mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-25 11:11:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audiorate/gstaudiorate.c

816 lines
25 KiB
C
Raw Blame History

/* 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.
*/
/**
* SECTION:element-audiorate
* @see_also: #GstVideoRate
*
* This element takes an incoming stream of timestamped raw audio frames and
* produces a perfect stream by inserting or dropping samples as needed.
*
* This operation may be of use to link to elements that require or otherwise
* implicitly assume a perfect stream as they do not store timestamps,
* but derive this by some means (e.g. bitrate for some AVI cases).
*
* The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add
* and #GstAudioRate:drop can be read to obtain information about number of
* input samples, output samples, dropped samples (i.e. the number of unused
* input samples) and inserted samples (i.e. the number of samples added to
* stream).
*
* When the #GstAudioRate:silent property is set to FALSE, a GObject property
* notification will be emitted whenever one of the #GstAudioRate:add or
* #GstAudioRate:drop values changes.
* This can potentially cause performance degradation.
* Note that property notification will happen from the streaming thread, so
* applications should be prepared for this.
*
* If the #GstAudioRate:tolerance property is non-zero, and an incoming buffer's
* timestamp deviates less than the property indicates from what would make a
* 'perfect time', then no samples will be added or dropped.
* Note that the output is still guaranteed to be a perfect stream, which means
* that the incoming data is then simply shifted (by less than the indicated
* tolerance) to a perfect time.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v alsasrc ! audiorate ! wavenc ! filesink location=alsa.wav
* ]| Capture audio from an ALSA device, and turn it into a perfect stream
* for saving in a raw audio file.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include "gstaudiorate.h"
#define GST_CAT_DEFAULT audio_rate_debug
GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);
/* 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
#define DEFAULT_TOLERANCE 0
enum
{
ARG_0,
ARG_IN,
ARG_OUT,
ARG_ADD,
ARG_DROP,
ARG_SILENT,
ARG_TOLERANCE,
/* 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_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
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 | G_PARAM_STATIC_STRINGS));
/**
* GstAudioRate:tolerance
*
* The difference between incoming timestamp and next timestamp must exceed
* the given value for audiorate to add or drop samples.
*
* Since: 0.10.26
**/
g_object_class_install_property (object_class, ARG_TOLERANCE,
g_param_spec_uint64 ("tolerance", "tolerance",
"Only act if timestamp jitter/imperfection exceeds indicated tolerance (ns)",
0, G_MAXUINT64, DEFAULT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
element_class->change_state = gst_audio_rate_change_state;
}
static void
gst_audio_rate_reset (GstAudioRate * audiorate)
{
audiorate->next_offset = -1;
audiorate->next_ts = -1;
audiorate->discont = TRUE;
gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED);
gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME);
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;
audiorate->tolerance = DEFAULT_TOLERANCE;
}
static void
gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time)
{
GstBuffer *buf;
GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT
", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts),
GST_TIME_ARGS (time));
if (!GST_CLOCK_TIME_IS_VALID (time) ||
!GST_CLOCK_TIME_IS_VALID (audiorate->next_ts))
return;
/* feed an empty buffer to chain with the given timestamp,
* it will take care of filling */
buf = gst_buffer_new ();
GST_BUFFER_TIMESTAMP (buf) = time;
gst_audio_rate_chain (audiorate->sinkpad, buf);
}
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: bad things will likely happen if rate < 0 ... */
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. */
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);
audiorate->next_offset = -1;
audiorate->next_ts = -1;
} else {
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);
}
/* we accept all formats */
gst_segment_set_newsegment_full (&audiorate->sink_segment, update, rate,
arate, format, start, stop, time);
GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
&audiorate->sink_segment);
if (format == GST_FORMAT_TIME) {
/* TIME formats can be copied to src and forwarded */
res = gst_pad_push_event (audiorate->srcpad, event);
memcpy (&audiorate->src_segment, &audiorate->sink_segment,
sizeof (GstSegment));
} else {
/* other formats will be handled in the _chain function */
gst_event_unref (event);
res = TRUE;
}
break;
}
case GST_EVENT_EOS:
/* FIXME, fill last segment */
res = gst_pad_push_event (audiorate->srcpad, event);
break;
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 gboolean
gst_audio_rate_convert (GstAudioRate * audiorate,
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
if (src_fmt == dest_fmt) {
*dest_val = src_val;
return TRUE;
}
switch (src_fmt) {
case GST_FORMAT_DEFAULT:
switch (dest_fmt) {
case GST_FORMAT_BYTES:
*dest_val = src_val * audiorate->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_val =
gst_util_uint64_scale_int (src_val, GST_SECOND, audiorate->rate);
break;
default:
return FALSE;;
}
break;
case GST_FORMAT_BYTES:
switch (dest_fmt) {
case GST_FORMAT_DEFAULT:
*dest_val = src_val / audiorate->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND,
audiorate->rate * audiorate->bytes_per_sample);
break;
default:
return FALSE;;
}
break;
case GST_FORMAT_TIME:
switch (dest_fmt) {
case GST_FORMAT_BYTES:
*dest_val = gst_util_uint64_scale_int (src_val,
audiorate->rate * audiorate->bytes_per_sample, GST_SECOND);
break;
case GST_FORMAT_DEFAULT:
*dest_val =
gst_util_uint64_scale_int (src_val, audiorate->rate, GST_SECOND);
break;
default:
return FALSE;;
}
break;
default:
return FALSE;
}
return TRUE;
}
static gboolean
gst_audio_rate_convert_segments (GstAudioRate * audiorate)
{
GstFormat src_fmt, dst_fmt;
src_fmt = audiorate->sink_segment.format;
dst_fmt = audiorate->src_segment.format;
#define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \
src_fmt, audiorate->sink_segment.field, \
dst_fmt, &audiorate->src_segment.field);
audiorate->sink_segment.rate = audiorate->src_segment.rate;
audiorate->sink_segment.abs_rate = audiorate->src_segment.abs_rate;
audiorate->sink_segment.flags = audiorate->src_segment.flags;
audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;
CONVERT_VAL (start);
CONVERT_VAL (stop);
CONVERT_VAL (time);
CONVERT_VAL (accum);
CONVERT_VAL (last_stop);
#undef CONVERT_VAL
return TRUE;
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time;
guint64 in_offset, in_offset_end, in_samples;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTimeDiff diff;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
/* need to be negotiated now */
if (audiorate->bytes_per_sample == 0)
goto not_negotiated;
/* we have a new pending segment */
if (audiorate->next_offset == -1) {
gint64 pos;
/* update the TIME segment */
gst_audio_rate_convert_segments (audiorate);
/* first buffer, we are negotiated and we have a segment, calculate the
* current expected offsets based on the segment.start, 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.
*/
/* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (audiorate->src_segment.start,
audiorate->rate, GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
/* resyncing is a discont */
audiorate->discont = TRUE;
audiorate->next_offset = pos;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
GST_SECOND, audiorate->rate);
}
audiorate->in++;
in_time = GST_BUFFER_TIMESTAMP (buf);
if (in_time == GST_CLOCK_TIME_NONE) {
GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time");
in_time = audiorate->next_ts;
}
in_size = GST_BUFFER_SIZE (buf);
in_samples = in_size / audiorate->bytes_per_sample;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int_round (in_time, audiorate->rate,
GST_SECOND);
in_offset_end = in_offset + in_samples;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%"
G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT,
GST_TIME_ARGS (in_time),
GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, audiorate->rate)),
in_size, in_offset, in_offset_end, audiorate->next_offset);
diff = in_time - audiorate->next_ts;
if (diff <= (GstClockTimeDiff) audiorate->tolerance &&
diff >= (GstClockTimeDiff) - audiorate->tolerance) {
/* buffer time close enough to expected time,
* so produce a perfect stream by simply 'shifting'
* it to next ts and offset and sending */
GST_LOG_OBJECT (audiorate, "within tolerance %" GST_TIME_FORMAT,
GST_TIME_ARGS (audiorate->tolerance));
goto send;
}
/* do we need to insert samples */
if (in_offset > audiorate->next_offset) {
GstBuffer *fill;
gint fillsize;
guint64 fillsamples;
/* We don't want to allocate a single unreasonably huge buffer - it might
be hundreds of megabytes. So, limit each output buffer to one second of
audio */
fillsamples = in_offset - audiorate->next_offset;
while (fillsamples > 0) {
guint64 cursamples = MIN (fillsamples, audiorate->rate);
fillsamples -= cursamples;
fillsize = cursamples * audiorate->bytes_per_sample;
fill = gst_buffer_new_and_alloc (fillsize);
/* FIXME, 0 might not be the silence byte for the negotiated format. */
memset (GST_BUFFER_DATA (fill), 0, fillsize);
GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples",
cursamples);
GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
audiorate->next_offset += cursamples;
GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
/* Use next timestamp, then calculate following timestamp based on
* offset to get duration. Neccesary complexity to get 'perfect'
* streams */
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
GST_SECOND, audiorate->rate);
GST_BUFFER_DURATION (fill) = audiorate->next_ts -
GST_BUFFER_TIMESTAMP (fill);
/* we created this buffer to fill a 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;
}
gst_buffer_set_caps (fill, GST_PAD_CAPS (audiorate->srcpad));
ret = gst_pad_push (audiorate->srcpad, fill);
if (ret != GST_FLOW_OK)
goto beach;
audiorate->out++;
audiorate->add += cursamples;
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_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " 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_unref (buf);
buf = trunc;
gst_buffer_set_caps (buf, GST_PAD_CAPS (audiorate->srcpad));
audiorate->drop += truncsamples;
GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",
truncsamples);
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "drop");
}
}
if (GST_BUFFER_SIZE (buf) == 0)
goto beach;
send:
/* Now calculate parameters for whichever buffer (either the original
* or truncated one) we're pushing. */
GST_BUFFER_OFFSET (buf) = audiorate->next_offset;
GST_BUFFER_OFFSET_END (buf) = in_offset_end;
GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (in_offset_end,
GST_SECOND, audiorate->rate);
GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);
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->src_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:
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;
case ARG_TOLERANCE:
audiorate->tolerance = g_value_get_uint64 (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;
case ARG_TOLERANCE:
g_value_set_uint64 (value, audiorate->tolerance);
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)
Reference in a new issue View git blame Copy permalink