diff -r 29ecd5cb86b3 -r d43ce56a1534 gst_plugins_good/gst/audiofx/audiofxbasefirfilter.c --- 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 - * 2006 Dreamlab Technologies Ltd. - * 2007-2009 Sebastian Dröge - * - * 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 -#include -#include -#include -#include - -#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); -}