From: Thomas Vander Stichele Date: Tue, 22 Nov 2005 13:13:21 +0000 (+0000) Subject: gst/cutter/gstcutter.c: copy calculation code from level; remove use of some audio... X-Git-Tag: RELEASE-0_9_6~17 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=baba27fb184cad400e2315f2f159767b4b4ca74b;p=platform%2Fupstream%2Fgst-plugins-good.git gst/cutter/gstcutter.c: copy calculation code from level; remove use of some audio functions Original commit message from CVS: * gst/cutter/gstcutter.c: (gst_cutter_chain), (gst_cutter_set_property), (gst_cutter_get_caps): copy calculation code from level; remove use of some audio functions --- diff --git a/ChangeLog b/ChangeLog index 4bedc7c..2d7f3f9 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,10 @@ +2005-11-22 Thomas Vander Stichele + + * gst/cutter/gstcutter.c: (gst_cutter_chain), + (gst_cutter_set_property), (gst_cutter_get_caps): + copy calculation code from level; remove use of some audio + functions + 2005-11-22 Andy Wingo * Update for gst_tag_setter API changes. diff --git a/gst/cutter/gstcutter.c b/gst/cutter/gstcutter.c index 78f35a8..ec56cdd 100644 --- a/gst/cutter/gstcutter.c +++ b/gst/cutter/gstcutter.c @@ -30,7 +30,6 @@ GST_DEBUG_CATEGORY (cutter_debug); #define GST_CAT_DEFAULT cutter_debug - #define CUTTER_DEFAULT_THRESHOLD_LEVEL 0.1 #define CUTTER_DEFAULT_THRESHOLD_LENGTH (500 * GST_MSECOND) #define CUTTER_DEFAULT_PRE_LENGTH (200 * GST_MSECOND) @@ -45,16 +44,24 @@ static GstElementDetails cutter_details = { static GstStaticPadTemplate cutter_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; " - GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS) + GST_STATIC_CAPS ("audio/x-raw-int, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) { 8, 16 }, " + "depth = (int) { 8, 16 }, " "signed = (boolean) true") ); static GstStaticPadTemplate cutter_sink_factory = - GST_STATIC_PAD_TEMPLATE ("sink", +GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; " - GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS) + GST_STATIC_CAPS ("audio/x-raw-int, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) { 8, 16 }, " + "depth = (int) { 8, 16 }, " "signed = (boolean) true") ); enum @@ -75,8 +82,6 @@ static void gst_cutter_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstFlowReturn gst_cutter_chain (GstPad * pad, GstBuffer * buffer); -static double inline gst_cutter_16bit_ms (gint16 * data, guint numsamples); -static double inline gst_cutter_8bit_ms (gint8 * data, guint numsamples); void gst_cutter_get_caps (GstPad * pad, GstCutter * filter); @@ -171,13 +176,46 @@ gst_cutter_message_new (GstCutter * c, gboolean above, GstClockTime timestamp) return gst_message_new_element (GST_OBJECT (c), s); } +/* Calculate the Normalized Cumulative Square over a buffer of the given type + * and over all channels combined */ + +#define DEFINE_CUTTER_CALCULATOR(TYPE, RESOLUTION) \ +static void inline \ +gst_cutter_calculate_##TYPE (TYPE * in, guint num, \ + double *NCS) \ +{ \ + register int j; \ + double squaresum = 0.0; /* square sum of the integer samples */ \ + register double square = 0.0; /* Square */ \ + gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \ + \ + *NCS = 0.0; /* Normalized Cumulative Square */ \ + \ + normalizer = (double) (1 << (RESOLUTION * 2)); \ + \ + for (j = 0; j < num; j++) \ + { \ + square = ((double) in[j]) * in[j]; \ + squaresum += square; \ + } \ + \ + \ + *NCS = squaresum / normalizer; \ +} + +DEFINE_CUTTER_CALCULATOR (gint16, 15); +DEFINE_CUTTER_CALCULATOR (gint8, 7); + + static GstFlowReturn gst_cutter_chain (GstPad * pad, GstBuffer * buf) { GstCutter *filter; gint16 *in_data; - double RMS = 0.0; /* RMS of signal in buffer */ - double ms = 0.0; /* mean square value of buffer */ + guint num_samples; + gdouble NCS = 0.0; /* Normalized Cumulative Square of buffer */ + gdouble RMS = 0.0; /* RMS of signal in buffer */ + gdouble NMS = 0.0; /* Normalized Mean Square of buffer */ static gboolean silent_prev = FALSE; /* previous value of silent */ GstBuffer *prebuf; /* pointer to a prebuffer element */ @@ -204,10 +242,14 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) /* calculate mean square value on buffer */ switch (filter->width) { case 16: - ms = gst_cutter_16bit_ms (in_data, GST_BUFFER_SIZE (buf) / 2); + num_samples = GST_BUFFER_SIZE (buf) / 2; + gst_cutter_calculate_gint16 (in_data, num_samples, &NCS); + NMS = NCS / num_samples; break; case 8: - ms = gst_cutter_8bit_ms ((gint8 *) in_data, GST_BUFFER_SIZE (buf)); + num_samples = GST_BUFFER_SIZE (buf); + gst_cutter_calculate_gint8 ((gint8 *) in_data, num_samples, &NCS); + NMS = NCS / num_samples; break; default: /* this shouldn't happen */ @@ -217,12 +259,12 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) silent_prev = filter->silent; - RMS = sqrt (ms) / (double) filter->max_sample; + RMS = sqrt (NMS); /* if RMS below threshold, add buffer length to silent run length count * if not, reset */ - GST_LOG_OBJECT (filter, "buffer stats: ms %f, RMS %f, audio length %f", - ms, RMS, gst_audio_duration_from_pad_buffer (filter->sinkpad, buf)); + GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", + NMS, RMS, gst_audio_duration_from_pad_buffer (filter->sinkpad, buf)); if (RMS < filter->threshold_level) filter->silent_run_length += gst_audio_duration_from_pad_buffer (filter->sinkpad, buf); @@ -267,13 +309,6 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) /* now check if we have to send the new buffer to the internal buffer cache * or to the srcpad */ if (filter->silent) { - /* we ref it before putting it in the pre_buffer */ - /* FIXME: we shouldn't probably do this, because the buffer - * arrives reffed already; the plugin should just push it - * or unref it to make it disappear */ - /* - gst_buffer_ref (buf); - */ filter->pre_buffer = g_list_append (filter->pre_buffer, buf); filter->pre_run_length += gst_audio_duration_from_pad_buffer (filter->sinkpad, buf); @@ -293,13 +328,8 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) return GST_FLOW_OK; } -static double inline -gst_cutter_16bit_ms (gint16 * data, guint num_samples) -#include "filter.func" - static double inline gst_cutter_8bit_ms (gint8 * data, guint num_samples) -#include "filter.func" - static void - gst_cutter_set_property (GObject * object, guint prop_id, +static void +gst_cutter_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstCutter *filter; @@ -309,7 +339,6 @@ gst_cutter_16bit_ms (gint16 * data, guint num_samples) switch (prop_id) { case PROP_THRESHOLD: - /* set the level */ filter->threshold_level = g_value_get_double (value); GST_DEBUG ("DEBUG: set threshold level to %f", filter->threshold_level); break; @@ -319,7 +348,8 @@ gst_cutter_16bit_ms (gint16 * data, guint num_samples) * values in dB < 0 result in values between 0 and 1 */ filter->threshold_level = pow (10, g_value_get_double (value) / 20); - GST_DEBUG ("DEBUG: set threshold level to %f", filter->threshold_level); + GST_DEBUG_OBJECT (filter, "set threshold level to %f", + filter->threshold_level); break; case PROP_RUN_LENGTH: /* set the minimum length of the silent run required */ @@ -397,6 +427,6 @@ gst_cutter_get_caps (GstPad * pad, GstCutter * filter) g_assert (caps != NULL); structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "width", &filter->width); - filter->max_sample = gst_audio_highest_sample_value (pad); + filter->max_sample = 1 << (filter->width - 1); /* signed */ filter->have_caps = TRUE; }