mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-21 15:56:42 +00:00
Code Issues 1 Projects Releases Wiki Activity

audiofxbasefirfilter: Add special processing functions for Mono/Stereo

Browse source 
This provides another 7% speedup for the time domain convolution and 1.5%
speedup for the FFT convolution on Mono input.

This optimization assumes that the compiler simplifies calculations
and conditions on constant numbers and unrolls loops with a constant
number of repeats.
This commit is contained in:
Sebastian Dröge 2009-12-06 16:58:51 +01:00
parent a3d7321c50
commit cd2b1c1b58
1 changed files with 79 additions and 16 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

93
gst/audiofx/audiofxbasefirfilter.c
View file

@ -100,9 +100,20 @@ static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *
static guint \ static guint \
process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \ process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
{ \ { \
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels; \
TIME_DOMAIN_CONVOLUTION_BODY (channels); \
}
#define DEFINE_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
static guint \
process_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype * dst, guint input_samples) \
{ \
TIME_DOMAIN_CONVOLUTION_BODY (channels); \
}
#define TIME_DOMAIN_CONVOLUTION_BODY(channels) G_STMT_START { \
gint kernel_length = self->kernel_length; \ gint kernel_length = self->kernel_length; \
gint i, j, k, l; \ gint i, j, k, l; \
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels; \
gint res_start; \ gint res_start; \
gint from_input; \ gint from_input; \
gint off; \ gint off; \
@ -155,12 +166,20 @@ process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype
self->buffer_fill = kernel_length; \ self->buffer_fill = kernel_length; \
\ \
return input_samples / channels; \ return input_samples / channels; \
} } G_STMT_END
DEFINE_PROCESS_FUNC (32, float); DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double); DEFINE_PROCESS_FUNC (64, double);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
DEFINE_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
#undef TIME_DOMAIN_CONVOLUTION_BODY
#undef DEFINE_PROCESS_FUNC #undef DEFINE_PROCESS_FUNC
#undef DEFINE_PROCESS_FUNC_FIXED_CHANNELS
/* This implements FFT convolution and uses the overlap-save algorithm. /* This implements FFT convolution and uses the overlap-save algorithm.
* See http://cnx.org/content/m12022/latest/ or your favorite * See http://cnx.org/content/m12022/latest/ or your favorite
@ -221,6 +240,18 @@ process_fft_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
g##ctype * dst, guint input_samples) \ g##ctype * dst, guint input_samples) \
{ \ { \
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels; \ gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels; \
FFT_CONVOLUTION_BODY (channels); \
}
#define DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS(width,channels,ctype) \
static guint \
process_fft_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
g##ctype * dst, guint input_samples) \
{ \
FFT_CONVOLUTION_BODY (channels); \
}
#define FFT_CONVOLUTION_BODY(channels) G_STMT_START { \
gint i, j; \ gint i, j; \
guint pass; \ guint pass; \
guint kernel_length = self->kernel_length; \ guint kernel_length = self->kernel_length; \
@ -323,12 +354,20 @@ process_fft_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, \
self->buffer_fill = buffer_fill; \ self->buffer_fill = buffer_fill; \
\ \
return generated; \ return generated; \
} } G_STMT_END
DEFINE_FFT_PROCESS_FUNC (32, float); DEFINE_FFT_PROCESS_FUNC (32, float);
DEFINE_FFT_PROCESS_FUNC (64, double); DEFINE_FFT_PROCESS_FUNC (64, double);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 1, float);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 1, double);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (32, 2, float);
DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS (64, 2, double);
#undef FFT_CONVOLUTION_BODY
#undef DEFINE_FFT_PROCESS_FUNC #undef DEFINE_FFT_PROCESS_FUNC
#undef DEFINE_FFT_PROCESS_FUNC_FIXED_CHANNELS
/* Element class */ /* Element class */
static void static void
@ -376,18 +415,39 @@ static void
/* Must be called with base transform lock! */ /* Must be called with base transform lock! */
static void static void
gst_audio_fx_base_fir_filter_select_process_function (GstAudioFXBaseFIRFilter * gst_audio_fx_base_fir_filter_select_process_function (GstAudioFXBaseFIRFilter *
self, gint width) self, gint width, gint channels)
{ {
if (width == 32 && self->fft && !self->low_latency) if (width == 32 && self->fft && !self->low_latency) {
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32; if (channels == 1)
else if (width == 64 && self->fft && !self->low_latency) self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_32;
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64; else if (channels == 2)
else if (width == 32) self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_32;
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
else if (width == 64)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
else else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_32;
} else if (width == 64 && self->fft && !self->low_latency) {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_1_64;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_2_64;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_fft_64;
} else if (width == 32) {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_32;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_32;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
} else if (width == 64) {
if (channels == 1)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_1_64;
else if (channels == 2)
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_2_64;
else
self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
} else {
self->process = NULL; self->process = NULL;
}
} }
static void static void
@ -439,7 +499,8 @@ gst_audio_fx_base_fir_filter_set_property (GObject * object, guint prop_id,
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_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self, gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_CAST (self)->format.width); GST_AUDIO_FILTER_CAST (self)->format.width,
GST_AUDIO_FILTER_CAST (self)->format.channels);
} }
GST_BASE_TRANSFORM_UNLOCK (self); GST_BASE_TRANSFORM_UNLOCK (self);
break; break;
@ -677,7 +738,8 @@ gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
self->nsamples_in = 0; self->nsamples_in = 0;
} }
gst_audio_fx_base_fir_filter_select_process_function (self, format->width); gst_audio_fx_base_fir_filter_select_process_function (self, format->width,
format->channels);
return (self->process != NULL); return (self->process != NULL);
} }
@ -973,7 +1035,8 @@ gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
gst_audio_fx_base_fir_filter_calculate_frequency_response (self); gst_audio_fx_base_fir_filter_calculate_frequency_response (self);
gst_audio_fx_base_fir_filter_select_process_function (self, gst_audio_fx_base_fir_filter_select_process_function (self,
GST_AUDIO_FILTER_CAST (self)->format.width); GST_AUDIO_FILTER_CAST (self)->format.width,
GST_AUDIO_FILTER_CAST (self)->format.channels);
if (latency_changed) { if (latency_changed) {
self->latency = latency; self->latency = latency;