/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
*
* 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.
*/
/* Element-Checklist-Version: 5 */
/**
* SECTION:element-audioresample
*
* <refsect2>
* Audioresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
* </programlisting>
* Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </para>
* </refsect2>
*
* Last reviewed on 2006-03-02 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
#define GST_CAT_DEFAULT audioresample_debug
/* elementfactory information */
static const GstElementDetails gst_audioresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
"David Schleef <ds@schleef.org>");
#define DEFAULT_FILTERLEN 16
enum
{
PROP_0,
PROP_FILTERLEN
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS ( \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true;" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"depth = (int) 32, " \
"signed = (boolean) true;" \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32; " \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
)
static GstStaticPadTemplate gst_audioresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
static gboolean audioresample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
static gboolean audioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
static gboolean audioresample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn audioresample_pushthrough (GstAudioresample *
audioresample);
static GstFlowReturn audioresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
static gboolean audioresample_start (GstBaseTransform * base);
static gboolean audioresample_stop (GstBaseTransform * base);
static gboolean audioresample_query (GstPad * pad, GstQuery * query);
static const GstQueryType *audioresample_query_type (GstPad * pad);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
static void
gst_audioresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_sink_template));
gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
}
static void
gst_audioresample_class_init (GstAudioresampleClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
g_object_class_install_property (gobject_class, PROP_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, DEFAULT_FILTERLEN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->start =
GST_DEBUG_FUNCPTR (audioresample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
GST_DEBUG_FUNCPTR (audioresample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (audioresample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
GST_DEBUG_FUNCPTR (audioresample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (audioresample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (audioresample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
GST_DEBUG_FUNCPTR (audioresample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
gst_audioresample_init (GstAudioresample * audioresample,
GstAudioresampleClass * klass)
{
GstBaseTransform *trans;
trans = GST_BASE_TRANSFORM (audioresample);
/* buffer alloc passthrough is too impossible. FIXME, it
* is trivial in the passthrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
audioresample->filter_length = DEFAULT_FILTERLEN;
audioresample->need_discont = FALSE;
gst_pad_set_query_function (trans->srcpad, audioresample_query);
gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
}
/* vmethods */
static gboolean
audioresample_start (GstBaseTransform * base)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
audioresample->resample = resample_new ();
audioresample->ts_offset = -1;
audioresample->offset = -1;
audioresample->next_ts = -1;
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
return TRUE;
}
static gboolean
audioresample_stop (GstBaseTransform * base)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
if (audioresample->resample) {
resample_free (audioresample->resample);
audioresample->resample = NULL;
}
gst_caps_replace (&audioresample->sinkcaps, NULL);
gst_caps_replace (&audioresample->srccaps, NULL);
return TRUE;
}
static gboolean
audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
g_assert (size);
/* this works for both float and int */
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
g_return_val_if_fail (ret, FALSE);
*size = width * channels / 8;
return TRUE;
}
static GstCaps *
audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
GstStructure *structure;
/* transform caps gives one single caps so we can just replace
* the rate property with our range. */
res = gst_caps_copy (caps);
structure = gst_caps_get_structure (res, 0);
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
return res;
}
static gboolean
resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
{
GstStructure *structure;
gboolean ret;
gint myinrate, myoutrate;
int mychannels;
gint width, depth;
ResampleFormat format;
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
structure = gst_caps_get_structure (incaps, 0);
/* get width */
ret = gst_structure_get_int (structure, "width", &width);
if (!ret)
goto no_width;
/* figure out the format */
if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
if (width == 32)
format = RESAMPLE_FORMAT_F32;
else if (width == 64)
format = RESAMPLE_FORMAT_F64;
else
goto wrong_depth;
} else {
/* for int, depth and width must be the same */
ret = gst_structure_get_int (structure, "depth", &depth);
if (!ret || width != depth)
goto not_equal;
if (width == 16)
format = RESAMPLE_FORMAT_S16;
else if (width == 32)
format = RESAMPLE_FORMAT_S32;
else
goto wrong_depth;
}
ret = gst_structure_get_int (structure, "rate", &myinrate);
ret &= gst_structure_get_int (structure, "channels", &mychannels);
if (!ret)
goto no_in_rate_channels;
structure = gst_caps_get_structure (outcaps, 0);
ret = gst_structure_get_int (structure, "rate", &myoutrate);
if (!ret)
goto no_out_rate;
if (channels)
*channels = mychannels;
if (inrate)
*inrate = myinrate;
if (outrate)
*outrate = myoutrate;
resample_set_format (state, format);
resample_set_n_channels (state, mychannels);
resample_set_input_rate (state, myinrate);
resample_set_output_rate (state, myoutrate);
return TRUE;
/* ERRORS */
no_width:
{
GST_DEBUG ("failed to get width from caps");
return FALSE;
}
not_equal:
{
GST_DEBUG ("width %d and depth %d must be the same", width, depth);
return FALSE;
}
wrong_depth:
{
GST_DEBUG ("unknown depth %d found", depth);
return FALSE;
}
no_in_rate_channels:
{
GST_DEBUG ("could not get input rate and channels");
return FALSE;
}
no_out_rate:
{
GST_DEBUG ("could not get output rate");
return FALSE;
}
}
static gboolean
audioresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *state;
GstCaps *srccaps, *sinkcaps;
gboolean use_internal = FALSE; /* whether we use the internal state */
gboolean ret = TRUE;
GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
size, direction == GST_PAD_SINK ? "SINK" : "SRC");
if (direction == GST_PAD_SINK) {
sinkcaps = caps;
srccaps = othercaps;
} else {
sinkcaps = othercaps;
srccaps = caps;
}
/* if the caps are the ones that _set_caps got called with; we can use
* our own state; otherwise we'll have to create a state */
if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
gst_caps_is_equal (srccaps, audioresample->srccaps)) {
use_internal = TRUE;
state = audioresample->resample;
} else {
GST_DEBUG_OBJECT (audioresample,
"caps are not the set caps, creating state");
state = resample_new ();
resample_set_filter_length (state, audioresample->filter_length);
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
}
if (direction == GST_PAD_SINK) {
/* asked to convert size of an incoming buffer */
*othersize = resample_get_output_size_for_input (state, size);
} else {
/* asked to convert size of an outgoing buffer */
*othersize = resample_get_input_size_for_output (state, size);
}
g_assert (*othersize % state->sample_size == 0);
/* we make room for one extra sample, given that the resampling filter
* can output an extra one for non-integral i_rate/o_rate */
GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
if (!use_internal) {
resample_free (state);
}
return ret;
}
static gboolean
audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate, outrate;
int channels;
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
&channels, &inrate, &outrate);
g_return_val_if_fail (ret, FALSE);
audioresample->channels = channels;
GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
audioresample->i_rate = inrate;
GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
audioresample->o_rate = outrate;
GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
gst_caps_replace (&audioresample->sinkcaps, incaps);
gst_caps_replace (&audioresample->srccaps, outcaps);
return TRUE;
}
static gboolean
audioresample_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioresample *audioresample;
audioresample = GST_AUDIORESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
resample_input_flush (audioresample->resample);
audioresample->ts_offset = -1;
audioresample->next_ts = -1;
audioresample->offset = -1;
break;
case GST_EVENT_NEWSEGMENT:
resample_input_pushthrough (audioresample->resample);
audioresample_pushthrough (audioresample);
audioresample->ts_offset = -1;
audioresample->next_ts = -1;
audioresample->offset = -1;
break;
case GST_EVENT_EOS:
resample_input_eos (audioresample->resample);
audioresample_pushthrough (audioresample);
break;
default:
break;
}
parent_class->event (base, event);
return TRUE;
}
static GstFlowReturn
audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
{
int outsize;
int outsamples;
ResampleState *r;
r = audioresample->resample;
outsize = resample_get_output_size (r);
GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
/* protect against mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
GST_WARNING_OBJECT (audioresample,
"overriding audioresample's outsize %d with outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
outsize = GST_BUFFER_SIZE (outbuf);
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
GST_WARNING_OBJECT (audioresample,
"audioresample's outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
outsamples = outsize / r->sample_size;
GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
if (audioresample->ts_offset != -1) {
audioresample->offset += outsamples;
audioresample->ts_offset += outsamples;
audioresample->next_ts =
gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
audioresample->o_rate);
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND,
audioresample->o_rate);
}
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
/* this is an error that when it happens, would need fixing in the
* resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
* and it gave us more ! */
GST_WARNING_OBJECT (audioresample,
"audioresample, you memory corrupting bastard. "
"you gave me outsize %d while my buffer was size %d",
outsize, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_ERROR;
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
GST_WARNING_OBJECT (audioresample,
"audioresample's written outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
GST_BUFFER_SIZE (outbuf) = outsize;
if (G_UNLIKELY (audioresample->need_discont)) {
GST_DEBUG_OBJECT (audioresample,
"marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
audioresample->need_discont = FALSE;
}
GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
return GST_FLOW_OK;
}
static gboolean
audioresample_check_discont (GstAudioresample * audioresample,
GstClockTime timestamp)
{
if (timestamp != GST_CLOCK_TIME_NONE &&
audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
timestamp != audioresample->prev_ts + audioresample->prev_duration) {
/* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
GstClockTimeDiff diff = timestamp -
(audioresample->prev_ts + audioresample->prev_duration);
if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
GST_WARNING_OBJECT (audioresample,
"encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
return TRUE;
}
}
return FALSE;
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioresample *audioresample;
ResampleState *r;
guchar *data, *datacopy;
gulong size;
GstClockTime timestamp;
audioresample = GST_AUDIORESAMPLE (base);
r = audioresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
/* Flush internal samples */
audioresample_pushthrough (audioresample);
/* Inform downstream element about discontinuity */
audioresample->need_discont = TRUE;
/* We want to recalculate the offset */
audioresample->ts_offset = -1;
}
if (audioresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstClockTime stime;
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
audioresample->next_ts = timestamp;
audioresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
audioresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
audioresample->prev_ts = timestamp;
audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
/* need to memdup, resample takes ownership. */
datacopy = g_memdup (data, size);
resample_add_input_data (r, datacopy, size, g_free, datacopy);
return audioresample_do_output (audioresample, outbuf);
}
/* push remaining data in the buffers out */
static GstFlowReturn
audioresample_pushthrough (GstAudioresample * audioresample)
{
int outsize;
ResampleState *r;
GstBuffer *outbuf;
GstFlowReturn res = GST_FLOW_OK;
GstBaseTransform *trans;
r = audioresample->resample;
outsize = resample_get_output_size (r);
if (outsize == 0) {
GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
goto done;
}
trans = GST_BASE_TRANSFORM (audioresample);
res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (trans->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
outsize);
goto done;
}
res = audioresample_do_output (audioresample, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK))
goto done;
res = gst_pad_push (trans->srcpad, outbuf);
done:
return res;
}
static gboolean
audioresample_query (GstPad * pad, GstQuery * query)
{
GstAudioresample *audioresample =
GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = audioresample->i_rate;
gint resampler_latency = audioresample->filter_length / 2;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
if ((peer = gst_pad_get_peer (trans->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG ("Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
if (rate != 0 && resampler_latency != 0)
latency =
gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
else
latency = 0;
GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG ("Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (audioresample);
return res;
}
static const GstQueryType *
audioresample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
audioresample->filter_length);
if (audioresample->resample) {
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
gst_element_post_message (GST_ELEMENT (audioresample),
gst_message_new_latency (GST_OBJECT (audioresample)));
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
g_value_set_int (value, audioresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
resample_init ();
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIORESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioresample",
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);
#ifdef __SYMBIAN32__
EXPORT_C
#endif
GstPluginDesc* _GST_PLUGIN_DESC()
{
return &gst_plugin_desc;
}