/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include "audiofxbasefirfilter.h"
#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define ALLOWED_CAPS \
"audio/x-raw, " \
" format=(string){"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ], " \
" layout=(string) interleaved"
/* Switch from time-domain to FFT convolution for kernels >= this */
#define FFT_THRESHOLD 32
enum
{
PROP_0 = 0,
PROP_LOW_LATENCY,
PROP_DRAIN_ON_CHANGES
};
#define DEFAULT_LOW_LATENCY FALSE
#define DEFAULT_DRAIN_ON_CHANGES TRUE
#define gst_audio_fx_base_fir_filter_parent_class parent_class
G_DEFINE_TYPE (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
GST_TYPE_AUDIO_FILTER);
static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
base, GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);
static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
static gboolean gst_audio_fx_base_fir_filter_sink_event (GstBaseTransform *
base, GstEvent * event);
static gboolean gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform *
base, GstPadDirection direction, GstCaps * caps, gsize size,
GstCaps * othercaps, gsize * othersize);
static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
const GstAudioInfo * info);
static gboolean gst_audio_fx_base_fir_filter_query (GstBaseTransform * trans,
GstPadDirection direction, GstQuery * quer);
/*
* The code below calculates the linear convolution:
*
* y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
*
* where y is the output, x is the input, M is the length
* of the filter kernel and h is the filter kernel. For x
* holds: x[t] == 0 \forall t < 0.
*
* The runtime complexity of this is O (M) per sample.
*
*/
#define DEFINE_PROCESS_FUNC(width,ctype) \
static guint \
process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
{ \
gint channels = GST_AUDIO_FILTER_CHANNELS (self); \
TIME_DOMAIN_CONVOLUTION_BODY (channels); \
}
#define DEFINE_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
static guint \
process_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
{ \
TIME_DOMAIN_CONVOLUTION_BODY (channels); \
}
#define TIME_DOMAIN_CONVOLUTION_BODY(channels) G_STMT_START { \
gint kernel_length = self->kernel_length; \
gint i, j, k, l; \
gint res_start; \
gint from_input; \
gint off; \
gdouble *buffer = self->buffer; \
gdouble *kernel = self->kernel; \
\
if (!buffer) { \
self->buffer_length = kernel_length * channels; \
self->buffer = buffer = g_new0 (gdouble, self->buffer_length); \
} \
\
input_samples *= channels; \
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
k = i % channels; \
l = i / channels; \
from_input = MIN (l, kernel_length-1); \
off = l * channels + k; \
for (j = 0; j <= from_input; j++) { \
dst[i] += src[off] * kernel[j]; \
off -= channels; \
} \
/* j == from_input && off == (l - j) * channels + k */ \
off += kernel_length * channels; \
for (; j < kernel_length; j++) { \
dst[i] += buffer[off] * kernel[j]; \
off -= channels; \
} \
} \
\
/* 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 */ \
/* from now on take kernel length as length over all channels */ \
kernel_length *= channels; \
if (input_samples < kernel_length) \
res_start = kernel_length - input_samples; \
else \
res_start = 0; \
\
for (i = 0; i < res_start; i++) \
buffer[i] = buffer[i + input_samples]; \
/* i == res_start */ \
for (; i < kernel_length; i++) \
buffer[i] = src[input_samples - kernel_length + i]; \
\
self->buffer_fill += kernel_length - res_start; \
if (self->buffer_fill > kernel_length) \
self->buffer_fill = kernel_length; \
\
return input_samples / channels; \
} G_STMT_END
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
#undef TIME_DOMAIN_CONVOLUTION_BODY
#undef DEFINE_PROCESS_FUNC
#undef DEFINE_PROCESS_FUNC_FIXED_CHANNELS
/* This implements FFT convolution and uses the overlap-save algorithm.
* See http://cnx.org/content/m12022/latest/ or your favorite
* digital signal processing book for details.
*
* In every pass the following is calculated:
*
* y = IFFT (FFT(x) * FFT(h))
*
* where y is the output in the time domain, x the
* input and h the filter kernel. * is the multiplication
* of complex numbers.
*
* Due to the circular convolution theorem this
* gives in the time domain:
*
* y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
*
* where y is the output, M is the kernel length,
* x the periodically extended[0] input and h the
* filter kernel.
*
* ([0] Periodically extended means: )
* ( x[t] = x[t+kN] \forall k \in Z )
* ( where N is the length of x )
*
* This means:
* - Obviously x and h need to be of the same size for the FFT
* - The first M-1 output values are useless because they're
* built from 1 up to M-1 values from the end of the input
* (circular convolusion!).
* - The last M-1 input values are only used for 1 up to M-1
* output values, i.e. they need to be used again in the
* next pass for the first M-1 input values.
*
* => The first pass needs M-1 zeroes at the beginning of the
* input and the last M-1 input values of every pass need to
* be used as the first M-1 input values of the next pass.
*
* => x must be larger than h to give a useful number of output
* samples and h needs to be padded by zeroes at the end to give
* it virtually the same size as x (by M we denote the number of
* non-padding samples of h). If len(x)==len(h)==M only 1 output
* sample would be calculated per pass, len(x)==2*len(h) would
* give M+1 output samples, etc. Usually a factor between 4 and 8
* gives a low number of operations per output samples (see website
* given above).
*
* Overall this gives a runtime complexity per sample of
*
* ( N log N )
* O ( --------- ) compared to O (M) for the direct calculation.
* ( N - M + 1 )
*/
#define DEFINE_FFT_PROCESS_FUNC(width,ctype) \
static guint \
process_fft_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
g##ctype * dst, guint input_samples) \
{ \
gint channels = GST_AUDIO_FILTER_CHANNELS (self); \
FFT_CONVOLUTION_BODY (channels); \
}
#define DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
static guint \
process_fft_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
g##ctype * dst, guint input_samples) \
{ \
FFT_CONVOLUTION_BODY (channels); \
}
#define FFT_CONVOLUTION_BODY(channels) G_STMT_START { \
gint i, j; \
guint pass; \
guint kernel_length = self->kernel_length; \
guint block_length = self->block_length; \
guint buffer_length = self->buffer_length; \
guint real_buffer_length = buffer_length + kernel_length - 1; \
guint buffer_fill = self->buffer_fill; \
GstFFTF64 *fft = self->fft; \
GstFFTF64 *ifft = self->ifft; \
GstFFTF64Complex *frequency_response = self->frequency_response; \
GstFFTF64Complex *fft_buffer = self->fft_buffer; \
guint frequency_response_length = self->frequency_response_length; \
gdouble *buffer = self->buffer; \
guint generated = 0; \
gdouble re, im; \
\
if (!fft_buffer) \
self->fft_buffer = fft_buffer = \
g_new (GstFFTF64Complex, frequency_response_length); \
\
/* Buffer contains the time domain samples of input data for one chunk \
* plus some more space for the inverse FFT below. \
* \
* The samples are put at offset kernel_length, the inverse FFT \
* overwrites everthing from offset 0 to length-kernel_length+1, keeping \
* the last kernel_length-1 samples for copying to the next processing \
* step. \
*/ \
if (!buffer) { \
self->buffer_length = buffer_length = block_length; \
real_buffer_length = buffer_length + kernel_length - 1; \
\
self->buffer = buffer = g_new0 (gdouble, real_buffer_length * channels); \
\
/* Beginning has kernel_length-1 zeroes at the beginning */ \
self->buffer_fill = buffer_fill = kernel_length - 1; \
} \
\
g_assert (self->buffer_length == block_length); \
\
while (input_samples) { \
pass = MIN (buffer_length - buffer_fill, input_samples); \
\
/* Deinterleave channels */ \
for (i = 0; i < pass; i++) { \
for (j = 0; j < channels; j++) { \
buffer[real_buffer_length * j + buffer_fill + kernel_length - 1 + i] = \
src[i * channels + j]; \
} \
} \
buffer_fill += pass; \
src += channels * pass; \
input_samples -= pass; \
\
/* If we don't have a complete buffer go out */ \
if (buffer_fill < buffer_length) \
break; \
\
for (j = 0; j < channels; j++) { \
/* Calculate FFT of input block */ \
gst_fft_f64_fft (fft, \
buffer + real_buffer_length * j + kernel_length - 1, fft_buffer); \
\
/* Complex multiplication of input and filter spectrum */ \
for (i = 0; i < frequency_response_length; i++) { \
re = fft_buffer[i].r; \
im = fft_buffer[i].i; \
\
fft_buffer[i].r = \
re * frequency_response[i].r - \
im * frequency_response[i].i; \
fft_buffer[i].i = \
re * frequency_response[i].i + \
im * frequency_response[i].r; \
} \
\
/* Calculate inverse FFT of the result */ \
gst_fft_f64_inverse_fft (ifft, fft_buffer, \
buffer + real_buffer_length * j); \
\
/* Copy all except the first kernel_length-1 samples to the output */ \
for (i = 0; i < buffer_length - kernel_length + 1; i++) { \
dst[i * channels + j] = \
buffer[real_buffer_length * j + kernel_length - 1 + i]; \
} \
\
/* Copy the last kernel_length-1 samples to the beginning for the next block */ \
for (i = 0; i < kernel_length - 1; i++) { \
buffer[real_buffer_length * j + kernel_length - 1 + i] = \
buffer[real_buffer_length * j + buffer_length + i]; \
} \
} \
\
generated += buffer_length - kernel_length + 1; \
dst += channels * (buffer_length - kernel_length + 1); \
\
/* The the first kernel_length-1 samples are there already */ \
buffer_fill = kernel_length - 1; \
} \
\
/* Write back cached buffer_fill value */ \
self->buffer_fill = buffer_fill; \
\
return generated; \
} G_STMT_END
DEFINE_FFT_PROCESS_FUNC (32, float);
DEFINE_FFT_PROCESS_FUNC (64, double);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
#undef FFT_CONVOLUTION_BODY
#undef DEFINE_FFT_PROCESS_FUNC
#undef DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS
/* Element class */
static void
gst_audio_fx_base_fir_filter_calculate_frequency_response
(GstAudioFXBaseFIRFilter * self)
{
gst_fft_f64_free (self->fft);
self->fft = NULL;
gst_fft_f64_free (self->ifft);
self->ifft = NULL;
g_free (self->frequency_response);
self->frequency_response_length = 0;
g_free (self->fft_buffer);
self->fft_buffer = NULL;
if (self->kernel && self->kernel_length >= FFT_THRESHOLD
&& !self->low_latency) {
guint block_length, i;
gdouble *kernel_tmp, *kernel = self->kernel;
/* We process 4 * kernel_length samples per pass in FFT mode */
block_length = 4 * self->kernel_length;
block_length = gst_fft_next_fast_length (block_length);
self->block_length = block_length;
kernel_tmp = g_new0 (gdouble, block_length);
memcpy (kernel_tmp, kernel, self->kernel_length * sizeof (gdouble));
self->fft = gst_fft_f64_new (block_length, FALSE);
self->ifft = gst_fft_f64_new (block_length, TRUE);
self->frequency_response_length = block_length / 2 + 1;
self->frequency_response =
g_new (GstFFTF64Complex, self->frequency_response_length);
gst_fft_f64_fft (self->fft, kernel_tmp, self->frequency_response);
g_free (kernel_tmp);
/* Normalize to make sure IFFT(FFT(x)) == x */
for (i = 0; i < self->frequency_response_length; i++) {
self->frequency_response[i].r /= block_length;
self->frequency_response[i].i /= block_length;
}
}
}
/* Must be called with base transform lock! */
static void
gst_audio_fx_base_fir_filter_select_process_function (GstAudioFXBaseFIRFilter *
self, GstAudioFormat format, gint channels)
{
switch (format) {
case GST_AUDIO_FORMAT_F32:
if (self->fft && !self->low_latency) {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_32;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_32;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
} else {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_32;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_32;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
}
break;
case GST_AUDIO_FORMAT_F64:
if (self->fft && !self->low_latency) {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_64;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_64;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
} else {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_64;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_64;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
}
break;
default:
self->process = NULL;
break;
}
}
static void
gst_audio_fx_base_fir_filter_finalize (GObject * object)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
g_free (self->buffer);
g_free (self->kernel);
gst_fft_f64_free (self->fft);
gst_fft_f64_free (self->ifft);
g_free (self->frequency_response);
g_free (self->fft_buffer);
g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_audio_fx_base_fir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
switch (prop_id) {
case PROP_LOW_LATENCY:{
gboolean low_latency;
if (GST_STATE (self) >= GST_STATE_PAUSED) {
g_warning ("Changing the \"low-latency\" property "
"is only allowed in states < PAUSED");
return;
}
g_mutex_lock (&self->lock);
low_latency = g_value_get_boolean (value);
if (self->low_latency != low_latency) {
self->low_latency = low_latency;
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_FORMAT (self), GST_AUDIO_FILTER_CHANNELS (self));
}
g_mutex_unlock (&self->lock);
break;
}
case PROP_DRAIN_ON_CHANGES:{
g_mutex_lock (&self->lock);
self->drain_on_changes = g_value_get_boolean (value);
g_mutex_unlock (&self->lock);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_fx_base_fir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
switch (prop_id) {
case PROP_LOW_LATENCY:
g_value_set_boolean (value, self->low_latency);
break;
case PROP_DRAIN_ON_CHANGES:
g_value_set_boolean (value, self->drain_on_changes);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug,
"audiofxbasefirfilter", 0, "FIR filter base class");
gobject_class->finalize = gst_audio_fx_base_fir_filter_finalize;
gobject_class->set_property = gst_audio_fx_base_fir_filter_set_property;
gobject_class->get_property = gst_audio_fx_base_fir_filter_get_property;
/**
* GstAudioFXBaseFIRFilter:low-latency:
*
* Work in low-latency mode. This mode is much slower for large filter sizes
* but the latency is always only the pre-latency of the filter.
*/
g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
g_param_spec_boolean ("low-latency", "Low latency",
"Operate in low latency mode. This mode is slower but the "
"latency will only be the filter pre-latency. "
"Can only be changed in states < PAUSED!", DEFAULT_LOW_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioFXBaseFIRFilter:drain-on-changes:
*
* Whether the filter should be drained when its coeficients change
*
* Note: Currently this only works if the kernel size is not changed!
* Support for drainless kernel size changes will be added in the future.
*/
g_object_class_install_property (gobject_class, PROP_DRAIN_ON_CHANGES,
g_param_spec_boolean ("drain-on-changes", "Drain on changes",
"Drains the filter when its coeficients change",
DEFAULT_DRAIN_ON_CHANGES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
trans_class->transform =
GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);
trans_class->sink_event =
GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_sink_event);
trans_class->query = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_query);
trans_class->transform_size =
GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform_size);
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
}
static void
gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self)
{
self->kernel = NULL;
self->buffer = NULL;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
self->low_latency = DEFAULT_LOW_LATENCY;
self->drain_on_changes = DEFAULT_DRAIN_ON_CHANGES;
g_mutex_init (&self->lock);
}
void
gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER_RATE (self);
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
gint outsize, outsamples;
GstMapInfo map;
guint8 *in, *out;
if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = self->nsamples_in - (self->nsamples_out - self->latency);
if (outsamples <= 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
outsize = outsamples * channels * bps;
if (!self->fft || self->low_latency) {
gint64 diffsize, diffsamples;
/* 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 =
((gint64) self->latency) - ((gint64) self->buffer_fill) / channels;
if (diffsamples > 0) {
diffsize = diffsamples * channels * bps;
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->nsamples_out += self->process (self, in, out, diffsamples);
g_free (in);
g_free (out);
}
outbuf = gst_buffer_new_and_alloc (outsize);
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
self->nsamples_out += self->process (self, in, map.data, outsamples);
gst_buffer_unmap (outbuf, &map);
g_free (in);
} else {
guint gensamples = 0;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
while (gensamples < outsamples) {
guint step_insamples = self->block_length - self->buffer_fill;
guint8 *zeroes = g_new0 (guint8, step_insamples * channels * bps);
guint8 *out = g_new (guint8, self->block_length * channels * bps);
guint step_gensamples;
step_gensamples = self->process (self, zeroes, out, step_insamples);
g_free (zeroes);
memcpy (map.data + gensamples * bps, out, MIN (step_gensamples,
outsamples - gensamples) * bps);
gensamples += MIN (step_gensamples, outsamples - gensamples);
g_free (out);
}
self->nsamples_out += gensamples;
gst_buffer_unmap (outbuf, &map);
}
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_TIMESTAMP (outbuf) +=
gst_util_uint64_scale_int (self->nsamples_out - outsamples -
self->latency, GST_SECOND, rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) =
self->start_off + self->nsamples_out - outsamples - self->latency;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
}
GST_DEBUG_OBJECT (self,
"Pushing residue buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
gst_buffer_get_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_CAST (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
self->buffer_fill = 0;
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
const GstAudioInfo * info)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
g_mutex_lock (&self->lock);
if (self->buffer) {
gst_audio_fx_base_fir_filter_push_residue (self);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
}
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info));
g_mutex_unlock (&self->lock);
return (self->process != NULL);
}
/* GstBaseTransform vmethod implementations */
static gboolean
gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
gsize * othersize)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
guint blocklen;
GstAudioInfo info;
gint bpf;
if (!self->fft || self->low_latency || direction == GST_PAD_SRC) {
*othersize = size;
return TRUE;
}
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
bpf = GST_AUDIO_INFO_BPF (&info);
size /= bpf;
blocklen = self->block_length - self->kernel_length + 1;
*othersize = ((size + blocklen - 1) / blocklen) * blocklen;
*othersize *= bpf;
return TRUE;
}
static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
GstClockTime timestamp, expected_timestamp;
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint rate = GST_AUDIO_FILTER_RATE (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
GstMapInfo inmap, outmap;
guint input_samples;
guint output_samples;
guint generated_samples;
guint64 output_offset;
gint64 diff = 0;
GstClockTime stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (!GST_CLOCK_TIME_IS_VALID (timestamp)
&& !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
GST_ERROR_OBJECT (self, "Invalid timestamp");
return GST_FLOW_ERROR;
}
g_mutex_lock (&self->lock);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (self), stream_time);
g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
expected_timestamp =
self->start_ts + gst_util_uint64_scale_int (self->nsamples_in,
GST_SECOND, rate);
else
expected_timestamp = GST_CLOCK_TIME_NONE;
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf)
|| (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
&& (ABS (GST_CLOCK_DIFF (timestamp,
expected_timestamp)) > 5 * GST_MSECOND))) {
GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
gst_audio_fx_base_fir_filter_push_residue (self);
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
self->nsamples_out = 0;
self->nsamples_in = 0;
} else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
}
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
input_samples = (inmap.size / bps) / channels;
output_samples = (outmap.size / bps) / channels;
self->nsamples_in += input_samples;
generated_samples =
self->process (self, inmap.data, outmap.data, input_samples);
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
g_assert (generated_samples <= output_samples);
self->nsamples_out += generated_samples;
if (generated_samples == 0)
goto no_samples;
/* Calculate the number of samples we can push out now without outputting
* latency zeros in the beginning */
diff = ((gint64) self->nsamples_out) - ((gint64) self->latency);
if (diff < 0)
goto no_samples;
if (diff < generated_samples) {
gint64 tmp = diff;
diff = generated_samples - diff;
generated_samples = tmp;
} else {
diff = 0;
}
gst_buffer_resize (outbuf, diff * bps * channels,
generated_samples * bps * channels);
output_offset = self->nsamples_out - self->latency - generated_samples;
GST_BUFFER_TIMESTAMP (outbuf) =
self->start_ts + gst_util_uint64_scale_int (output_offset, GST_SECOND,
rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (output_samples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + output_offset;
GST_BUFFER_OFFSET_END (outbuf) =
GST_BUFFER_OFFSET (outbuf) + generated_samples;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (self,
"Pushing buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
gst_buffer_get_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), generated_samples);
return GST_FLOW_OK;
no_samples:
{
g_mutex_unlock (&self->lock);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
static gboolean
gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
return TRUE;
}
static gboolean
gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_length = 0;
return TRUE;
}
static gboolean
gst_audio_fx_base_fir_filter_query (GstBaseTransform * trans,
GstPadDirection direction, GstQuery * query)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (trans);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
gint rate = GST_AUDIO_FILTER_RATE (self);
if (rate == 0) {
res = FALSE;
} else if ((res =
gst_pad_peer_query (GST_BASE_TRANSFORM (self)->sinkpad, 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));
if (self->fft && !self->low_latency)
latency = self->block_length - self->kernel_length + 1;
else
latency = self->latency;
/* add our own latency */
latency = gst_util_uint64_scale_round (latency, GST_SECOND, rate);
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);
}
break;
}
default:
res =
GST_BASE_TRANSFORM_CLASS (parent_class)->query (trans, direction,
query);
break;
}
return res;
}
static gboolean
gst_audio_fx_base_fir_filter_sink_event (GstBaseTransform * base,
GstEvent * event)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_audio_fx_base_fir_filter_push_residue (self);
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event);
}
void
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gdouble * kernel, guint kernel_length, guint64 latency,
const GstAudioInfo * info)
{
gboolean latency_changed;
GstAudioFormat format;
gint channels;
g_return_if_fail (kernel != NULL);
g_return_if_fail (self != NULL);
g_mutex_lock (&self->lock);
latency_changed = (self->latency != latency
|| (!self->low_latency && self->kernel_length < FFT_THRESHOLD
&& kernel_length >= FFT_THRESHOLD)
|| (!self->low_latency && self->kernel_length >= FFT_THRESHOLD
&& kernel_length < FFT_THRESHOLD));
/* FIXME: If the latency changes, the buffer size changes too and we
* have to drain in any case until this is fixed in the future */
if (self->buffer && (!self->drain_on_changes || latency_changed)) {
gst_audio_fx_base_fir_filter_push_residue (self);
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
self->buffer_fill = 0;
}
g_free (self->kernel);
if (!self->drain_on_changes || latency_changed) {
g_free (self->buffer);
self->buffer = NULL;
self->buffer_fill = 0;
self->buffer_length = 0;
}
self->kernel = kernel;
self->kernel_length = kernel_length;
if (info) {
format = GST_AUDIO_INFO_FORMAT (info);
channels = GST_AUDIO_INFO_CHANNELS (info);
} else {
format = GST_AUDIO_FILTER_FORMAT (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
}
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self, format, channels);
if (latency_changed) {
self->latency = latency;
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
g_mutex_unlock (&self->lock);
}