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gstreamer/gst/audiofx/audiowsinclimit.c
Sebastian Dröge a490cffe5f gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ... 
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
2007-08-16 09:48:27 +00:00

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/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007 Sebastian Dröge <slomo@circular-chaos.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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
* TODO: - Implement the convolution in place, probably only makes sense
* when using FFT convolution as currently the convolution itself
* is probably the bottleneck
* - Maybe allow cascading the filter to get a better stopband attenuation.
* Can be done by convolving a filter kernel with itself
* - Drop the first kernel_length/2 samples and append the same number of
* samples on EOS as the first few samples are essentialy zero.
*/
/**
* SECTION:element-lpwsinc
* @short_description: Windows Sinc low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! lpwsinc mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "gstlpwsinc.h"
#define GST_CAT_DEFAULT gst_lpwsinc_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails lpwsinc_details = GST_ELEMENT_DETAILS ("LPWSinc",
"Filter/Effect/Audio",
"Low-pass and High-pass Windowed sinc filter",
"Thomas <thomas@apestaart.org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LENGTH,
PROP_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_LPWSINC_MODE (gst_lpwsinc_mode_get_type ())
static GType
gst_lpwsinc_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstLPWSincMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN
};
#define GST_TYPE_LPWSINC_WINDOW (gst_lpwsinc_window_get_type ())
static GType
gst_lpwsinc_window_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{WINDOW_HAMMING, "Hamming window (default)",
"hamming"},
{WINDOW_BLACKMAN, "Blackman window",
"blackman"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstLPWSincWindow", values);
}
return gtype;
}
#define ALLOWED_CAPS \
"audio/x-raw-float, " \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_lpwsinc_debug, "lpwsinc", 0, "Low-pass and High-pass Windowed sinc filter plugin");
GST_BOILERPLATE_FULL (GstLPWSinc, gst_lpwsinc, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void lpwsinc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void lpwsinc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn lpwsinc_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean lpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size);
static gboolean lpwsinc_start (GstBaseTransform * base);
static gboolean lpwsinc_event (GstBaseTransform * base, GstEvent * event);
static gboolean lpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
static gboolean lpwsinc_query (GstPad * pad, GstQuery * query);
static const GstQueryType *lpwsinc_query_type (GstPad * pad);
/* Element class */
static void
gst_lpwsinc_dispose (GObject * object)
{
GstLPWSinc *self = GST_LPWSINC (object);
if (self->residue) {
g_free (self->residue);
self->residue = NULL;
}
if (self->kernel) {
g_free (self->kernel);
self->kernel = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_lpwsinc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &lpwsinc_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_lpwsinc_class_init (GstLPWSincClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = lpwsinc_set_property;
gobject_class->get_property = lpwsinc_get_property;
gobject_class->dispose = gst_lpwsinc_dispose;
g_object_class_install_property (gobject_class, PROP_FREQUENCY,
g_param_spec_double ("frequency", "Frequency",
"Cut-off Frequency (Hz)", 0.0, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
3, G_MAXINT, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_LPWSINC_MODE,
MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_LPWSINC_WINDOW,
WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
trans_class->transform = GST_DEBUG_FUNCPTR (lpwsinc_transform);
trans_class->get_unit_size = GST_DEBUG_FUNCPTR (lpwsinc_get_unit_size);
trans_class->start = GST_DEBUG_FUNCPTR (lpwsinc_start);
trans_class->event = GST_DEBUG_FUNCPTR (lpwsinc_event);
filter_class->setup = GST_DEBUG_FUNCPTR (lpwsinc_setup);
}
static void
gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
{
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
self->latency = 50;
self->frequency = 0.0;
self->kernel = NULL;
self->residue = NULL;
self->have_kernel = FALSE;
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, lpwsinc_query);
gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
lpwsinc_query_type);
}
static void
process_32 (GstLPWSinc * self, gfloat * src, gfloat * dst, guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
self->residue_length += kernel_length * channels - res_start;
if (self->residue_length > kernel_length * channels)
self->residue_length = kernel_length * channels;
}
static void
process_64 (GstLPWSinc * self, gdouble * src, gdouble * dst,
guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
self->residue_length += kernel_length * channels - res_start;
if (self->residue_length > kernel_length * channels)
self->residue_length = kernel_length * channels;
}
static void
lpwsinc_build_kernel (GstLPWSinc * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
len = self->kernel_length;
if (GST_AUDIO_FILTER (self)->format.rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (GST_AUDIO_FILTER (self)->format.channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->frequency =
CLAMP (self->frequency, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
GST_DEBUG ("lpwsinc: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->frequency,
(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
/* fill the kernel */
w = 2 * M_PI * (self->frequency / GST_AUDIO_FILTER (self)->format.rate);
if (self->kernel)
g_free (self->kernel);
self->kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
self->kernel[i] = w;
else
self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
/* windowing */
if (self->window == WINDOW_HAMMING)
self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
self->kernel[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += self->kernel[i];
for (i = 0; i < len; ++i)
self->kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
self->kernel[i] = -self->kernel[i];
self->kernel[len / 2] += 1.0;
}
/* set up the residue memory space */
if (!self->residue)
self->residue =
g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->have_kernel = TRUE;
self->residue_length = 0;
}
static void
lpwsinc_push_residue (GstLPWSinc * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint outsize, outsamples;
gint diffsize, diffsamples;
guint8 *in, *out;
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = MIN (self->latency, self->residue_length / channels);
outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (outsize == 0)
return;
/* Process the difference between latency and residue_length samples
* to start at the actual data instead of starting at the zeros before
* when we only got one buffer smaller than latency */
diffsamples = self->latency - self->residue_length / channels;
diffsize =
diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (diffsize > 0) {
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->process (self, in, out, diffsamples * channels);
g_free (in);
g_free (out);
}
res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
return;
}
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
g_free (in);
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (outsamples, GST_SECOND, rate);
self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
}
GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), outsamples);
res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
lpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstLPWSinc *self = GST_LPWSINC (base);
gboolean ret = TRUE;
if (format->width == 32)
self->process = (GstLPWSincProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstLPWSincProcessFunc) process_64;
else
ret = FALSE;
self->have_kernel = FALSE;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static gboolean
lpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
g_assert (size);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
*size = width * channels / 8;
return ret;
}
static GstFlowReturn
lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstLPWSinc *self = GST_LPWSINC (base);
GstClockTime timestamp;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
gint output_samples = input_samples;
gint diff;
/* don't process data in passthrough-mode */
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
if (!self->have_kernel)
lpwsinc_build_kernel (self);
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf)) {
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
}
/* Calculate the number of samples we can push out now without outputting
* kernel_length/2 zeros in the beginning */
diff = (self->kernel_length / 2) * channels - self->residue_length;
if (diff > 0)
output_samples -= diff;
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
if (output_samples <= 0) {
/* Drop buffer and save original timestamp/offset for later use */
if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)
&& GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf);
if (self->next_off == GST_BUFFER_OFFSET_NONE
&& GST_BUFFER_OFFSET_IS_VALID (outbuf))
self->next_off = GST_BUFFER_OFFSET (outbuf);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
} else if (output_samples < input_samples) {
/* First (probably partial) buffer after starting from
* a clean residue. Use stored timestamp/offset here */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) =
self->next_off + output_samples / channels;
} else {
/* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) -= diff / channels;
}
if (GST_BUFFER_DURATION_IS_VALID (outbuf))
GST_BUFFER_DURATION (outbuf) -=
gst_util_uint64_scale (diff, GST_SECOND, channels * rate);
GST_BUFFER_DATA (outbuf) +=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
GST_BUFFER_SIZE (outbuf) -=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
} else {
GstClockTime ts_latency =
gst_util_uint64_scale (self->latency, GST_SECOND, rate);
/* Normal buffer, adjust timestamp/offset/etc by latency */
if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) {
GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency");
GST_BUFFER_TIMESTAMP (outbuf) = 0;
} else {
GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency;
}
if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET (outbuf) > self->latency) {
GST_BUFFER_OFFSET (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency");
GST_BUFFER_OFFSET (outbuf) = 0;
}
}
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) {
GST_BUFFER_OFFSET_END (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency");
GST_BUFFER_OFFSET_END (outbuf) = 0;
}
}
}
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
return GST_FLOW_OK;
}
static gboolean
lpwsinc_start (GstBaseTransform * base)
{
GstLPWSinc *self = GST_LPWSINC (base);
gint channels = GST_AUDIO_FILTER (self)->format.channels;
/* Reset the residue if already existing */
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
return TRUE;
}
static gboolean
lpwsinc_query (GstPad * pad, GstQuery * query)
{
GstLPWSinc *self = GST_LPWSINC (gst_pad_get_parent (pad));
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG_OBJECT (self, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
latency =
(rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND,
rate) : 0;
GST_DEBUG_OBJECT (self, "Our latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG_OBJECT (self, "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 (self);
return res;
}
static const GstQueryType *
lpwsinc_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static gboolean
lpwsinc_event (GstBaseTransform * base, GstEvent * event)
{
GstLPWSinc *self = GST_LPWSINC (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
lpwsinc_push_residue (self);
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
}
static void
lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
g_return_if_fail (GST_IS_LPWSINC (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
GST_BASE_TRANSFORM_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
if (self->residue) {
lpwsinc_push_residue (self);
g_free (self->residue);
self->residue = NULL;
}
self->kernel_length = val;
self->latency = val / 2;
lpwsinc_build_kernel (self);
}
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
case PROP_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->frequency = g_value_get_double (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (self);
self->mode = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_WINDOW:
GST_BASE_TRANSFORM_LOCK (self);
self->window = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
lpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_FREQUENCY:
g_value_set_double (value, self->frequency);
break;
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_WINDOW:
g_value_set_enum (value, self->window);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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