--- a/gst_plugins_good/gst/audiofx/audiofxbasefirfilter.c Tue Jul 06 14:35:10 2010 +0300
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,527 +0,0 @@
-/* -*- 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- *
- *
- * 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
- */
-
-#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 "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-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_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \
- "FIR filter base class");
-
-GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
- GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
-
-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_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
- GstRingBufferSpec * format);
-
-static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
- GstQuery * query);
-static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *
- pad);
-
-/* Element class */
-
-static void
-gst_audio_fx_base_fir_filter_dispose (GObject * object)
-{
- GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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_audio_fx_base_fir_filter_base_init (gpointer g_class)
-{
- GstCaps *caps;
-
- 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_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
- GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
- GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
-
- gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;
-
- 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->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);
- filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
-}
-
-static void
-gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
- GstAudioFXBaseFIRFilterClass * g_class)
-{
- self->kernel = NULL;
- self->residue = NULL;
-
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
-
- gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
- gst_audio_fx_base_fir_filter_query);
- gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
- gst_audio_fx_base_fir_filter_query_type);
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioFXBaseFIRFilter * self, g##ctype * src, g##ctype * 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; \
-}
-
-DEFINE_PROCESS_FUNC (32, float);
-DEFINE_PROCESS_FUNC (64, double);
-
-#undef DEFINE_PROCESS_FUNC
-
-void
-gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * 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;
-
- if (channels == 0 || rate == 0) {
- self->residue_length = 0;
- return;
- }
-
- /* 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) {
- self->residue_length = 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);
- self->residue_length = 0;
- 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;
- self->next_off = GST_BUFFER_OFFSET_END (outbuf);
- }
-
- 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");
- }
-
- self->residue_length = 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,
- GstRingBufferSpec * format)
-{
- GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
- gboolean ret = TRUE;
-
- if (self->residue) {
- gst_audio_fx_base_fir_filter_push_residue (self);
- g_free (self->residue);
- self->residue = NULL;
- self->residue_length = 0;
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
- }
-
- if (format->width == 32)
- self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
- else if (format->width == 64)
- self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
- else
- ret = FALSE;
-
- return TRUE;
-}
-
-/* GstBaseTransform vmethod implementations */
-
-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;
- 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 = 0;
-
- timestamp = GST_BUFFER_TIMESTAMP (outbuf);
- if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GST_ERROR_OBJECT (self, "Invalid timestamp");
- return GST_FLOW_ERROR;
- }
-
- gst_object_sync_values (G_OBJECT (self), timestamp);
-
- g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
- g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
-
- if (!self->residue)
- self->residue = g_new0 (gdouble, self->kernel_length * channels);
-
- /* Reset the residue if already existing on discont buffers */
- if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)
- && timestamp - gst_util_uint64_scale (MIN (self->latency,
- self->residue_length / channels), GST_SECOND,
- rate) - self->next_ts > 5 * GST_MSECOND)) {
- GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
- if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
- gst_audio_fx_base_fir_filter_push_residue (self);
- self->residue_length = 0;
- self->next_ts = timestamp;
- self->next_off = GST_BUFFER_OFFSET (inbuf);
- } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {
- self->next_ts = timestamp;
- self->next_off = GST_BUFFER_OFFSET (inbuf);
- }
-
- /* Calculate the number of samples we can push out now without outputting
- * latency zeros in the beginning */
- diff = self->latency * 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) {
- return GST_BASE_TRANSFORM_FLOW_DROPPED;
- }
-
- GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);
- GST_BUFFER_OFFSET (outbuf) = self->next_off;
- if (GST_BUFFER_OFFSET_IS_VALID (outbuf))
- GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;
- else
- GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
-
- if (output_samples < input_samples) {
- GST_BUFFER_DATA (outbuf) +=
- diff * (GST_AUDIO_FILTER (self)->format.width / 8);
- GST_BUFFER_SIZE (outbuf) -=
- diff * (GST_AUDIO_FILTER (self)->format.width / 8);
- }
-
- self->next_ts += GST_BUFFER_DURATION (outbuf);
- self->next_off = GST_BUFFER_OFFSET_END (outbuf);
-
- 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);
-
- return GST_FLOW_OK;
-}
-
-static gboolean
-gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
-{
- GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
-
- self->residue_length = 0;
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
-
- 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->residue);
- self->residue = NULL;
-
- return TRUE;
-}
-
-static gboolean
-gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
-{
- GstAudioFXBaseFIRFilter *self =
- GST_AUDIO_FX_BASE_FIR_FILTER (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 (rate == 0) {
- res = FALSE;
- } else 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 = gst_util_uint64_scale (self->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);
- }
- gst_object_unref (peer);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
- gst_object_unref (self);
- return res;
-}
-
-static const GstQueryType *
-gst_audio_fx_base_fir_filter_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
-static gboolean
-gst_audio_fx_base_fir_filter_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->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
- break;
- default:
- break;
- }
-
- return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
-}
-
-void
-gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
- gdouble * kernel, guint kernel_length, guint64 latency)
-{
- g_return_if_fail (kernel != NULL);
- g_return_if_fail (self != NULL);
-
- GST_BASE_TRANSFORM_LOCK (self);
- if (self->residue) {
- gst_audio_fx_base_fir_filter_push_residue (self);
- self->next_ts = GST_CLOCK_TIME_NONE;
- self->next_off = GST_BUFFER_OFFSET_NONE;
- self->residue_length = 0;
- }
-
- g_free (self->kernel);
- g_free (self->residue);
-
- self->kernel = kernel;
- self->kernel_length = kernel_length;
-
- if (GST_AUDIO_FILTER (self)->format.channels) {
- self->residue =
- g_new0 (gdouble,
- kernel_length * GST_AUDIO_FILTER (self)->format.channels);
- self->residue_length = 0;
- }
-
- if (self->latency != latency) {
- self->latency = latency;
- gst_element_post_message (GST_ELEMENT (self),
- gst_message_new_latency (GST_OBJECT (self)));
- }
-
- GST_BASE_TRANSFORM_UNLOCK (self);
-}