1 /* -*- c-basic-offset: 2 -*-
4 * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
5 * 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
6 * 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Library General Public License for more details.
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
21 * Boston, MA 02111-1307, USA.
32 #include <gst/audio/gstaudiofilter.h>
34 #include "audiofxbasefirfilter.h"
36 #define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
37 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
39 #define ALLOWED_CAPS \
41 " format=(string){"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \
42 " rate = (int) [ 1, MAX ], " \
43 " channels = (int) [ 1, MAX ], " \
44 " layout=(string) interleaved"
46 /* Switch from time-domain to FFT convolution for kernels >= this */
47 #define FFT_THRESHOLD 32
56 #define DEFAULT_LOW_LATENCY FALSE
57 #define DEFAULT_DRAIN_ON_CHANGES TRUE
59 #define gst_audio_fx_base_fir_filter_parent_class parent_class
60 G_DEFINE_TYPE (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
61 GST_TYPE_AUDIO_FILTER);
63 static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
64 base, GstBuffer * inbuf, GstBuffer * outbuf);
65 static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);
66 static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
67 static gboolean gst_audio_fx_base_fir_filter_sink_event (GstBaseTransform *
68 base, GstEvent * event);
69 static gboolean gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform *
70 base, GstPadDirection direction, GstCaps * caps, gsize size,
71 GstCaps * othercaps, gsize * othersize);
72 static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
73 const GstAudioInfo * info);
75 static gboolean gst_audio_fx_base_fir_filter_query (GstBaseTransform * trans,
76 GstPadDirection direction, GstQuery * quer);
79 * The code below calculates the linear convolution:
81 * y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
83 * where y is the output, x is the input, M is the length
84 * of the filter kernel and h is the filter kernel. For x
85 * holds: x[t] == 0 \forall t < 0.
87 * The runtime complexity of this is O (M) per sample.
90 #define DEFINE_PROCESS_FUNC(width,ctype) \
92 process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
94 gint channels = GST_AUDIO_FILTER_CHANNELS (self); \
95 TIME_DOMAIN_CONVOLUTION_BODY (channels); \
98 #define DEFINE_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
100 process_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
102 TIME_DOMAIN_CONVOLUTION_BODY (channels); \
105 #define TIME_DOMAIN_CONVOLUTION_BODY(channels) G_STMT_START { \
106 gint kernel_length = self->kernel_length; \
111 gdouble *buffer = self->buffer; \
112 gdouble *kernel = self->kernel; \
113 guint buffer_length = self->buffer_length; \
116 self->buffer_length = buffer_length = kernel_length * channels; \
117 self->buffer = buffer = g_new0 (gdouble, self->buffer_length); \
121 for (i = 0; i < input_samples; i++) { \
125 from_input = MIN (l, kernel_length-1); \
126 off = l * channels + k; \
127 for (j = 0; j <= from_input; j++) { \
128 dst[i] += src[off] * kernel[j]; \
131 /* j == from_input && off == (l - j) * channels + k */ \
132 off += kernel_length * channels; \
133 for (; j < kernel_length; j++) { \
134 dst[i] += buffer[off] * kernel[j]; \
139 /* copy the tail of the current input buffer to the residue, while \
140 * keeping parts of the residue if the input buffer is smaller than \
141 * the kernel length */ \
142 /* from now on take kernel length as length over all channels */ \
143 kernel_length *= channels; \
144 if (input_samples < kernel_length) \
145 res_start = kernel_length - input_samples; \
149 for (i = 0; i < res_start; i++) \
150 buffer[i] = buffer[i + input_samples]; \
151 /* i == res_start */ \
152 for (; i < kernel_length; i++) \
153 buffer[i] = src[input_samples - kernel_length + i]; \
155 self->buffer_fill += kernel_length - res_start; \
156 if (self->buffer_fill > kernel_length) \
157 self->buffer_fill = kernel_length; \
159 return input_samples / channels; \
162 DEFINE_PROCESS_FUNC (32, float);
163 DEFINE_PROCESS_FUNC (64, double);
165 DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
166 DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
168 DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
169 DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
171 #undef TIME_DOMAIN_CONVOLUTION_BODY
172 #undef DEFINE_PROCESS_FUNC
173 #undef DEFINE_PROCESS_FUNC_FIXED_CHANNELS
175 /* This implements FFT convolution and uses the overlap-save algorithm.
176 * See http://cnx.org/content/m12022/latest/ or your favorite
177 * digital signal processing book for details.
179 * In every pass the following is calculated:
181 * y = IFFT (FFT(x) * FFT(h))
183 * where y is the output in the time domain, x the
184 * input and h the filter kernel. * is the multiplication
185 * of complex numbers.
187 * Due to the circular convolution theorem this
188 * gives in the time domain:
190 * y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
192 * where y is the output, M is the kernel length,
193 * x the periodically extended[0] input and h the
196 * ([0] Periodically extended means: )
197 * ( x[t] = x[t+kN] \forall k \in Z )
198 * ( where N is the length of x )
201 * - Obviously x and h need to be of the same size for the FFT
202 * - The first M-1 output values are useless because they're
203 * built from 1 up to M-1 values from the end of the input
204 * (circular convolusion!).
205 * - The last M-1 input values are only used for 1 up to M-1
206 * output values, i.e. they need to be used again in the
207 * next pass for the first M-1 input values.
209 * => The first pass needs M-1 zeroes at the beginning of the
210 * input and the last M-1 input values of every pass need to
211 * be used as the first M-1 input values of the next pass.
213 * => x must be larger than h to give a useful number of output
214 * samples and h needs to be padded by zeroes at the end to give
215 * it virtually the same size as x (by M we denote the number of
216 * non-padding samples of h). If len(x)==len(h)==M only 1 output
217 * sample would be calculated per pass, len(x)==2*len(h) would
218 * give M+1 output samples, etc. Usually a factor between 4 and 8
219 * gives a low number of operations per output samples (see website
222 * Overall this gives a runtime complexity per sample of
225 * O ( --------- ) compared to O (M) for the direct calculation.
228 #define DEFINE_FFT_PROCESS_FUNC(width,ctype) \
230 process_fft_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
231 g##ctype * dst, guint input_samples) \
233 gint channels = GST_AUDIO_FILTER_CHANNELS (self); \
234 FFT_CONVOLUTION_BODY (channels); \
237 #define DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
239 process_fft_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
240 g##ctype * dst, guint input_samples) \
242 FFT_CONVOLUTION_BODY (channels); \
245 #define FFT_CONVOLUTION_BODY(channels) G_STMT_START { \
248 guint kernel_length = self->kernel_length; \
249 guint block_length = self->block_length; \
250 guint buffer_length = self->buffer_length; \
251 guint real_buffer_length = buffer_length + kernel_length - 1; \
252 guint buffer_fill = self->buffer_fill; \
253 GstFFTF64 *fft = self->fft; \
254 GstFFTF64 *ifft = self->ifft; \
255 GstFFTF64Complex *frequency_response = self->frequency_response; \
256 GstFFTF64Complex *fft_buffer = self->fft_buffer; \
257 guint frequency_response_length = self->frequency_response_length; \
258 gdouble *buffer = self->buffer; \
259 guint generated = 0; \
263 self->fft_buffer = fft_buffer = \
264 g_new (GstFFTF64Complex, frequency_response_length); \
266 /* Buffer contains the time domain samples of input data for one chunk \
267 * plus some more space for the inverse FFT below. \
269 * The samples are put at offset kernel_length, the inverse FFT \
270 * overwrites everthing from offset 0 to length-kernel_length+1, keeping \
271 * the last kernel_length-1 samples for copying to the next processing \
275 self->buffer_length = buffer_length = block_length; \
276 real_buffer_length = buffer_length + kernel_length - 1; \
278 self->buffer = buffer = g_new0 (gdouble, real_buffer_length * channels); \
280 /* Beginning has kernel_length-1 zeroes at the beginning */ \
281 self->buffer_fill = buffer_fill = kernel_length - 1; \
284 g_assert (self->buffer_length == block_length); \
286 while (input_samples) { \
287 pass = MIN (buffer_length - buffer_fill, input_samples); \
289 /* Deinterleave channels */ \
290 for (i = 0; i < pass; i++) { \
291 for (j = 0; j < channels; j++) { \
292 buffer[real_buffer_length * j + buffer_fill + kernel_length - 1 + i] = \
293 src[i * channels + j]; \
296 buffer_fill += pass; \
297 src += channels * pass; \
298 input_samples -= pass; \
300 /* If we don't have a complete buffer go out */ \
301 if (buffer_fill < buffer_length) \
304 for (j = 0; j < channels; j++) { \
305 /* Calculate FFT of input block */ \
306 gst_fft_f64_fft (fft, \
307 buffer + real_buffer_length * j + kernel_length - 1, fft_buffer); \
309 /* Complex multiplication of input and filter spectrum */ \
310 for (i = 0; i < frequency_response_length; i++) { \
311 re = fft_buffer[i].r; \
312 im = fft_buffer[i].i; \
315 re * frequency_response[i].r - \
316 im * frequency_response[i].i; \
318 re * frequency_response[i].i + \
319 im * frequency_response[i].r; \
322 /* Calculate inverse FFT of the result */ \
323 gst_fft_f64_inverse_fft (ifft, fft_buffer, \
324 buffer + real_buffer_length * j); \
326 /* Copy all except the first kernel_length-1 samples to the output */ \
327 for (i = 0; i < buffer_length - kernel_length + 1; i++) { \
328 dst[i * channels + j] = \
329 buffer[real_buffer_length * j + kernel_length - 1 + i]; \
332 /* Copy the last kernel_length-1 samples to the beginning for the next block */ \
333 for (i = 0; i < kernel_length - 1; i++) { \
334 buffer[real_buffer_length * j + kernel_length - 1 + i] = \
335 buffer[real_buffer_length * j + buffer_length + i]; \
339 generated += buffer_length - kernel_length + 1; \
340 dst += channels * (buffer_length - kernel_length + 1); \
342 /* The the first kernel_length-1 samples are there already */ \
343 buffer_fill = kernel_length - 1; \
346 /* Write back cached buffer_fill value */ \
347 self->buffer_fill = buffer_fill; \
352 DEFINE_FFT_PROCESS_FUNC (32, float);
353 DEFINE_FFT_PROCESS_FUNC (64, double);
355 DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
356 DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
358 DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
359 DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
361 #undef FFT_CONVOLUTION_BODY
362 #undef DEFINE_FFT_PROCESS_FUNC
363 #undef DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS
367 gst_audio_fx_base_fir_filter_calculate_frequency_response
368 (GstAudioFXBaseFIRFilter * self)
370 gst_fft_f64_free (self->fft);
372 gst_fft_f64_free (self->ifft);
374 g_free (self->frequency_response);
375 self->frequency_response_length = 0;
376 g_free (self->fft_buffer);
377 self->fft_buffer = NULL;
379 if (self->kernel && self->kernel_length >= FFT_THRESHOLD
380 && !self->low_latency) {
381 guint block_length, i;
382 gdouble *kernel_tmp, *kernel = self->kernel;
384 /* We process 4 * kernel_length samples per pass in FFT mode */
385 block_length = 4 * self->kernel_length;
386 block_length = gst_fft_next_fast_length (block_length);
387 self->block_length = block_length;
389 kernel_tmp = g_new0 (gdouble, block_length);
390 memcpy (kernel_tmp, kernel, self->kernel_length * sizeof (gdouble));
392 self->fft = gst_fft_f64_new (block_length, FALSE);
393 self->ifft = gst_fft_f64_new (block_length, TRUE);
394 self->frequency_response_length = block_length / 2 + 1;
395 self->frequency_response =
396 g_new (GstFFTF64Complex, self->frequency_response_length);
397 gst_fft_f64_fft (self->fft, kernel_tmp, self->frequency_response);
400 /* Normalize to make sure IFFT(FFT(x)) == x */
401 for (i = 0; i < self->frequency_response_length; i++) {
402 self->frequency_response[i].r /= block_length;
403 self->frequency_response[i].i /= block_length;
408 /* Must be called with base transform lock! */
410 gst_audio_fx_base_fir_filter_select_process_function (GstAudioFXBaseFIRFilter *
411 self, GstAudioFormat format, gint channels)
414 case GST_AUDIO_FORMAT_F32:
415 if (self->fft && !self->low_latency) {
417 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_32;
418 else if (channels == 2)
419 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_32;
421 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
424 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_32;
425 else if (channels == 2)
426 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_32;
428 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
431 case GST_AUDIO_FORMAT_F64:
432 if (self->fft && !self->low_latency) {
434 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_64;
435 else if (channels == 2)
436 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_64;
438 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
441 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_64;
442 else if (channels == 2)
443 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_64;
445 self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
449 self->process = NULL;
455 gst_audio_fx_base_fir_filter_finalize (GObject * object)
457 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
459 g_free (self->buffer);
460 g_free (self->kernel);
461 gst_fft_f64_free (self->fft);
462 gst_fft_f64_free (self->ifft);
463 g_free (self->frequency_response);
464 g_free (self->fft_buffer);
465 g_mutex_clear (&self->lock);
467 G_OBJECT_CLASS (parent_class)->finalize (object);
471 gst_audio_fx_base_fir_filter_set_property (GObject * object, guint prop_id,
472 const GValue * value, GParamSpec * pspec)
474 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
477 case PROP_LOW_LATENCY:{
478 gboolean low_latency;
480 if (GST_STATE (self) >= GST_STATE_PAUSED) {
481 g_warning ("Changing the \"low-latency\" property "
482 "is only allowed in states < PAUSED");
487 g_mutex_lock (&self->lock);
488 low_latency = g_value_get_boolean (value);
490 if (self->low_latency != low_latency) {
491 self->low_latency = low_latency;
492 gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
493 gst_audio_fx_base_fir_filter_select_process_function (self,
494 GST_AUDIO_FILTER_FORMAT (self), GST_AUDIO_FILTER_CHANNELS (self));
496 g_mutex_unlock (&self->lock);
499 case PROP_DRAIN_ON_CHANGES:{
500 g_mutex_lock (&self->lock);
501 self->drain_on_changes = g_value_get_boolean (value);
502 g_mutex_unlock (&self->lock);
506 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
512 gst_audio_fx_base_fir_filter_get_property (GObject * object, guint prop_id,
513 GValue * value, GParamSpec * pspec)
515 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
518 case PROP_LOW_LATENCY:
519 g_value_set_boolean (value, self->low_latency);
521 case PROP_DRAIN_ON_CHANGES:
522 g_value_set_boolean (value, self->drain_on_changes);
525 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
531 gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
533 GObjectClass *gobject_class = (GObjectClass *) klass;
534 GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
535 GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
538 GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug,
539 "audiofxbasefirfilter", 0, "FIR filter base class");
541 gobject_class->finalize = gst_audio_fx_base_fir_filter_finalize;
542 gobject_class->set_property = gst_audio_fx_base_fir_filter_set_property;
543 gobject_class->get_property = gst_audio_fx_base_fir_filter_get_property;
546 * GstAudioFXBaseFIRFilter::low-latency:
548 * Work in low-latency mode. This mode is much slower for large filter sizes
549 * but the latency is always only the pre-latency of the filter.
553 g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
554 g_param_spec_boolean ("low-latency", "Low latency",
555 "Operate in low latency mode. This mode is slower but the "
556 "latency will only be the filter pre-latency. "
557 "Can only be changed in states < PAUSED!", DEFAULT_LOW_LATENCY,
558 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
561 * GstAudioFXBaseFIRFilter::drain-on-changes:
563 * Whether the filter should be drained when its coeficients change
565 * Note: Currently this only works if the kernel size is not changed!
566 * Support for drainless kernel size changes will be added in the future.
570 g_object_class_install_property (gobject_class, PROP_DRAIN_ON_CHANGES,
571 g_param_spec_boolean ("drain-on-changes", "Drain on changes",
572 "Drains the filter when its coeficients change",
573 DEFAULT_DRAIN_ON_CHANGES,
574 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
576 caps = gst_caps_from_string (ALLOWED_CAPS);
577 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
579 gst_caps_unref (caps);
581 trans_class->transform =
582 GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
583 trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
584 trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);
585 trans_class->sink_event =
586 GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_sink_event);
587 trans_class->query = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_query);
588 trans_class->transform_size =
589 GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform_size);
590 filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
594 gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self)
598 self->buffer_length = 0;
600 self->start_ts = GST_CLOCK_TIME_NONE;
601 self->start_off = GST_BUFFER_OFFSET_NONE;
602 self->nsamples_out = 0;
603 self->nsamples_in = 0;
605 self->low_latency = DEFAULT_LOW_LATENCY;
606 self->drain_on_changes = DEFAULT_DRAIN_ON_CHANGES;
608 g_mutex_init (&self->lock);
612 gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
616 gint rate = GST_AUDIO_FILTER_RATE (self);
617 gint channels = GST_AUDIO_FILTER_CHANNELS (self);
618 gint bps = GST_AUDIO_FILTER_BPS (self);
619 gint outsize, outsamples;
623 if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
624 self->buffer_fill = 0;
625 g_free (self->buffer);
630 /* Calculate the number of samples and their memory size that
631 * should be pushed from the residue */
632 outsamples = self->nsamples_in - (self->nsamples_out - self->latency);
633 if (outsamples <= 0) {
634 self->buffer_fill = 0;
635 g_free (self->buffer);
639 outsize = outsamples * channels * bps;
641 if (!self->fft || self->low_latency) {
642 gint64 diffsize, diffsamples;
644 /* Process the difference between latency and residue length samples
645 * to start at the actual data instead of starting at the zeros before
646 * when we only got one buffer smaller than latency */
648 ((gint64) self->latency) - ((gint64) self->buffer_fill) / channels;
649 if (diffsamples > 0) {
650 diffsize = diffsamples * channels * bps;
651 in = g_new0 (guint8, diffsize);
652 out = g_new0 (guint8, diffsize);
653 self->nsamples_out += self->process (self, in, out, diffsamples);
658 outbuf = gst_buffer_new_and_alloc (outsize);
660 /* Convolve the residue with zeros to get the actual remaining data */
661 in = g_new0 (guint8, outsize);
662 gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
663 self->nsamples_out += self->process (self, in, map.data, outsamples);
664 gst_buffer_unmap (outbuf, &map);
668 guint gensamples = 0;
670 outbuf = gst_buffer_new_and_alloc (outsize);
671 gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
673 while (gensamples < outsamples) {
674 guint step_insamples = self->block_length - self->buffer_fill;
675 guint8 *zeroes = g_new0 (guint8, step_insamples * channels * bps);
676 guint8 *out = g_new (guint8, self->block_length * channels * bps);
677 guint step_gensamples;
679 step_gensamples = self->process (self, zeroes, out, step_insamples);
682 memcpy (map.data + gensamples * bps, out, MIN (step_gensamples,
683 outsamples - gensamples) * bps);
684 gensamples += MIN (step_gensamples, outsamples - gensamples);
688 self->nsamples_out += gensamples;
690 gst_buffer_unmap (outbuf, &map);
693 /* Set timestamp, offset, etc from the values we
694 * saved when processing the regular buffers */
695 if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
696 GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
698 GST_BUFFER_TIMESTAMP (outbuf) = 0;
699 GST_BUFFER_TIMESTAMP (outbuf) +=
700 gst_util_uint64_scale_int (self->nsamples_out - outsamples -
701 self->latency, GST_SECOND, rate);
703 GST_BUFFER_DURATION (outbuf) =
704 gst_util_uint64_scale_int (outsamples, GST_SECOND, rate);
706 if (self->start_off != GST_BUFFER_OFFSET_NONE) {
707 GST_BUFFER_OFFSET (outbuf) =
708 self->start_off + self->nsamples_out - outsamples - self->latency;
709 GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
712 GST_DEBUG_OBJECT (self,
713 "Pushing residue buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
714 GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
715 G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
716 gst_buffer_get_size (outbuf),
717 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
718 GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
719 GST_BUFFER_OFFSET_END (outbuf), outsamples);
721 res = gst_pad_push (GST_BASE_TRANSFORM_CAST (self)->srcpad, outbuf);
723 if (G_UNLIKELY (res != GST_FLOW_OK)) {
724 GST_WARNING_OBJECT (self, "failed to push residue");
727 self->buffer_fill = 0;
730 /* GstAudioFilter vmethod implementations */
732 /* get notified of caps and plug in the correct process function */
734 gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
735 const GstAudioInfo * info)
737 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
739 g_mutex_lock (&self->lock);
741 gst_audio_fx_base_fir_filter_push_residue (self);
742 g_free (self->buffer);
744 self->buffer_fill = 0;
745 self->buffer_length = 0;
746 self->start_ts = GST_CLOCK_TIME_NONE;
747 self->start_off = GST_BUFFER_OFFSET_NONE;
748 self->nsamples_out = 0;
749 self->nsamples_in = 0;
752 gst_audio_fx_base_fir_filter_select_process_function (self,
753 GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info));
754 g_mutex_unlock (&self->lock);
756 return (self->process != NULL);
759 /* GstBaseTransform vmethod implementations */
762 gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform * base,
763 GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
766 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
771 if (!self->fft || self->low_latency || direction == GST_PAD_SRC) {
776 if (!gst_audio_info_from_caps (&info, caps))
779 bpf = GST_AUDIO_INFO_BPF (&info);
782 blocklen = self->block_length - self->kernel_length + 1;
783 *othersize = ((size + blocklen - 1) / blocklen) * blocklen;
790 gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
791 GstBuffer * inbuf, GstBuffer * outbuf)
793 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
794 GstClockTime timestamp, expected_timestamp;
795 gint channels = GST_AUDIO_FILTER_CHANNELS (self);
796 gint rate = GST_AUDIO_FILTER_RATE (self);
797 gint bps = GST_AUDIO_FILTER_BPS (self);
798 GstMapInfo inmap, outmap;
800 guint output_samples;
801 guint generated_samples;
802 guint64 output_offset;
804 GstClockTime stream_time;
806 timestamp = GST_BUFFER_TIMESTAMP (outbuf);
808 if (!GST_CLOCK_TIME_IS_VALID (timestamp)
809 && !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
810 GST_ERROR_OBJECT (self, "Invalid timestamp");
811 return GST_FLOW_ERROR;
814 g_mutex_lock (&self->lock);
816 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
818 GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
819 GST_TIME_ARGS (timestamp));
821 if (GST_CLOCK_TIME_IS_VALID (stream_time))
822 gst_object_sync_values (GST_OBJECT (self), stream_time);
824 g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
825 g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
827 if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
829 self->start_ts + gst_util_uint64_scale_int (self->nsamples_in,
832 expected_timestamp = GST_CLOCK_TIME_NONE;
834 /* Reset the residue if already existing on discont buffers */
835 if (GST_BUFFER_IS_DISCONT (inbuf)
836 || (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
837 && (ABS (GST_CLOCK_DIFF (timestamp,
838 expected_timestamp) > 5 * GST_MSECOND)))) {
839 GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
840 if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
841 gst_audio_fx_base_fir_filter_push_residue (self);
842 self->buffer_fill = 0;
843 g_free (self->buffer);
845 self->start_ts = timestamp;
846 self->start_off = GST_BUFFER_OFFSET (inbuf);
847 self->nsamples_out = 0;
848 self->nsamples_in = 0;
849 } else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
850 self->start_ts = timestamp;
851 self->start_off = GST_BUFFER_OFFSET (inbuf);
854 gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
855 gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
857 input_samples = (inmap.size / bps) / channels;
858 output_samples = (outmap.size / bps) / channels;
860 self->nsamples_in += input_samples;
863 self->process (self, inmap.data, outmap.data, input_samples);
865 gst_buffer_unmap (inbuf, &inmap);
866 gst_buffer_unmap (outbuf, &outmap);
868 g_assert (generated_samples <= output_samples);
869 self->nsamples_out += generated_samples;
870 if (generated_samples == 0)
873 /* Calculate the number of samples we can push out now without outputting
874 * latency zeros in the beginning */
875 diff = ((gint64) self->nsamples_out) - ((gint64) self->latency);
879 if (diff < generated_samples) {
881 diff = generated_samples - diff;
882 generated_samples = tmp;
885 gst_buffer_resize (outbuf, diff * bps * channels,
886 generated_samples * bps * channels);
888 output_offset = self->nsamples_out - self->latency - generated_samples;
889 GST_BUFFER_TIMESTAMP (outbuf) =
890 self->start_ts + gst_util_uint64_scale_int (output_offset, GST_SECOND,
892 GST_BUFFER_DURATION (outbuf) =
893 gst_util_uint64_scale_int (output_samples, GST_SECOND, rate);
894 if (self->start_off != GST_BUFFER_OFFSET_NONE) {
895 GST_BUFFER_OFFSET (outbuf) = self->start_off + output_offset;
896 GST_BUFFER_OFFSET_END (outbuf) =
897 GST_BUFFER_OFFSET (outbuf) + generated_samples;
899 GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
900 GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
902 g_mutex_unlock (&self->lock);
904 GST_DEBUG_OBJECT (self,
905 "Pushing buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
906 GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
907 G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
908 gst_buffer_get_size (outbuf),
909 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
910 GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
911 GST_BUFFER_OFFSET_END (outbuf), generated_samples);
917 g_mutex_unlock (&self->lock);
918 return GST_BASE_TRANSFORM_FLOW_DROPPED;
923 gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
925 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
927 self->buffer_fill = 0;
928 g_free (self->buffer);
930 self->start_ts = GST_CLOCK_TIME_NONE;
931 self->start_off = GST_BUFFER_OFFSET_NONE;
932 self->nsamples_out = 0;
933 self->nsamples_in = 0;
939 gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
941 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
943 g_free (self->buffer);
945 self->buffer_length = 0;
951 gst_audio_fx_base_fir_filter_query (GstBaseTransform * trans,
952 GstPadDirection direction, GstQuery * query)
954 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (trans);
957 switch (GST_QUERY_TYPE (query)) {
958 case GST_QUERY_LATENCY:
960 GstClockTime min, max;
963 gint rate = GST_AUDIO_FILTER_RATE (self);
968 gst_pad_peer_query (GST_BASE_TRANSFORM (self)->sinkpad, query))) {
969 gst_query_parse_latency (query, &live, &min, &max);
971 GST_DEBUG_OBJECT (self, "Peer latency: min %"
972 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
973 GST_TIME_ARGS (min), GST_TIME_ARGS (max));
975 if (self->fft && !self->low_latency)
976 latency = self->block_length - self->kernel_length + 1;
978 latency = self->latency;
980 /* add our own latency */
981 latency = gst_util_uint64_scale_round (latency, GST_SECOND, rate);
983 GST_DEBUG_OBJECT (self, "Our latency: %"
984 GST_TIME_FORMAT, GST_TIME_ARGS (latency));
987 if (max != GST_CLOCK_TIME_NONE)
990 GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
991 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
992 GST_TIME_ARGS (min), GST_TIME_ARGS (max));
994 gst_query_set_latency (query, live, min, max);
1000 GST_BASE_TRANSFORM_CLASS (parent_class)->query (trans, direction,
1008 gst_audio_fx_base_fir_filter_sink_event (GstBaseTransform * base,
1011 GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
1013 switch (GST_EVENT_TYPE (event)) {
1015 gst_audio_fx_base_fir_filter_push_residue (self);
1016 self->start_ts = GST_CLOCK_TIME_NONE;
1017 self->start_off = GST_BUFFER_OFFSET_NONE;
1018 self->nsamples_out = 0;
1019 self->nsamples_in = 0;
1025 return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event);
1029 gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
1030 gdouble * kernel, guint kernel_length, guint64 latency)
1032 gboolean latency_changed;
1034 g_return_if_fail (kernel != NULL);
1035 g_return_if_fail (self != NULL);
1037 g_mutex_lock (&self->lock);
1039 latency_changed = (self->latency != latency
1040 || (!self->low_latency && self->kernel_length < FFT_THRESHOLD
1041 && kernel_length >= FFT_THRESHOLD)
1042 || (!self->low_latency && self->kernel_length >= FFT_THRESHOLD
1043 && kernel_length < FFT_THRESHOLD));
1045 /* FIXME: If the latency changes, the buffer size changes too and we
1046 * have to drain in any case until this is fixed in the future */
1047 if (self->buffer && (!self->drain_on_changes || latency_changed)) {
1048 gst_audio_fx_base_fir_filter_push_residue (self);
1049 self->start_ts = GST_CLOCK_TIME_NONE;
1050 self->start_off = GST_BUFFER_OFFSET_NONE;
1051 self->nsamples_out = 0;
1052 self->nsamples_in = 0;
1053 self->buffer_fill = 0;
1056 g_free (self->kernel);
1057 if (!self->drain_on_changes || latency_changed) {
1058 g_free (self->buffer);
1059 self->buffer = NULL;
1060 self->buffer_fill = 0;
1061 self->buffer_length = 0;
1064 self->kernel = kernel;
1065 self->kernel_length = kernel_length;
1067 gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
1068 gst_audio_fx_base_fir_filter_select_process_function (self,
1069 GST_AUDIO_FILTER_FORMAT (self), GST_AUDIO_FILTER_CHANNELS (self));
1071 if (latency_changed) {
1072 self->latency = latency;
1073 gst_element_post_message (GST_ELEMENT (self),
1074 gst_message_new_latency (GST_OBJECT (self)));
1077 g_mutex_unlock (&self->lock);