1 /* -*- c-basic-offset: 2 -*-

     2  * 

     3  * GStreamer

     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>

     7  *

     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.

    12  *

    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.

    17  *

    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.

    22  * 

    23  * 

    24  * TODO:  - Implement the convolution in place, probably only makes sense

    25  *          when using FFT convolution as currently the convolution itself

    26  *          is probably the bottleneck

    27  *        - Maybe allow cascading the filter to get a better stopband attenuation.

    28  *          Can be done by convolving a filter kernel with itself

    29  */

    30 

    31 #ifdef HAVE_CONFIG_H

    32 #include "config.h"

    33 #endif

    34 

    35 #include <string.h>

    36 #include <math.h>

    37 #include <gst/gst.h>

    38 #include <gst/audio/gstaudiofilter.h>

    39 #include <gst/controller/gstcontroller.h>

    40 

    41 #include "audiofxbasefirfilter.h"

    42 

    43 #define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug

    44 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

    45 

    46 #define ALLOWED_CAPS \

    47     "audio/x-raw-float, "                                             \

    48     " width = (int) { 32, 64 }, "                                     \

    49     " endianness = (int) BYTE_ORDER, "                                \

    50     " rate = (int) [ 1, MAX ], "                                      \

    51     " channels = (int) [ 1, MAX ]"

    52 

    53 #define DEBUG_INIT(bla) \

    54   GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \

    55       "FIR filter base class");

    56 

    57 GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,

    58     GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);

    59 

    60 static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *

    61     base, GstBuffer * inbuf, GstBuffer * outbuf);

    62 static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);

    63 static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);

    64 static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,

    65     GstEvent * event);

    66 static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,

    67     GstRingBufferSpec * format);

    68 

    69 static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,

    70     GstQuery * query);

    71 static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *

    72     pad);

    73 

    74 /* Element class */

    75 

    76 static void

    77 gst_audio_fx_base_fir_filter_dispose (GObject * object)

    78 {

    79   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);

    80 

    81   if (self->residue) {

    82     g_free (self->residue);

    83     self->residue = NULL;

    84   }

    85 

    86   if (self->kernel) {

    87     g_free (self->kernel);

    88     self->kernel = NULL;

    89   }

    90 

    91   G_OBJECT_CLASS (parent_class)->dispose (object);

    92 }

    93 

    94 static void

    95 gst_audio_fx_base_fir_filter_base_init (gpointer g_class)

    96 {

    97   GstCaps *caps;

    98 

    99   caps = gst_caps_from_string (ALLOWED_CAPS);

   100   gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),

   101       caps);

   102   gst_caps_unref (caps);

   103 }

   104 

   105 static void

   106 gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)

   107 {

   108   GObjectClass *gobject_class = (GObjectClass *) klass;

   109   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;

   110   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;

   111 

   112   gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;

   113 

   114   trans_class->transform =

   115       GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);

   116   trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);

   117   trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);

   118   trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);

   119   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);

   120 }

   121 

   122 static void

   123 gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,

   124     GstAudioFXBaseFIRFilterClass * g_class)

   125 {

   126   self->kernel = NULL;

   127   self->residue = NULL;

   128 

   129   self->next_ts = GST_CLOCK_TIME_NONE;

   130   self->next_off = GST_BUFFER_OFFSET_NONE;

   131 

   132   gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,

   133       gst_audio_fx_base_fir_filter_query);

   134   gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,

   135       gst_audio_fx_base_fir_filter_query_type);

   136 }

   137 

   138 #define DEFINE_PROCESS_FUNC(width,ctype) \

   139 static void \

   140 process_##width (GstAudioFXBaseFIRFilter * self, g##ctype * src, g##ctype * dst, guint input_samples) \

   141 { \

   142   gint kernel_length = self->kernel_length; \

   143   gint i, j, k, l; \

   144   gint channels = GST_AUDIO_FILTER (self)->format.channels; \

   145   gint res_start; \

   146   \

   147   /* convolution */ \

   148   for (i = 0; i < input_samples; i++) { \

   149     dst[i] = 0.0; \

   150     k = i % channels; \

   151     l = i / channels; \

   152     for (j = 0; j < kernel_length; j++) \

   153       if (l < j) \

   154         dst[i] += \

   155             self->residue[(kernel_length + l - j) * channels + \

   156             k] * self->kernel[j]; \

   157       else \

   158         dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \

   159   } \

   160   \

   161   /* copy the tail of the current input buffer to the residue, while \

   162    * keeping parts of the residue if the input buffer is smaller than \

   163    * the kernel length */ \

   164   if (input_samples < kernel_length * channels) \

   165     res_start = kernel_length * channels - input_samples; \

   166   else \

   167     res_start = 0; \

   168   \

   169   for (i = 0; i < res_start; i++) \

   170     self->residue[i] = self->residue[i + input_samples]; \

   171   for (i = res_start; i < kernel_length * channels; i++) \

   172     self->residue[i] = src[input_samples - kernel_length * channels + i]; \

   173   \

   174   self->residue_length += kernel_length * channels - res_start; \

   175   if (self->residue_length > kernel_length * channels) \

   176     self->residue_length = kernel_length * channels; \

   177 }

   178 

   179 DEFINE_PROCESS_FUNC (32, float);

   180 DEFINE_PROCESS_FUNC (64, double);

   181 

   182 #undef DEFINE_PROCESS_FUNC

   183 

   184 void

   185 gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)

   186 {

   187   GstBuffer *outbuf;

   188   GstFlowReturn res;

   189   gint rate = GST_AUDIO_FILTER (self)->format.rate;

   190   gint channels = GST_AUDIO_FILTER (self)->format.channels;

   191   gint outsize, outsamples;

   192   gint diffsize, diffsamples;

   193   guint8 *in, *out;

   194 

   195   if (channels == 0 || rate == 0) {

   196     self->residue_length = 0;

   197     return;

   198   }

   199 

   200   /* Calculate the number of samples and their memory size that

   201    * should be pushed from the residue */

   202   outsamples = MIN (self->latency, self->residue_length / channels);

   203   outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);

   204   if (outsize == 0) {

   205     self->residue_length = 0;

   206     return;

   207   }

   208 

   209   /* Process the difference between latency and residue_length samples

   210    * to start at the actual data instead of starting at the zeros before

   211    * when we only got one buffer smaller than latency */

   212   diffsamples = self->latency - self->residue_length / channels;

   213   diffsize =

   214       diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);

   215   if (diffsize > 0) {

   216     in = g_new0 (guint8, diffsize);

   217     out = g_new0 (guint8, diffsize);

   218     self->process (self, in, out, diffsamples * channels);

   219     g_free (in);

   220     g_free (out);

   221   }

   222 

   223   res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,

   224       GST_BUFFER_OFFSET_NONE, outsize,

   225       GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);

   226 

   227   if (G_UNLIKELY (res != GST_FLOW_OK)) {

   228     GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);

   229     self->residue_length = 0;

   230     return;

   231   }

   232 

   233   /* Convolve the residue with zeros to get the actual remaining data */

   234   in = g_new0 (guint8, outsize);

   235   self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);

   236   g_free (in);

   237 

   238   /* Set timestamp, offset, etc from the values we

   239    * saved when processing the regular buffers */

   240   if (GST_CLOCK_TIME_IS_VALID (self->next_ts))

   241     GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;

   242   else

   243     GST_BUFFER_TIMESTAMP (outbuf) = 0;

   244   GST_BUFFER_DURATION (outbuf) =

   245       gst_util_uint64_scale (outsamples, GST_SECOND, rate);

   246   self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);

   247 

   248   if (self->next_off != GST_BUFFER_OFFSET_NONE) {

   249     GST_BUFFER_OFFSET (outbuf) = self->next_off;

   250     GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;

   251     self->next_off = GST_BUFFER_OFFSET_END (outbuf);

   252   }

   253 

   254   GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"

   255       GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"

   256       " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),

   257       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),

   258       GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),

   259       GST_BUFFER_OFFSET_END (outbuf), outsamples);

   260 

   261   res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);

   262 

   263   if (G_UNLIKELY (res != GST_FLOW_OK)) {

   264     GST_WARNING_OBJECT (self, "failed to push residue");

   265   }

   266 

   267   self->residue_length = 0;

   268 }

   269 

   270 /* GstAudioFilter vmethod implementations */

   271 

   272 /* get notified of caps and plug in the correct process function */

   273 static gboolean

   274 gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,

   275     GstRingBufferSpec * format)

   276 {

   277   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

   278   gboolean ret = TRUE;

   279 

   280   if (self->residue) {

   281     gst_audio_fx_base_fir_filter_push_residue (self);

   282     g_free (self->residue);

   283     self->residue = NULL;

   284     self->residue_length = 0;

   285     self->next_ts = GST_CLOCK_TIME_NONE;

   286     self->next_off = GST_BUFFER_OFFSET_NONE;

   287   }

   288 

   289   if (format->width == 32)

   290     self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;

   291   else if (format->width == 64)

   292     self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;

   293   else

   294     ret = FALSE;

   295 

   296   return TRUE;

   297 }

   298 

   299 /* GstBaseTransform vmethod implementations */

   300 

   301 static GstFlowReturn

   302 gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,

   303     GstBuffer * inbuf, GstBuffer * outbuf)

   304 {

   305   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

   306   GstClockTime timestamp;

   307   gint channels = GST_AUDIO_FILTER (self)->format.channels;

   308   gint rate = GST_AUDIO_FILTER (self)->format.rate;

   309   gint input_samples =

   310       GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);

   311   gint output_samples = input_samples;

   312   gint diff = 0;

   313 

   314   timestamp = GST_BUFFER_TIMESTAMP (outbuf);

   315   if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {

   316     GST_ERROR_OBJECT (self, "Invalid timestamp");

   317     return GST_FLOW_ERROR;

   318   }

   319 

   320   gst_object_sync_values (G_OBJECT (self), timestamp);

   321 

   322   g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);

   323   g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);

   324 

   325   if (!self->residue)

   326     self->residue = g_new0 (gdouble, self->kernel_length * channels);

   327 

   328   /* Reset the residue if already existing on discont buffers */

   329   if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)

   330           && timestamp - gst_util_uint64_scale (MIN (self->latency,

   331                   self->residue_length / channels), GST_SECOND,

   332               rate) - self->next_ts > 5 * GST_MSECOND)) {

   333     GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");

   334     if (GST_CLOCK_TIME_IS_VALID (self->next_ts))

   335       gst_audio_fx_base_fir_filter_push_residue (self);

   336     self->residue_length = 0;

   337     self->next_ts = timestamp;

   338     self->next_off = GST_BUFFER_OFFSET (inbuf);

   339   } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {

   340     self->next_ts = timestamp;

   341     self->next_off = GST_BUFFER_OFFSET (inbuf);

   342   }

   343 

   344   /* Calculate the number of samples we can push out now without outputting

   345    * latency zeros in the beginning */

   346   diff = self->latency * channels - self->residue_length;

   347   if (diff > 0)

   348     output_samples -= diff;

   349 

   350   self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),

   351       input_samples);

   352 

   353   if (output_samples <= 0) {

   354     return GST_BASE_TRANSFORM_FLOW_DROPPED;

   355   }

   356 

   357   GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;

   358   GST_BUFFER_DURATION (outbuf) =

   359       gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);

   360   GST_BUFFER_OFFSET (outbuf) = self->next_off;

   361   if (GST_BUFFER_OFFSET_IS_VALID (outbuf))

   362     GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;

   363   else

   364     GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;

   365 

   366   if (output_samples < input_samples) {

   367     GST_BUFFER_DATA (outbuf) +=

   368         diff * (GST_AUDIO_FILTER (self)->format.width / 8);

   369     GST_BUFFER_SIZE (outbuf) -=

   370         diff * (GST_AUDIO_FILTER (self)->format.width / 8);

   371   }

   372 

   373   self->next_ts += GST_BUFFER_DURATION (outbuf);

   374   self->next_off = GST_BUFFER_OFFSET_END (outbuf);

   375 

   376   GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"

   377       GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"

   378       " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),

   379       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),

   380       GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),

   381       GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);

   382 

   383   return GST_FLOW_OK;

   384 }

   385 

   386 static gboolean

   387 gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)

   388 {

   389   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

   390 

   391   self->residue_length = 0;

   392   self->next_ts = GST_CLOCK_TIME_NONE;

   393   self->next_off = GST_BUFFER_OFFSET_NONE;

   394 

   395   return TRUE;

   396 }

   397 

   398 static gboolean

   399 gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)

   400 {

   401   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

   402 

   403   g_free (self->residue);

   404   self->residue = NULL;

   405 

   406   return TRUE;

   407 }

   408 

   409 static gboolean

   410 gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)

   411 {

   412   GstAudioFXBaseFIRFilter *self =

   413       GST_AUDIO_FX_BASE_FIR_FILTER (gst_pad_get_parent (pad));

   414   gboolean res = TRUE;

   415 

   416   switch (GST_QUERY_TYPE (query)) {

   417     case GST_QUERY_LATENCY:

   418     {

   419       GstClockTime min, max;

   420       gboolean live;

   421       guint64 latency;

   422       GstPad *peer;

   423       gint rate = GST_AUDIO_FILTER (self)->format.rate;

   424 

   425       if (rate == 0) {

   426         res = FALSE;

   427       } else if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {

   428         if ((res = gst_pad_query (peer, query))) {

   429           gst_query_parse_latency (query, &live, &min, &max);

   430 

   431           GST_DEBUG_OBJECT (self, "Peer latency: min %"

   432               GST_TIME_FORMAT " max %" GST_TIME_FORMAT,

   433               GST_TIME_ARGS (min), GST_TIME_ARGS (max));

   434 

   435           /* add our own latency */

   436           latency = gst_util_uint64_scale (self->latency, GST_SECOND, rate);

   437 

   438           GST_DEBUG_OBJECT (self, "Our latency: %"

   439               GST_TIME_FORMAT, GST_TIME_ARGS (latency));

   440 

   441           min += latency;

   442           if (max != GST_CLOCK_TIME_NONE)

   443             max += latency;

   444 

   445           GST_DEBUG_OBJECT (self, "Calculated total latency : min %"

   446               GST_TIME_FORMAT " max %" GST_TIME_FORMAT,

   447               GST_TIME_ARGS (min), GST_TIME_ARGS (max));

   448 

   449           gst_query_set_latency (query, live, min, max);

   450         }

   451         gst_object_unref (peer);

   452       }

   453       break;

   454     }

   455     default:

   456       res = gst_pad_query_default (pad, query);

   457       break;

   458   }

   459   gst_object_unref (self);

   460   return res;

   461 }

   462 

   463 static const GstQueryType *

   464 gst_audio_fx_base_fir_filter_query_type (GstPad * pad)

   465 {

   466   static const GstQueryType types[] = {

   467     GST_QUERY_LATENCY,

   468     0

   469   };

   470 

   471   return types;

   472 }

   473 

   474 static gboolean

   475 gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)

   476 {

   477   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

   478 

   479   switch (GST_EVENT_TYPE (event)) {

   480     case GST_EVENT_EOS:

   481       gst_audio_fx_base_fir_filter_push_residue (self);

   482       self->next_ts = GST_CLOCK_TIME_NONE;

   483       self->next_off = GST_BUFFER_OFFSET_NONE;

   484       break;

   485     default:

   486       break;

   487   }

   488 

   489   return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);

   490 }

   491 

   492 void

   493 gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,

   494     gdouble * kernel, guint kernel_length, guint64 latency)

   495 {

   496   g_return_if_fail (kernel != NULL);

   497   g_return_if_fail (self != NULL);

   498 

   499   GST_BASE_TRANSFORM_LOCK (self);

   500   if (self->residue) {

   501     gst_audio_fx_base_fir_filter_push_residue (self);

   502     self->next_ts = GST_CLOCK_TIME_NONE;

   503     self->next_off = GST_BUFFER_OFFSET_NONE;

   504     self->residue_length = 0;

   505   }

   506 

   507   g_free (self->kernel);

   508   g_free (self->residue);

   509 

   510   self->kernel = kernel;

   511   self->kernel_length = kernel_length;

   512 

   513   if (GST_AUDIO_FILTER (self)->format.channels) {

   514     self->residue =

   515         g_new0 (gdouble,

   516         kernel_length * GST_AUDIO_FILTER (self)->format.channels);

   517     self->residue_length = 0;

   518   }

   519 

   520   if (self->latency != latency) {

   521     self->latency = latency;

   522     gst_element_post_message (GST_ELEMENT (self),

   523         gst_message_new_latency (GST_OBJECT (self)));

   524   }

   525 

   526   GST_BASE_TRANSFORM_UNLOCK (self);

   527 }