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audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once

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This commit is contained in:
Sebastian Dröge 2009-11-26 10:45:37 +01:00
parent abb437454e
commit c5f955a3b6
1 changed files with 27 additions and 25 deletions
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52
gst/audiofx/audiofxbasefirfilter.c
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@ -38,6 +38,11 @@
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
/* FIXME: Remove this once we depend on gst-plugins-base 0.10.26 */
#ifndef GST_AUDIO_FILTER_CAST
#define GST_AUDIO_FILTER_CAST(obj) ((GstAudioFilter *) (obj))
#endif
#include "audiofxbasefirfilter.h"
#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
@ -142,7 +147,7 @@ process_##width (GstAudioFXBaseFIRFilter * self, const g##ctype * src, g##ctype
{ \
gint kernel_length = self->kernel_length; \
gint i, j, k, l; \
gint channels = GST_AUDIO_FILTER (self)->format.channels; \
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels; \
gint res_start; \
\
/* convolution */ \
@ -187,10 +192,11 @@ 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 rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
gint outsize, outsamples;
gint diffsize, diffsamples;
gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
guint8 *in, *out;
if (channels == 0 || rate == 0) {
@ -201,7 +207,7 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = MIN (self->latency, self->buffer_fill / channels);
outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
outsize = outsamples * channels * width;
if (outsize == 0) {
self->buffer_fill = 0;
return;
@ -211,9 +217,8 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
* 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->buffer_fill / channels;
diffsize =
diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (diffsize > 0) {
if (diffsamples > 0) {
diffsize = diffsamples * channels * width;
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->process (self, in, out, diffsamples * channels);
@ -221,9 +226,9 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
g_free (out);
}
res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM_CAST (self)->srcpad,
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
GST_PAD_CAPS (GST_BASE_TRANSFORM_CAST (self)->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
@ -262,7 +267,7 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
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);
res = gst_pad_push (GST_BASE_TRANSFORM_CAST (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
@ -309,10 +314,10 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
GstClockTime timestamp, expected_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 channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
gint rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
gint input_samples = (GST_BUFFER_SIZE (outbuf) / width) / channels;
gint output_samples = input_samples;
gint diff = 0;
@ -358,12 +363,12 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
/* Calculate the number of samples we can push out now without outputting
* latency zeros in the beginning */
diff = self->latency * channels - self->buffer_fill;
diff = self->latency - self->buffer_fill / channels;
if (diff > 0)
output_samples -= diff;
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
input_samples * channels);
if (output_samples <= 0) {
return GST_BASE_TRANSFORM_FLOW_DROPPED;
@ -371,31 +376,28 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GST_BUFFER_TIMESTAMP (outbuf) = expected_timestamp;
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_round (output_samples / channels, GST_SECOND, rate);
gst_util_uint64_scale_round (output_samples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + self->nsamples;
GST_BUFFER_OFFSET_END (outbuf) =
self->start_off + output_samples / channels;
GST_BUFFER_OFFSET_END (outbuf) = self->start_off + output_samples;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
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);
GST_BUFFER_DATA (outbuf) += diff * width;
GST_BUFFER_SIZE (outbuf) -= diff * width;
}
self->nsamples += output_samples / channels;
self->nsamples += output_samples;
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", 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);
GST_BUFFER_OFFSET_END (outbuf), output_samples);
return GST_FLOW_OK;
}