gstreamer/gst/audiorate/gstaudiorate.c
Edward Hervey 317bb22aca gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Don't rely on incoming buffers offset anymore, since it is completely
broken when using multiple segments.
Instead convert the incoming buffers timestamp to running time, and
then convert that value to the offsets.
Also inform GstSegment of the last outputted stop position, which is
needed if we received several segments with an unknown stop value.
2006-08-29 10:32:34 +00:00

623 lines
18 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 <gst/audio/audio.h>
#define GST_CAT_DEFAULT audio_rate_debug
GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);
#define GST_TYPE_AUDIO_RATE \
(gst_audio_rate_get_type())
#define GST_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_AUDIO_RATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_IS_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RATE))
#define GST_IS_AUDIO_RATE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RATE))
typedef struct _GstAudioRate GstAudioRate;
typedef struct _GstAudioRateClass GstAudioRateClass;
struct _GstAudioRate
{
GstElement element;
GstPad *sinkpad, *srcpad;
/* audio format */
gint bytes_per_sample;
gint rate;
/* stats */
guint64 in, out, add, drop;
gboolean silent;
/* audio state */
guint64 offset;
guint64 next_offset;
gboolean discont;
GstSegment segment;
};
struct _GstAudioRateClass
{
GstElementClass parent_class;
};
/* elementfactory information */
static const GstElementDetails audio_rate_details =
GST_ELEMENT_DETAILS ("Audio rate adjuster",
"Filter/Effect/Audio",
"Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
"Wim Taymans <wim@fluendo.com>");
/* GstAudioRate signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_SILENT TRUE
enum
{
ARG_0,
ARG_IN,
ARG_OUT,
ARG_ADD,
ARG_DROP,
ARG_SILENT,
/* FILL ME */
};
static GstStaticPadTemplate gst_audio_rate_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static GstStaticPadTemplate gst_audio_rate_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static void gst_audio_rate_base_init (gpointer g_class);
static void gst_audio_rate_class_init (GstAudioRateClass * klass);
static void gst_audio_rate_init (GstAudioRate * audiorate);
static gboolean gst_audio_rate_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_audio_rate_src_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
static void gst_audio_rate_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_rate_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
/*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
static GType
gst_audio_rate_get_type (void)
{
static GType audio_rate_type = 0;
if (!audio_rate_type) {
static const GTypeInfo audio_rate_info = {
sizeof (GstAudioRateClass),
gst_audio_rate_base_init,
NULL,
(GClassInitFunc) gst_audio_rate_class_init,
NULL,
NULL,
sizeof (GstAudioRate),
0,
(GInstanceInitFunc) gst_audio_rate_init,
};
audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAudioRate", &audio_rate_info, 0);
}
return audio_rate_type;
}
static void
gst_audio_rate_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &audio_rate_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_rate_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_rate_src_template));
}
static void
gst_audio_rate_class_init (GstAudioRateClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
object_class->set_property = gst_audio_rate_set_property;
object_class->get_property = gst_audio_rate_get_property;
g_object_class_install_property (object_class, ARG_IN,
g_param_spec_uint64 ("in", "In",
"Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
g_object_class_install_property (object_class, ARG_OUT,
g_param_spec_uint64 ("out", "Out",
"Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
g_object_class_install_property (object_class, ARG_ADD,
g_param_spec_uint64 ("add", "Add",
"Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
g_object_class_install_property (object_class, ARG_DROP,
g_param_spec_uint64 ("drop", "Drop",
"Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
g_object_class_install_property (object_class, ARG_SILENT,
g_param_spec_boolean ("silent", "silent",
"Don't emit notify for dropped and duplicated frames",
DEFAULT_SILENT, G_PARAM_READWRITE));
element_class->change_state = gst_audio_rate_change_state;
}
static void
gst_audio_rate_reset (GstAudioRate * audiorate)
{
audiorate->offset = -1;
audiorate->next_offset = -1;
audiorate->discont = TRUE;
gst_segment_init (&audiorate->segment, GST_FORMAT_UNDEFINED);
GST_DEBUG_OBJECT (audiorate, "handle reset");
}
static gboolean
gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioRate *audiorate;
GstStructure *structure;
GstPad *otherpad;
gboolean ret = FALSE;
gint channels, width, rate;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "channels", &channels))
goto wrong_caps;
if (!gst_structure_get_int (structure, "width", &width))
goto wrong_caps;
if (!gst_structure_get_int (structure, "rate", &rate))
goto wrong_caps;
audiorate->bytes_per_sample = channels * (width / 8);
if (audiorate->bytes_per_sample == 0)
goto wrong_format;
audiorate->rate = rate;
/* the format is correct, configure caps on other pad */
otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
audiorate->srcpad;
ret = gst_pad_set_caps (otherpad, caps);
done:
gst_object_unref (audiorate);
return ret;
/* ERRORS */
wrong_caps:
{
GST_DEBUG_OBJECT (audiorate, "could not get channels/width from caps");
goto done;
}
wrong_format:
{
GST_DEBUG_OBJECT (audiorate, "bytes_per_samples gave 0");
goto done;
}
}
static void
gst_audio_rate_init (GstAudioRate * audiorate)
{
audiorate->sinkpad =
gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");
gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event);
gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);
gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
audiorate->srcpad =
gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");
gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event);
gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
audiorate->in = 0;
audiorate->out = 0;
audiorate->drop = 0;
audiorate->add = 0;
audiorate->silent = DEFAULT_SILENT;
}
static gboolean
gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP");
gst_audio_rate_reset (audiorate);
res = gst_pad_push_event (audiorate->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
/* FIXME:
* - sparse stream support. For this, the update flag is TRUE and the
* start/time positions are updated, meaning that time progressed by
* time - old_time amount and we need to fill that gap with empty
* samples.
* - fill the current segment if it has a valid stop position. This
* happens when the update flag is FALSE. With the segment helper we can
* calculate the accumulated time and compare this to the next_offset.
*/
if (!update) {
/* a new segment starts. We need to figure out what will be the next
* sample offset. We mark the offsets as invalid so that the _chain
* function will perform this calculation. */
audiorate->offset = -1;
audiorate->next_offset = -1;
}
gst_segment_set_newsegment_full (&audiorate->segment, update, rate, arate,
format, start, stop, time);
res = gst_pad_push_event (audiorate->srcpad, event);
break;
}
case GST_EVENT_EOS:
default:
res = gst_pad_push_event (audiorate->srcpad, event);
break;
}
gst_object_unref (audiorate);
return res;
}
static gboolean
gst_audio_rate_src_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
default:
res = gst_pad_push_event (audiorate->sinkpad, event);
break;
}
gst_object_unref (audiorate);
return res;
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time, in_duration, run_time;
guint64 in_offset, in_offset_end;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
if (audiorate->bytes_per_sample == 0)
goto not_negotiated;
if (audiorate->offset == -1) {
gint64 pos;
/* first buffer, we are negotiated and we have a segment, calculate the
* current expected offsets based on the segment.time, which is the first
* media time of the segment and should match the media time of the first
* buffer in that segment, which is the offset expressed in DEFAULT units.
*/
pos = audiorate->segment.time;
if (pos != 0) {
if (audiorate->segment.format == GST_FORMAT_TIME) {
/* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (pos, audiorate->rate, GST_SECOND);
} else {
/* FIXME, we don't know, start from 0 then... */
pos = 0;
}
}
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
audiorate->offset = pos;
audiorate->next_offset = pos;
}
audiorate->in++;
in_time = GST_BUFFER_TIMESTAMP (buf);
in_duration = GST_BUFFER_DURATION (buf);
in_size = GST_BUFFER_SIZE (buf);
/* don't really on buffer's offset */
/* We instead figure out using the runningtime version of the incoming buffer timestamp */
run_time =
gst_segment_to_running_time (&audiorate->segment, GST_FORMAT_TIME,
in_time);
in_offset = gst_util_uint64_scale_int (run_time, audiorate->rate, GST_SECOND);
in_offset_end = in_offset + in_size / audiorate->bytes_per_sample;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", run_time:%" GST_TIME_FORMAT
", in_duration:%" GST_TIME_FORMAT
", in_size:%u, in_offset:%lld, in_offset_end:%lld" ", ->next_offset:%lld",
GST_TIME_ARGS (in_time), GST_TIME_ARGS (run_time),
GST_TIME_ARGS (in_duration), in_size, in_offset, in_offset_end,
audiorate->next_offset);
/* do we need to insert samples */
if (in_offset > audiorate->next_offset) {
GstBuffer *fill;
gint fillsize;
guint64 fillsamples;
fillsamples = in_offset - audiorate->next_offset;
fillsize = fillsamples * audiorate->bytes_per_sample;
fill = gst_buffer_new_and_alloc (fillsize);
memset (GST_BUFFER_DATA (fill), 0, fillsize);
GST_LOG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size;
GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill);
GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
GST_BUFFER_OFFSET_END (fill) = in_offset;
/* we created this buffer to filla gap */
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
/* set discont if it's pending, this is mostly done for the first buffer and
* after a flushing seek */
if (audiorate->discont) {
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
audiorate->discont = FALSE;
}
ret = gst_pad_push (audiorate->srcpad, fill);
if (ret != GST_FLOW_OK)
goto beach;
audiorate->out++;
audiorate->add += fillsamples;
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "add");
} else if (in_offset < audiorate->next_offset) {
/* need to remove samples */
if (in_offset_end <= audiorate->next_offset) {
guint64 drop = in_size / audiorate->bytes_per_sample;
audiorate->drop += drop;
GST_LOG_OBJECT (audiorate, "dropping %lld samples", drop);
/* we can drop the buffer completely */
gst_buffer_unref (buf);
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "drop");
goto beach;
} else {
guint64 truncsamples;
guint truncsize, leftsize;
GstBuffer *trunc;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
truncsize = truncsamples * audiorate->bytes_per_sample;
leftsize = in_size - truncsize;
trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size;
GST_BUFFER_TIMESTAMP (trunc) =
in_time + in_duration - GST_BUFFER_DURATION (trunc);
GST_BUFFER_OFFSET (trunc) = audiorate->next_offset;
GST_BUFFER_OFFSET_END (trunc) = in_offset_end;
GST_LOG_OBJECT (audiorate, "truncating %lld samples", truncsamples);
gst_buffer_unref (buf);
buf = trunc;
audiorate->drop += truncsamples;
}
}
if (audiorate->discont) {
/* we need to output a discont buffer, do so now */
GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
audiorate->discont = FALSE;
} else if (GST_BUFFER_IS_DISCONT (buf)) {
/* else we make everything continuous so we can safely remove the DISCONT
* flag from the buffer if there was one */
GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer");
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
/* set last_stop on segment */
gst_segment_set_last_stop (&audiorate->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf));
ret = gst_pad_push (audiorate->srcpad, buf);
audiorate->out++;
audiorate->next_offset = in_offset_end;
beach:
audiorate->offset += in_size / audiorate->bytes_per_sample;
gst_object_unref (audiorate);
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT,
(NULL), ("pipeline error, format was not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static void
gst_audio_rate_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (object);
switch (prop_id) {
case ARG_SILENT:
audiorate->silent = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_rate_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (object);
switch (prop_id) {
case ARG_IN:
g_value_set_uint64 (value, audiorate->in);
break;
case ARG_OUT:
g_value_set_uint64 (value, audiorate->out);
break;
case ARG_ADD:
g_value_set_uint64 (value, audiorate->add);
break;
case ARG_DROP:
g_value_set_uint64 (value, audiorate->drop);
break;
case ARG_SILENT:
g_value_set_boolean (value, audiorate->silent);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
audiorate->in = 0;
audiorate->out = 0;
audiorate->drop = 0;
audiorate->bytes_per_sample = 0;
audiorate->add = 0;
gst_audio_rate_reset (audiorate);
break;
default:
break;
}
if (parent_class->change_state)
return parent_class->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,
"AudioRate stream fixer");
return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
GST_TYPE_AUDIO_RATE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audiorate",
"Adjusts audio frames",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)