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gstreamer/gst/audiorate/gstaudiorate.c

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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audiorate
* @title: 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.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
* ]|
* Capture audio from the sound card and turn it into a perfect stream
* for saving in a raw audio file.
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
* ]|
* Decodes an audio file and transforms it into a perfect stream for saving
* in a raw audio WAV file. Without the audio rate, the timing might not be
* preserved correctly in the WAV file in case the decoded stream is jittery
* or there are samples missing.
*
*/
#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);
/* GstAudioRate signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_SILENT TRUE
#define DEFAULT_TOLERANCE (40 * GST_MSECOND)
#define DEFAULT_SKIP_TO_FIRST FALSE
enum
{
PROP_0,
PROP_IN,
PROP_OUT,
PROP_ADD,
PROP_DROP,
PROP_SILENT,
PROP_TOLERANCE,
PROP_SKIP_TO_FIRST
};
static GstStaticPadTemplate gst_audio_rate_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) { interleaved, non-interleaved }")
);
static GstStaticPadTemplate gst_audio_rate_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) { interleaved, non-interleaved }")
);
static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent,
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 guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
static GParamSpec *pspec_drop = NULL;
static GParamSpec *pspec_add = NULL;
#define gst_audio_rate_parent_class parent_class
G_DEFINE_TYPE (GstAudioRate, gst_audio_rate, GST_TYPE_ELEMENT);
static void
gst_audio_rate_class_init (GstAudioRateClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (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, PROP_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, PROP_OUT,
g_param_spec_uint64 ("out", "Out", "Number of output samples", 0,
G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
pspec_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, PROP_ADD, pspec_add);
pspec_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, PROP_DROP, pspec_drop);
g_object_class_install_property (object_class, PROP_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.
*/
g_object_class_install_property (object_class, PROP_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));
/**
* GstAudioRate:skip-to-first:
*
* Don't produce buffers before the first one we receive.
*/
g_object_class_install_property (object_class, PROP_SKIP_TO_FIRST,
g_param_spec_boolean ("skip-to-first", "Skip to first buffer",
"Don't produce buffers before the first one we receive",
DEFAULT_SKIP_TO_FIRST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (element_class,
"Audio rate adjuster", "Filter/Effect/Audio",
"Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
"Wim Taymans <wim@fluendo.com>");
gst_element_class_add_static_pad_template (element_class,
&gst_audio_rate_sink_template);
gst_element_class_add_static_pad_template (element_class,
&gst_audio_rate_src_template);
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 (GstAudioRate * audiorate, GstCaps * caps)
{
GstAudioInfo info;
gint prev_rate = 0;
if (!gst_audio_info_from_caps (&info, caps))
goto wrong_caps;
prev_rate = audiorate->info.rate;
audiorate->info = info;
if (audiorate->next_offset >= 0 && prev_rate > 0 && prev_rate != info.rate) {
GST_DEBUG_OBJECT (audiorate,
"rate changed from %d to %d", prev_rate, info.rate);
/* calculate next_offset based on new rate value */
audiorate->next_offset =
gst_util_uint64_scale_int_round (audiorate->next_ts,
info.rate, GST_SECOND);
}
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_DEBUG_OBJECT (audiorate, "could not parse caps");
return FALSE;
}
}
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_PROXY_CAPS (audiorate->sinkpad);
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_PROXY_CAPS (audiorate->srcpad);
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, GST_OBJECT_CAST (audiorate), buf);
}
static gboolean
gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
audiorate = GST_AUDIO_RATE (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
if ((res = gst_audio_rate_setcaps (audiorate, caps))) {
res = gst_pad_push_event (audiorate->srcpad, event);
} else {
gst_event_unref (event);
}
break;
}
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_SEGMENT:
{
gst_event_copy_segment (event, &audiorate->sink_segment);
GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
#if 0
/* 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);
#endif
audiorate->next_offset = -1;
audiorate->next_ts = -1;
#if 0
} else {
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);
}
#endif
GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
&audiorate->sink_segment);
if (audiorate->sink_segment.format == GST_FORMAT_TIME) {
/* TIME formats can be copied to src and forwarded */
res = gst_pad_push_event (audiorate->srcpad, event);
gst_segment_copy_into (&audiorate->sink_segment,
&audiorate->src_segment);
} else {
/* other formats will be handled in the _chain function */
gst_event_unref (event);
res = TRUE;
}
break;
}
case GST_EVENT_EOS:
/* Fill segment until the end */
if (GST_CLOCK_TIME_IS_VALID (audiorate->src_segment.stop))
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);
res = gst_pad_push_event (audiorate->srcpad, event);
break;
case GST_EVENT_GAP:
{
/* Fill until end of gap */
GstClockTime timestamp, duration;
gst_event_parse_gap (event, &timestamp, &duration);
gst_event_unref (event);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (GST_CLOCK_TIME_IS_VALID (duration))
timestamp += duration;
gst_audio_rate_fill_to_time (audiorate, timestamp);
}
res = TRUE;
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
break;
}
return res;
}
static gboolean
gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
audiorate = GST_AUDIO_RATE (parent);
switch (GST_EVENT_TYPE (event)) {
default:
res = gst_pad_push_event (audiorate->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_audio_rate_convert (GstAudioRate * audiorate,
GstFormat src_fmt, guint64 src_val, GstFormat dest_fmt, guint64 * dest_val)
{
return gst_audio_info_convert (&audiorate->info, src_fmt, src_val, dest_fmt,
(gint64 *) dest_val);
}
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.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 (base);
CONVERT_VAL (position);
#undef CONVERT_VAL
return TRUE;
}
static void
gst_audio_rate_notify_drop (GstAudioRate * audiorate)
{
g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop);
}
static void
gst_audio_rate_notify_add (GstAudioRate * audiorate)
{
g_object_notify_by_pspec ((GObject *) audiorate, pspec_add);
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstObject * parent, 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;
gint rate, bpf;
GstAudioMeta *meta;
audiorate = GST_AUDIO_RATE (parent);
rate = GST_AUDIO_INFO_RATE (&audiorate->info);
bpf = GST_AUDIO_INFO_BPF (&audiorate->info);
/* need to be negotiated now */
if (bpf == 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_round (audiorate->src_segment.start,
GST_AUDIO_INFO_RATE (&audiorate->info), 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_round (audiorate->next_offset, GST_SECOND,
GST_AUDIO_INFO_RATE (&audiorate->info));
if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead");
pos = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf),
GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT,
pos);
audiorate->next_offset = pos;
audiorate->next_ts = GST_BUFFER_TIMESTAMP (buf);
}
}
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;
}
meta = gst_buffer_get_audio_meta (buf);
in_size = gst_buffer_get_size (buf);
in_samples = meta ? meta->samples : in_size / bpf;
audiorate->in += in_samples;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int_round (in_time, 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 ", ->next_ts:%"
GST_TIME_FORMAT, GST_TIME_ARGS (in_time),
GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, rate)),
in_size, in_offset, in_offset_end, audiorate->next_offset,
GST_TIME_ARGS (audiorate->next_ts));
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));
/* The outgoing buffer's offset will be set to ->next_offset, we also
* need to adjust the offset_end value accordingly */
in_offset_end = audiorate->next_offset + in_samples;
audiorate->out += in_samples;
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, rate);
GstMapInfo fillmap;
fillsamples -= cursamples;
fillsize = cursamples * bpf;
fill = gst_buffer_new_and_alloc (fillsize);
gst_buffer_map (fill, &fillmap, GST_MAP_WRITE);
gst_audio_format_info_fill_silence (audiorate->info.finfo, fillmap.data,
fillmap.size);
gst_buffer_unmap (fill, &fillmap);
if (audiorate->info.layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (fill, &audiorate->info, cursamples, NULL);
}
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. Necessary complexity to get 'perfect'
* streams */
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
audiorate->next_ts =
gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,
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;
}
fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf);
if (fill)
ret = gst_pad_push (audiorate->srcpad, fill);
if (ret != GST_FLOW_OK)
goto beach;
audiorate->out += cursamples;
audiorate->add += cursamples;
if (!audiorate->silent)
gst_audio_rate_notify_add (audiorate);
}
} else if (in_offset < audiorate->next_offset) {
/* need to remove samples */
if (in_offset_end <= audiorate->next_offset) {
guint64 drop = in_samples;
audiorate->drop += drop;
GST_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " samples",
drop);
/* we can drop the buffer completely */
gst_buffer_unref (buf);
buf = NULL;
if (!audiorate->silent)
gst_audio_rate_notify_drop (audiorate);
goto beach;
} else {
guint64 truncsamples, leftsamples;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
leftsamples = in_samples - truncsamples;
buf = gst_audio_buffer_truncate (buf, bpf, truncsamples, leftsamples);
audiorate->drop += truncsamples;
audiorate->out += leftsamples;
GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",
truncsamples);
if (!audiorate->silent)
gst_audio_rate_notify_drop (audiorate);
}
}
send:
if (gst_buffer_get_size (buf) == 0)
goto beach;
buf = gst_buffer_make_writable (buf);
/* 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_round (in_offset_end,
GST_SECOND, 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");
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");
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf);
if (buf) {
/* set last_stop on segment */
audiorate->src_segment.position =
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
ret = gst_pad_push (audiorate->srcpad, buf);
}
buf = NULL;
audiorate->next_offset = in_offset_end;
beach:
if (buf)
gst_buffer_unref (buf);
return ret;
/* ERRORS */
not_negotiated:
{
gst_buffer_unref (buf);
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 PROP_SILENT:
audiorate->silent = g_value_get_boolean (value);
break;
case PROP_TOLERANCE:
audiorate->tolerance = g_value_get_uint64 (value);
break;
case PROP_SKIP_TO_FIRST:
audiorate->skip_to_first = 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 PROP_IN:
g_value_set_uint64 (value, audiorate->in);
break;
case PROP_OUT:
g_value_set_uint64 (value, audiorate->out);
break;
case PROP_ADD:
g_value_set_uint64 (value, audiorate->add);
break;
case PROP_DROP:
g_value_set_uint64 (value, audiorate->drop);
break;
case PROP_SILENT:
g_value_set_boolean (value, audiorate->silent);
break;
case PROP_TOLERANCE:
g_value_set_uint64 (value, audiorate->tolerance);
break;
case PROP_SKIP_TO_FIRST:
g_value_set_boolean (value, audiorate->skip_to_first);
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->add = 0;
gst_audio_info_init (&audiorate->info);
gst_audio_rate_reset (audiorate);
break;
default:
break;
}
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
}
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)
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