for comparison with collect.buffer
to see if we need to update our
cached values. */
guint position, size;
guint64 output_offset; /* Offset in output segment that
collect.pos refers to in the
current buffer. */
guint64 next_offset; /* Next expected offset in the input segment */
/* Last time we noticed a discont */
GstClockTime discont_time;
/* A new unhandled segment event has been received */
gboolean new_segment;
};
/*****************************************
* GstAudioAggregatorPad implementation *
*****************************************/
G_DEFINE_TYPE (GstAudioAggregatorPad, gst_audio_aggregator_pad,
GST_TYPE_AGGREGATOR_PAD);
static gboolean
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
GstAggregator * aggregator);
static void
gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass)
{
GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;
g_type_class_add_private (klass, sizeof (GstAudioAggregatorPadPrivate));
aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad);
}
static void
gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad)
{
pad->priv =
G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_PAD,
GstAudioAggregatorPadPrivate);
gst_audio_info_init (&pad->info);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
GstAggregator * aggregator)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (aggpad);
pad->priv->position = pad->priv->size = 0;
pad->priv->output_offset = pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_OBJECT_UNLOCK (aggpad);
return TRUE;
}
/**************************************
* GstAudioAggregator implementation *
**************************************/
struct _GstAudioAggregatorPrivate
{
GMutex mutex;
gboolean send_caps; /* aagg lock */
/* All three properties are unprotected, can't be modified while streaming */
/* Size in frames that is output per buffer */
GstClockTime output_buffer_duration;
GstClockTime alignment_threshold;
GstClockTime discont_wait;
/* Protected by srcpad stream clock */
/* Buffer starting at offset containing block_size frames */
GstBuffer *current_buffer;
/* counters to keep track of timestamps */
/* Readable with object lock, writable with both aag lock and object lock */
gint64 offset;
};
#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);
#define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex);
static void gst_audio_aggregator_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_dispose (GObject * object);
static gboolean gst_audio_aggregator_src_event (GstAggregator * agg,
GstEvent * event);
static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg,
GstAggregatorPad * aggpad, GstEvent * event);
static gboolean gst_audio_aggregator_src_query (GstAggregator * agg,
GstQuery * query);
static gboolean gst_audio_aggregator_start (GstAggregator * agg);
static gboolean gst_audio_aggregator_stop (GstAggregator * agg);
static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg);
static GstBuffer *gst_audio_aggregator_create_output_buffer (GstAudioAggregator
* aagg, guint num_frames);
static GstFlowReturn gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg,
gboolean timeout);
#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
enum
{
PROP_0,
PROP_OUTPUT_BUFFER_DURATION,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
};
G_DEFINE_ABSTRACT_TYPE (GstAudioAggregator, gst_audio_aggregator,
GST_TYPE_AGGREGATOR);
static GstClockTime
gst_audio_aggregator_get_next_time (GstAggregator * agg)
{
GstClockTime next_time;
GST_OBJECT_LOCK (agg);
if (agg->segment.position == -1)
next_time = agg->segment.start;
else
next_time = agg->segment.position;
GST_OBJECT_UNLOCK (agg);
return next_time;
}
static void
gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass;
g_type_class_add_private (klass, sizeof (GstAudioAggregatorPrivate));
gobject_class->set_property = gst_audio_aggregator_set_property;
gobject_class->get_property = gst_audio_aggregator_get_property;
gobject_class->dispose = gst_audio_aggregator_dispose;
gstaggregator_class->src_event =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event);
gstaggregator_class->sink_event =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event);
gstaggregator_class->src_query =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query);
gstaggregator_class->start = gst_audio_aggregator_start;
gstaggregator_class->stop = gst_audio_aggregator_stop;
gstaggregator_class->flush = gst_audio_aggregator_flush;
gstaggregator_class->aggregate =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate);
gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip);
gstaggregator_class->get_next_time = gst_audio_aggregator_get_next_time;
klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration",
"Output block size in nanoseconds", 1,
G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 0,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_audio_aggregator_init (GstAudioAggregator * aagg)
{
aagg->priv =
G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR,
GstAudioAggregatorPrivate);
g_mutex_init (&aagg->priv->mutex);
aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
aagg->current_caps = NULL;
gst_audio_info_init (&aagg->info);
gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
}
static void
gst_audio_aggregator_dispose (GObject * object)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
gst_caps_replace (&aagg->current_caps, NULL);
g_mutex_clear (&aagg->priv->mutex);
G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object);
}
static void
gst_audio_aggregator_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
switch (prop_id) {
case PROP_OUTPUT_BUFFER_DURATION:
aagg->priv->output_buffer_duration = g_value_get_uint64 (value);
gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
aagg->priv->output_buffer_duration,
aagg->priv->output_buffer_duration);
break;
case PROP_ALIGNMENT_THRESHOLD:
aagg->priv->alignment_threshold = g_value_get_uint64 (value);
break;
case PROP_DISCONT_WAIT:
aagg->priv->discont_wait = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
switch (prop_id) {
case PROP_OUTPUT_BUFFER_DURATION:
g_value_set_uint64 (value, aagg->priv->output_buffer_duration);
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, aagg->priv->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, aagg->priv->discont_wait);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* event handling */
static gboolean
gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event)
{
gboolean result;
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GST_DEBUG_OBJECT (agg->srcpad, "Got %s event on src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_QOS:
/* QoS might be tricky */
gst_event_unref (event);
return FALSE;
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
gst_event_unref (event);
return FALSE;
break;
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
gdouble rate;
GstSeekType start_type, stop_type;
gint64 start, stop;
GstFormat seek_format, dest_format;
/* parse the seek parameters */
gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
&start, &stop_type, &stop);
/* Check the seeking parametters before linking up */
if ((start_type != GST_SEEK_TYPE_NONE)
&& (start_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (aagg,
"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 (aagg,
"seeking failed, unhandled seek type for end: %d", stop_type);
goto done;
}
GST_OBJECT_LOCK (agg);
dest_format = agg->segment.format;
GST_OBJECT_UNLOCK (agg);
if (seek_format != dest_format) {
result = FALSE;
GST_DEBUG_OBJECT (aagg,
"seeking failed, unhandled seek format: %s",
gst_format_get_name (seek_format));
goto done;
}
}
break;
default:
break;
}
return
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg,
event);
done:
return result;
}
static gboolean
gst_audio_aggregator_sink_event (GstAggregator * agg,
GstAggregatorPad * aggpad, GstEvent * event)
{
gboolean res = TRUE;
GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->format != GST_FORMAT_TIME) {
GST_ERROR_OBJECT (agg, "Segment of type %s are not supported,"
" only TIME segments are supported",
gst_format_get_name (segment->format));
gst_event_unref (event);
event = NULL;
res = FALSE;
break;
}
GST_OBJECT_LOCK (agg);
if (segment->rate != agg->segment.rate) {
GST_ERROR_OBJECT (aggpad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, agg->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
} else if (segment->rate < 0.0) {
GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet");
res = FALSE;
gst_event_unref (event);
event = NULL;
} else {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (pad);
pad->priv->new_segment = TRUE;
GST_OBJECT_UNLOCK (pad);
}
GST_OBJECT_UNLOCK (agg);
break;
}
default:
break;
}
if (event != NULL)
return
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event
(agg, aggpad, event);
return res;
}
/* 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_audio_aggregator_query_duration (GstAudioAggregator * aagg,
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 (aagg));
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 (aagg, "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_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
res = gst_audio_aggregator_query_duration (aagg, query);
break;
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
GST_OBJECT_LOCK (aagg);
switch (format) {
case GST_FORMAT_TIME:
/* FIXME, bring to stream time, might be tricky */
gst_query_set_position (query, format, agg->segment.position);
res = TRUE;
break;
case GST_FORMAT_BYTES:
if (GST_AUDIO_INFO_BPF (&aagg->info)) {
gst_query_set_position (query, format, aagg->priv->offset *
GST_AUDIO_INFO_BPF (&aagg->info));
res = TRUE;
}
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, aagg->priv->offset);
res = TRUE;
break;
default:
break;
}
GST_OBJECT_UNLOCK (aagg);
break;
}
default:
res =
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query
(agg, query);
break;
}
return res;
}
void
gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstCaps * caps)
{
gboolean valid;
GST_OBJECT_LOCK (pad);
valid = gst_audio_info_from_caps (&pad->info, caps);
GST_OBJECT_UNLOCK (pad);
g_assert (valid);
}
gboolean
gst_audio_aggregator_set_src_caps (GstAudioAggregator * aagg, GstCaps * caps)
{
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
return FALSE;
}
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
if (!gst_audio_info_is_equal (&info, &aagg->info)) {
GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
gst_caps_replace (&aagg->current_caps, caps);
memcpy (&aagg->info, &info, sizeof (info));
aagg->priv->send_caps = TRUE;
}
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
/* send caps event later, after stream-start event */
return TRUE;
}
/* Must hold object lock and aagg lock to call */
static void
gst_audio_aggregator_reset (GstAudioAggregator * aagg)
{
GstAggregator *agg = GST_AGGREGATOR (aagg);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
agg->segment.position = -1;
aagg->priv->offset = 0;
gst_audio_info_init (&aagg->info);
gst_caps_replace (&aagg->current_caps, NULL);
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
}
static gboolean
gst_audio_aggregator_start (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gst_audio_aggregator_reset (aagg);
return TRUE;
}
static gboolean
gst_audio_aggregator_stop (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gst_audio_aggregator_reset (aagg);
return TRUE;
}
static GstFlowReturn
gst_audio_aggregator_flush (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
agg->segment.position = -1;
aagg->priv->offset = 0;
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** out)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
GST_OBJECT_LOCK (bpad);
*out = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return GST_FLOW_OK;
}
/* Called with the object lock for both the element and pad held,
* as well as the aagg lock
*/
static gboolean
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstBuffer * inbuf)
{
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
gint rate, bpf;
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
g_assert (pad->priv->buffer == NULL);
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
pad->priv->position = 0;
pad->priv->size = gst_buffer_get_size (inbuf) / bpf;
if (!GST_BUFFER_PTS_IS_VALID (inbuf)) {
if (pad->priv->output_offset == -1)
pad->priv->output_offset = aagg->priv->offset;
if (pad->priv->next_offset == -1)
pad->priv->next_offset = pad->priv->size;
else
pad->priv->next_offset += pad->priv->size;
goto done;
}
start_time = GST_BUFFER_PTS (inbuf);
end_time =
start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
rate);
start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND);
end_offset = start_offset + pad->priv->size;
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| pad->priv->new_segment || pad->priv->next_offset == -1) {
discont = TRUE;
pad->priv->new_segment = FALSE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= pad->priv->next_offset)
diff = pad->priv->next_offset - start_offset;
else
diff = start_offset - pad->priv->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (aagg->priv->discont_wait > 0) {
if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
pad->priv->discont_time = start_time;
} else if (start_time - pad->priv->discont_time >=
aagg->priv->discont_wait) {
discont = TRUE;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (pad->priv->next_offset != -1)
GST_INFO_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
pad->priv->next_offset, start_offset);
pad->priv->output_offset = -1;
pad->priv->next_offset = end_offset;
} else {
pad->priv->next_offset += pad->priv->size;
}
if (pad->priv->output_offset == -1) {
GstClockTime start_running_time;
GstClockTime end_running_time;
guint64 start_running_time_offset;
guint64 end_running_time_offset;
start_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, start_time);
end_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, end_time);
start_running_time_offset =
gst_util_uint64_scale (start_running_time, rate, GST_SECOND);
end_running_time_offset =
gst_util_uint64_scale (end_running_time, rate, GST_SECOND);
if (end_running_time_offset < aagg->priv->offset) {
/* Before output segment, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_running_time_offset, aagg->priv->offset);
return FALSE;
}
if (start_running_time_offset < aagg->priv->offset) {
guint diff = aagg->priv->offset - start_running_time_offset;
pad->priv->position += diff;
if (pad->priv->position >= pad->priv->size) {
/* Empty buffer, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_running_time_offset,
aagg->priv->offset);
return FALSE;
}
}
pad->priv->output_offset =
MAX (start_running_time_offset, aagg->priv->offset);
GST_DEBUG_OBJECT (pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current audio aggregator offset %" G_GUINT64_FORMAT,
pad->priv->output_offset, aagg->priv->offset);
}
done:
GST_LOG_OBJECT (pad,
"Queued new buffer at offset %" G_GUINT64_FORMAT,
pad->priv->output_offset);
pad->priv->buffer = inbuf;
return TRUE;
}
/* Called with pad object lock held */
static gboolean
gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf)
{
guint overlap;
guint out_start;
gboolean filled;
guint blocksize;
blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
blocksize = MAX (1, blocksize);
/* Overlap => mix */
if (aagg->priv->offset < pad->priv->output_offset)
out_start = pad->priv->output_offset - aagg->priv->offset;
else
out_start = 0;
overlap = pad->priv->size - pad->priv->position;
if (overlap > blocksize - out_start)
overlap = blocksize - out_start;
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
/* skip gap buffer */
GST_LOG_OBJECT (pad, "skipping GAP buffer");
pad->priv->output_offset += pad->priv->size;
pad->priv->position = pad->priv->size;
gst_buffer_replace (&pad->priv->buffer, NULL);
return FALSE;
}
filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg,
pad, inbuf, pad->priv->position, outbuf, out_start, overlap);
if (filled)
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);
pad->priv->position += overlap;
pad->priv->output_offset += overlap;
if (pad->priv->position == pad->priv->size) {
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
return FALSE;
}
return TRUE;
}
static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
guint num_frames)
{
GstBuffer *outbuf = gst_buffer_new_allocate (NULL, num_frames *
GST_AUDIO_INFO_BPF (&aagg->info), NULL);
GstMapInfo outmap;
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
gst_buffer_unmap (outbuf, &outmap);
return outbuf;
}
static gboolean
sync_pad_values (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad)
{
GstAggregatorPad *bpad = GST_AGGREGATOR_PAD (pad);
GstClockTime timestamp, stream_time;
if (pad->priv->buffer == NULL)
return TRUE;
timestamp = GST_BUFFER_PTS (pad->priv->buffer);
GST_OBJECT_LOCK (bpad);
stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
timestamp);
GST_OBJECT_UNLOCK (bpad);
/* sync object properties on stream time */
/* TODO: Ideally we would want to do that on every sample */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time);
return TRUE;
}
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
/* Get all pads that have data for us and store them in a
* new list.
*
* Calculate the current output offset/timestamp and
* offset_end/timestamp_end. Allocate a silence buffer
* for this and store it.
*
* For all pads:
* 1) Once per input buffer (cached)
* 1) Check discont (flag and timestamp with tolerance)
* 2) If discont or new, resync. That means:
* 1) Drop all start data of the buffer that comes before
* the current position/offset.
* 2) Calculate the offset (output segment!) that the first
* frame of the input buffer corresponds to. Base this on
* the running time.
*
* 2) If the current pad's offset/offset_end overlaps with the output
* offset/offset_end, mix it at the appropiate position in the output
* buffer and advance the pad's position. Remember if this pad needs
* a new buffer to advance behind the output offset_end.
*
* 3) If we had no pad with a buffer, go EOS.
*
* 4) If we had at least one pad that did not advance behind output
* offset_end, let collected be called again for the current
* output offset/offset_end.
*/
GstElement *element;
GstAudioAggregator *aagg;
GList *iter;
GstFlowReturn ret;
GstBuffer *outbuf = NULL;
gint64 next_offset;
gint64 next_timestamp;
gint rate, bpf;
gboolean dropped = FALSE;
gboolean is_eos = TRUE;
gboolean is_done = TRUE;
guint blocksize;
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
/* Sync pad properties to the stream time */
gst_aggregator_iterate_sinkpads (agg,
(GstAggregatorPadForeachFunc) GST_DEBUG_FUNCPTR (sync_pad_values), NULL);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
/* Update position from the segment start/stop if needed */
if (agg->segment.position == -1) {
if (agg->segment.rate > 0.0)
agg->segment.position = agg->segment.start;
else
agg->segment.position = agg->segment.stop;
}
if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (timeout) {
GST_DEBUG_OBJECT (aagg,
"Got timeout before receiving any caps, don't output anything");
/* Advance position */
if (agg->segment.rate > 0.0)
agg->segment.position += aagg->priv->output_buffer_duration;
else if (agg->segment.position > aagg->priv->output_buffer_duration)
agg->segment.position -= aagg->priv->output_buffer_duration;
else
agg->segment.position = 0;
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
} else {
GST_OBJECT_UNLOCK (agg);
goto not_negotiated;
}
}
if (aagg->priv->send_caps) {
GST_OBJECT_UNLOCK (agg);
gst_aggregator_set_src_caps (agg, aagg->current_caps);
GST_OBJECT_LOCK (agg);
aagg->priv->offset = gst_util_uint64_scale (agg->segment.position,
GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
aagg->priv->send_caps = FALSE;
}
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
blocksize = MAX (1, blocksize);
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
/* FIXME: Reverse mixing does not work at all yet */
if (agg->segment.rate > 0.0) {
next_offset = aagg->priv->offset + blocksize;
} else {
next_offset = aagg->priv->offset - blocksize;
}
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
if (aagg->priv->current_buffer == NULL) {
GST_OBJECT_UNLOCK (agg);
aagg->priv->current_buffer =
GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
blocksize);
/* Be careful, some things could have changed ? */
GST_OBJECT_LOCK (agg);
GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
}
outbuf = aagg->priv->current_buffer;
GST_LOG_OBJECT (agg,
"Starting to mix %u samples for offset %" G_GUINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, blocksize,
aagg->priv->offset, GST_TIME_ARGS (agg->segment.position));
for (iter = element->sinkpads; iter; iter = iter->next) {
GstBuffer *inbuf;
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
gboolean drop_buf = FALSE;
gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);
if (!pad_eos)
is_eos = FALSE;
inbuf = gst_aggregator_pad_get_buffer (aggpad);
GST_OBJECT_LOCK (pad);
if (!inbuf) {
if (timeout) {
if (pad->priv->output_offset < next_offset) {
gint64 diff = next_offset - pad->priv->output_offset;
GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%"
GST_TIME_FORMAT ")", diff,
GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))));
}
} else if (!pad_eos) {
is_done = FALSE;
}
GST_OBJECT_UNLOCK (pad);
continue;
}
g_assert (!pad->priv->buffer || pad->priv->buffer == inbuf);
/* New buffer? */
if (!pad->priv->buffer) {
/* Takes ownership of buffer */
if (!gst_audio_aggregator_fill_buffer (aagg, pad, inbuf)) {
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
} else {
gst_buffer_unref (inbuf);
}
if (!pad->priv->buffer && !dropped && pad_eos) {
GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
GST_OBJECT_UNLOCK (pad);
continue;
}
g_assert (pad->priv->buffer);
/* This pad is lacking behind, we need to update the offset
* and maybe drop the current buffer */
if (pad->priv->output_offset < aagg->priv->offset) {
gint64 diff = aagg->priv->offset - pad->priv->output_offset;
gint64 odiff = diff;
if (pad->priv->position + diff > pad->priv->size)
diff = pad->priv->size - pad->priv->position;
pad->priv->position += diff;
pad->priv->output_offset += diff;
if (pad->priv->position == pad->priv->size) {
GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
", dropping %" GST_PTR_FORMAT,
GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
}
if (pad->priv->output_offset >= aagg->priv->offset
&& pad->priv->output_offset <
aagg->priv->offset + blocksize && pad->priv->buffer) {
GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
outbuf);
if (pad->priv->output_offset >= next_offset) {
GST_DEBUG_OBJECT (pad,
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
G_GUINT64_FORMAT, pad->priv->output_offset, next_offset);
} else {
is_done = FALSE;
}
}
GST_OBJECT_UNLOCK (pad);
if (drop_buf)
gst_aggregator_pad_drop_buffer (aggpad);
}
GST_OBJECT_UNLOCK (agg);
if (dropped) {
/* We dropped a buffer, retry */
GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
if (!is_done && !is_eos) {
/* Get more buffers */
GST_INFO_OBJECT (aagg,
"We're not done yet for the current offset," " waiting for more data");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
if (is_eos) {
gint64 max_offset = 0;
GST_DEBUG_OBJECT (aagg, "We're EOS");
GST_OBJECT_LOCK (agg);
for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset);
}
GST_OBJECT_UNLOCK (agg);
/* This means EOS or nothing mixed in at all */
if (aagg->priv->offset == max_offset) {
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_EOS;
}
if (max_offset <= next_offset) {
GST_DEBUG_OBJECT (aagg,
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GUINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
if (next_offset > aagg->priv->offset)
gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf);
}
}
/* set timestamps on the output buffer */
GST_OBJECT_LOCK (agg);
if (agg->segment.rate > 0.0) {
GST_BUFFER_PTS (outbuf) = agg->segment.position;
GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position;
} else {
GST_BUFFER_PTS (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset;
GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
}
aagg->priv->offset = next_offset;
agg->segment.position = next_timestamp;
GST_OBJECT_UNLOCK (agg);
/* send it out */
GST_LOG_OBJECT (aagg,
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
GST_BUFFER_OFFSET (outbuf));
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
ret = gst_aggregator_finish_buffer (agg, aagg->priv->current_buffer);
aagg->priv->current_buffer = NULL;
GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));
return ret;
/* ERRORS */
not_negotiated:
{
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
Reference in a new issue View git blame Copy permalink