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audiofxbasefirfilter: Add a "low-latency" mode

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This will always use time-domain convolution, which lowers the latency.
With FFT convolution it's always a multiple of the kernel length,
with time domain convolution it's only the pre-latency of the filter kernel.
This commit is contained in:
Sebastian Dröge 2009-12-04 09:25:49 +01:00
parent ca568ff079
commit a3d7321c50
2 changed files with 294 additions and 208 deletions
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501
gst/audiofx/audiofxbasefirfilter.c
View file

@ -56,6 +56,14 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* Switch from time-domain to FFT convolution for kernels >= this */
#define FFT_THRESHOLD 32
enum
{
PROP_0 = 0,
PROP_LOW_LATENCY
};
#define DEFAULT_LOW_LATENCY FALSE
GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
@ -76,82 +84,83 @@ static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
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);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_length = 0;
g_free (self->kernel);
self->kernel = NULL;
gst_fft_f64_free (self->fft);
self->fft = NULL;
gst_fft_f64_free (self->ifft);
self->ifft = NULL;
g_free (self->frequency_response);
self->frequency_response = NULL;
g_free (self->fft_buffer);
self->fft_buffer = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
/*
* The code below calculates the linear convolution:
*
* y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
*
* where y is the output, x is the input, M is the length
* of the filter kernel and h is the filter kernel. For x
* holds: x[t] == 0 \forall t < 0.
*
* The runtime complexity of this is O (M) per sample.
*
*/
#define DEFINE_PROCESS_FUNC(width,ctype) \
static guint \
process_##width (GstAudioFXBaseFIRFilter * self, const 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_CAST (self)->format.channels; \
gint res_start; \
gint from_input; \
gint off; \
gdouble *buffer = self->buffer; \
gdouble *kernel = self->kernel; \
guint buffer_length = self->buffer_length; \
\
if (!buffer) { \
self->buffer_length = buffer_length = kernel_length * channels; \
self->buffer = buffer = g_new0 (gdouble, self->buffer_length); \
} \
\
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
k = i % channels; \
l = i / channels; \
from_input = MIN (l, kernel_length-1); \
off = l * channels + k; \
for (j = 0; j <= from_input; j++) { \
dst[i] += src[off] * kernel[j]; \
off -= channels; \
} \
/* j == from_input && off == (l - j) * channels + k */ \
off += kernel_length * channels; \
for (; j < kernel_length; j++) { \
dst[i] += buffer[off] * kernel[j]; \
off -= channels; \
} \
} \
\
/* 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 */ \
/* from now on take kernel length as length over all channels */ \
kernel_length *= channels; \
if (input_samples < kernel_length) \
res_start = kernel_length - input_samples; \
else \
res_start = 0; \
\
for (i = 0; i < res_start; i++) \
buffer[i] = buffer[i + input_samples]; \
/* i == res_start */ \
for (; i < kernel_length; i++) \
buffer[i] = src[input_samples - kernel_length + i]; \
\
self->buffer_fill += kernel_length - res_start; \
if (self->buffer_fill > kernel_length) \
self->buffer_fill = kernel_length; \
\
return input_samples / channels; \
}
static void
gst_audio_fx_base_fir_filter_base_init (gpointer g_class)
{
GstCaps *caps;
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
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);
trans_class->transform_size =
GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform_size);
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->buffer = NULL;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
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);
}
#undef DEFINE_PROCESS_FUNC
/* This implements FFT convolution and uses the overlap-save algorithm.
* See http://cnx.org/content/m12022/latest/ or your favorite
@ -321,83 +330,209 @@ DEFINE_FFT_PROCESS_FUNC (64, double);
#undef DEFINE_FFT_PROCESS_FUNC
/*
* The code below calculates the linear convolution:
*
* y[t] = \sum_{u=0}^{M-1} x[t - u] * h[u]
*
* where y is the output, x is the input, M is the length
* of the filter kernel and h is the filter kernel. For x
* holds: x[t] == 0 \forall t < 0.
*
* The runtime complexity of this is O (M) per sample.
*
*/
#define DEFINE_PROCESS_FUNC(width,ctype) \
static guint \
process_##width (GstAudioFXBaseFIRFilter * self, const 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_CAST (self)->format.channels; \
gint res_start; \
gint from_input; \
gint off; \
gdouble *buffer = self->buffer; \
gdouble *kernel = self->kernel; \
guint buffer_length = self->buffer_length; \
\
if (!buffer) { \
self->buffer_length = buffer_length = kernel_length * channels; \
self->buffer = buffer = g_new0 (gdouble, self->buffer_length); \
} \
\
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
k = i % channels; \
l = i / channels; \
from_input = MIN (l, kernel_length-1); \
off = l * channels + k; \
for (j = 0; j <= from_input; j++) { \
dst[i] += src[off] * kernel[j]; \
off -= channels; \
} \
/* j == from_input && off == (l - j) * channels + k */ \
off += kernel_length * channels; \
for (; j < kernel_length; j++) { \
dst[i] += buffer[off] * kernel[j]; \
off -= channels; \
} \
} \
\
/* 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 */ \
/* from now on take kernel length as length over all channels */ \
kernel_length *= channels; \
if (input_samples < kernel_length) \
res_start = kernel_length - input_samples; \
else \
res_start = 0; \
\
for (i = 0; i < res_start; i++) \
buffer[i] = buffer[i + input_samples]; \
/* i == res_start */ \
for (; i < kernel_length; i++) \
buffer[i] = src[input_samples - kernel_length + i]; \
\
self->buffer_fill += kernel_length - res_start; \
if (self->buffer_fill > kernel_length) \
self->buffer_fill = kernel_length; \
\
return input_samples / channels; \
/* Element class */
static void
gst_audio_fx_base_fir_filter_calculate_frequency_response
(GstAudioFXBaseFIRFilter * self)
{
gst_fft_f64_free (self->fft);
self->fft = NULL;
gst_fft_f64_free (self->ifft);
self->ifft = NULL;
g_free (self->frequency_response);
self->frequency_response_length = 0;
g_free (self->fft_buffer);
self->fft_buffer = NULL;
if (self->kernel && self->kernel_length >= FFT_THRESHOLD
&& !self->low_latency) {
guint block_length, i;
gdouble *kernel_tmp, *kernel = self->kernel;
/* We process 4 * kernel_length samples per pass in FFT mode */
block_length = 4 * self->kernel_length;
block_length = gst_fft_next_fast_length (block_length);
self->block_length = block_length;
kernel_tmp = g_new0 (gdouble, block_length);
memcpy (kernel_tmp, kernel, self->block_length * sizeof (gdouble));
self->fft = gst_fft_f64_new (block_length, FALSE);
self->ifft = gst_fft_f64_new (block_length, TRUE);
self->frequency_response_length = block_length / 2 + 1;
self->frequency_response =
g_new (GstFFTF64Complex, self->frequency_response_length);
gst_fft_f64_fft (self->fft, kernel_tmp, self->frequency_response);
g_free (kernel_tmp);
/* Normalize to make sure IFFT(FFT(x)) == x */
for (i = 0; i < self->frequency_response_length; i++) {
self->frequency_response[i].r /= block_length;
self->frequency_response[i].i /= block_length;
}
}
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
/* Must be called with base transform lock! */
static void
gst_audio_fx_base_fir_filter_select_process_function (GstAudioFXBaseFIRFilter *
self, gint width)
{
if (width == 32 && self->fft && !self->low_latency)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
else if (width == 64 && self->fft && !self->low_latency)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
else if (width == 32)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
else if (width == 64)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
else
self->process = NULL;
}
#undef DEFINE_PROCESS_FUNC
static void
gst_audio_fx_base_fir_filter_dispose (GObject * object)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
g_free (self->buffer);
self->buffer = NULL;
self->buffer_length = 0;
g_free (self->kernel);
self->kernel = NULL;
gst_fft_f64_free (self->fft);
self->fft = NULL;
gst_fft_f64_free (self->ifft);
self->ifft = NULL;
g_free (self->frequency_response);
self->frequency_response = NULL;
g_free (self->fft_buffer);
self->fft_buffer = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_fx_base_fir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
switch (prop_id) {
case PROP_LOW_LATENCY:{
gboolean low_latency;
if (GST_STATE (self) >= GST_STATE_PAUSED) {
g_warning ("Changing the \"low-latency\" property "
"is only allowed in states < PAUSED");
return;
}
GST_BASE_TRANSFORM_LOCK (self);
low_latency = g_value_get_boolean (value);
if (self->low_latency != low_latency) {
self->low_latency = low_latency;
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_CAST (self)->format.width);
}
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_fx_base_fir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);
switch (prop_id) {
case PROP_LOW_LATENCY:
g_value_set_boolean (value, self->low_latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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;
gobject_class->set_property = gst_audio_fx_base_fir_filter_set_property;
gobject_class->get_property = gst_audio_fx_base_fir_filter_get_property;
/**
* GstAudioFXBaseFIRFilter::low-latency:
*
* Work in low-latency mode. This mode is much slower for large filter sizes
* but the latency is always only the pre-latency of the filter.
*
* Since: 0.10.18
*/
g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
g_param_spec_boolean ("low-latency", "Low latency",
"Operate in low latency mode. This mode is slower but the "
"latency will only be the filter pre-latency. "
"Can only be changed in states < PAUSED!", DEFAULT_LOW_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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);
trans_class->transform_size =
GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform_size);
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->buffer = NULL;
self->buffer_length = 0;
self->start_ts = GST_CLOCK_TIME_NONE;
self->start_off = GST_BUFFER_OFFSET_NONE;
self->nsamples_out = 0;
self->nsamples_in = 0;
self->low_latency = DEFAULT_LOW_LATENCY;
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);
}
void
gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
@ -428,7 +563,7 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
}
outsize = outsamples * channels * width;
if (!self->fft) {
if (!self->fft || self->low_latency) {
gint64 diffsize, diffsamples;
/* Process the difference between latency and residue length samples
@ -529,7 +664,6 @@ 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->buffer) {
gst_audio_fx_base_fir_filter_push_residue (self);
@ -543,17 +677,9 @@ gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
self->nsamples_in = 0;
}
if (format->width == 32 && self->fft)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
else if (format->width == 64 && self->fft)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
else if (format->width == 32)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
ret = FALSE;
gst_audio_fx_base_fir_filter_select_process_function (self, format->width);
return TRUE;
return (self->process != NULL);
}
/* GstBaseTransform vmethod implementations */
@ -568,7 +694,7 @@ gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform * base,
GstStructure *s;
gint width, channels;
if (!self->fft || direction == GST_PAD_SRC) {
if (!self->fft || self->low_latency || direction == GST_PAD_SRC) {
*othersize = size;
return TRUE;
}
@ -747,7 +873,7 @@ gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
if (self->fft)
if (self->fft && !self->low_latency)
latency = self->block_length - self->kernel_length + 1;
else
latency = self->latency;
@ -815,10 +941,7 @@ void
gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gdouble * kernel, guint kernel_length, guint64 latency)
{
gdouble *kernel_tmp;
guint i;
gboolean latency_changed;
gint width;
g_return_if_fail (kernel != NULL);
g_return_if_fail (self != NULL);
@ -834,8 +957,9 @@ gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
}
latency_changed = (self->latency != latency
|| (self->kernel_length < FFT_THRESHOLD && kernel_length >= FFT_THRESHOLD)
|| (self->kernel_length >= FFT_THRESHOLD
|| (!self->low_latency && self->kernel_length < FFT_THRESHOLD
&& kernel_length >= FFT_THRESHOLD)
|| (!self->low_latency && self->kernel_length >= FFT_THRESHOLD
&& kernel_length < FFT_THRESHOLD));
g_free (self->kernel);
@ -844,51 +968,12 @@ gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
self->buffer_fill = 0;
self->buffer_length = 0;
gst_fft_f64_free (self->fft);
self->fft = NULL;
gst_fft_f64_free (self->ifft);
self->ifft = NULL;
g_free (self->frequency_response);
self->frequency_response_length = 0;
g_free (self->fft_buffer);
self->fft_buffer = NULL;
self->kernel = kernel;
self->kernel_length = kernel_length;
if (kernel_length >= FFT_THRESHOLD) {
/* We process 4 * kernel_length samples per pass in FFT mode */
kernel_length = 4 * kernel_length;
kernel_length = gst_fft_next_fast_length (kernel_length);
self->block_length = kernel_length;
kernel_tmp = g_new0 (gdouble, kernel_length);
memcpy (kernel_tmp, kernel, self->kernel_length * sizeof (gdouble));
self->fft = gst_fft_f64_new (kernel_length, FALSE);
self->ifft = gst_fft_f64_new (kernel_length, TRUE);
self->frequency_response_length = kernel_length / 2 + 1;
self->frequency_response =
g_new (GstFFTF64Complex, self->frequency_response_length);
gst_fft_f64_fft (self->fft, kernel_tmp, self->frequency_response);
g_free (kernel_tmp);
/* Normalize to make sure IFFT(FFT(x)) == x */
for (i = 0; i < self->frequency_response_length; i++) {
self->frequency_response[i].r /= kernel_length;
self->frequency_response[i].i /= kernel_length;
}
}
width = GST_AUDIO_FILTER_CAST (self)->format.width;
if (width == 32 && self->fft)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
else if (width == 64 && self->fft)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
else if (width == 32)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
else if (width == 64)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_CAST (self)->format.width);
if (latency_changed) {
self->latency = latency;

1
gst/audiofx/audiofxbasefirfilter.h
View file

@ -60,6 +60,7 @@ struct _GstAudioFXBaseFIRFilter {
guint kernel_length; /* length of the filter kernel -- time domain */
guint64 latency; /* pre-latency of the filter kernel */
gboolean low_latency; /* work in slower low latency mode */
/* < private > */
GstAudioFXBaseFIRFilterProcessFunc process;