diff options
Diffstat (limited to 'gst/audiofx/audiofxbasefirfilter.c')
-rw-r--r-- | gst/audiofx/audiofxbasefirfilter.c | 527 |
1 files changed, 527 insertions, 0 deletions
diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c new file mode 100644 index 00000000..059c2aa3 --- /dev/null +++ b/gst/audiofx/audiofxbasefirfilter.c @@ -0,0 +1,527 @@ +/* -*- 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); +} |