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Add new libsoundtouch-based pitch plugin (#331335).

Browse source 
Original commit message from CVS:
Patch by: Wouter Paeson  <wouter at kangaroot dot net>
* configure.ac:
* ext/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Add new libsoundtouch-based pitch plugin (#331335).
This commit is contained in:
Wouter Paeson 2006-03-22 14:31:47 +00:00 committed by Tim-Philipp Müller
parent 2fd4b22a83
commit 89ab6598cb
6 changed files with 838 additions and 0 deletions
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11
ChangeLog
View file

@ -1,3 +1,14 @@
2006-03-22 Tim-Philipp Müller <tim at centricular dot net>
Patch by: Wouter Paeson <wouter at kangaroot dot net>
* configure.ac:
* ext/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Add new libsoundtouch-based pitch plugin (#331335).
2006-03-21 Tim-Philipp Müller <tim at centricular dot net>
* gst/modplug/libmodplug/load_ptm.cpp:

13
configure.ac
View file

@ -540,6 +540,18 @@ GST_CHECK_FEATURE(SDL, [SDL plug-in], sdlvideosink sdlaudiosink, [
AM_PATH_SDL(, HAVE_SDL=yes, HAVE_SDL=no)
])
dnl *** soundtouch ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SOUNDTOUCH, true)
GST_CHECK_FEATURE(SOUNDTOUCH, [soundtouch plug-in], soundtouch, [
PKG_CHECK_MODULES(SOUNDTOUCH, libSoundTouch, HAVE_SOUNDTOUCH=yes, HAVE_SOUNDTOUCH=no)
AC_SUBST(SOUNDTOUCH_CFLAGS)
AC_SUBST(SOUNDTOUCH_LIBS)
if test "x$HAVE_CXX" != "xyes"; then
USE_SOUNDTOUCH=false
AC_MSG_NOTICE([Not building soundtouch plugin: no C++ compiler found])
fi
])
dnl *** swfdec ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SWFDEC, true)
GST_CHECK_FEATURE(SWFDEC, [swfdec plug-in], swfdec, [
@ -814,6 +826,7 @@ ext/bz2/Makefile
ext/directfb/Makefile
ext/faac/Makefile
ext/faad/Makefile
ext/soundtouch/Makefile
ext/wavpack/Makefile
ext/ivorbis/Makefile
ext/gsm/Makefile

8
ext/Makefile.am
View file

@ -184,6 +184,12 @@ SMOOTHWAVE_DIR=
SNDFILE_DIR=
# endif
if USE_SOUNDTOUCH
SOUNDTOUCH_DIR=soundtouch
else
SOUNDTOUCH_DIR=
endif
if USE_SWFDEC
SWFDEC_DIR=swfdec
else
@ -245,6 +251,7 @@ SUBDIRS=\
$(SHOUT_DIR) \
$(SMOOTHWAVE_DIR) \
$(SNDFILE_DIR) \
$(SOUNDTOUCH_DIR) \
$(SWFDEC_DIR) \
$(TAGLIB_DIR) \
$(TARKIN_DIR) \
@ -266,6 +273,7 @@ DIST_SUBDIRS= \
neon \
sdl \
swfdec \
soundtouch \
taglib \
wavpack \
xvid

9
ext/soundtouch/Makefile.am Normal file
View file

@ -0,0 +1,9 @@
plugin_LTLIBRARIES = libgstpitch.la
libgstpitch_la_SOURCES = gstpitch.cc
libgstpitch_la_CXXFLAGS = @GST_CFLAGS@ @GST_BASE_CFLAGS@ @SOUNDTOUCH_CFLAGS@
libgstpitch_la_LIBADD = @GST_LIBS@ @GST_BASE_LIBS@ @SOUNDTOUCH_LIBS@
libgstpitch_la_LDFLAGS = @GST_PLUGIN_LDFLAGS@
noinst_HEADERS = gstpitch.hh

709
ext/soundtouch/gstpitch.cc Normal file
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@ -0,0 +1,709 @@
/* GStreamer pitch controller element
* Copyright (C) 2006 Wouter Paesen <wouter@blue-gate.be>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include "gstpitch.hh"
#include <math.h>
#define FLOAT_SAMPLES
#include <soundtouch/SoundTouch.h>
/* wtf ?
#ifdef G_PARAM_READWRITE
# undef G_PARAM_READWRITE
#endif
#define G_PARAM_READWRITE ((GParamFlags)(G_PARAM_READABLE | G_PARAM_WRITABLE))
*/
GST_DEBUG_CATEGORY_STATIC (pitch_debug);
#define GST_CAT_DEFAULT pitch_debug
#define GST_PITCH_GET_PRIVATE(o) (o->priv)
struct _GstPitchPrivate
{
gfloat stream_time_ratio;
soundtouch::SoundTouch * st;
};
static GstElementDetails gst_pitch_details =
GST_ELEMENT_DETAILS ("Pitch controller",
"Filter/Converter/Audio",
"Control the pitch of an audio stream",
"Wouter Paesen <wouter@kangaroot.net>");
enum
{
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_RATE,
ARG_TEMPO,
ARG_PITCH,
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS( \
"audio/x-raw-float, " \
"rate = (int) [ 8000, 48000 ], " \
"channels = (int) [ 1, 2 ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"buffer-frames = (int) [ 0, MAX ]" \
)
static GstStaticPadTemplate gst_pitch_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
SUPPORTED_CAPS);
static GstStaticPadTemplate gst_pitch_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
SUPPORTED_CAPS);
static void gst_pitch_dispose (GObject * object);
static void gst_pitch_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_pitch_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_pitch_chain (GstPad * pad, GstBuffer * buffer);
static GstStateChangeReturn gst_pitch_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_pitch_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_pitch_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_pitch_src_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_pitch_get_query_types (GstPad * pad);
GST_BOILERPLATE (GstPitch, gst_pitch, GstElement, GST_TYPE_ELEMENT);
static void
gst_pitch_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_pitch_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_pitch_sink_template));
gst_element_class_set_details (gstelement_class, &gst_pitch_details);
}
static void
gst_pitch_class_init (GstPitchClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
gobject_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
gobject_class->set_property = gst_pitch_set_property;
gobject_class->get_property = gst_pitch_get_property;
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pitch_dispose);
element_class->change_state = GST_DEBUG_FUNCPTR (gst_pitch_change_state);
g_object_class_install_property (gobject_class, ARG_PITCH,
g_param_spec_float ("pitch", "Pitch",
"Audio stream pitch", 0.1, 10.0, 1.0,
(GParamFlags) G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_TEMPO,
g_param_spec_float ("tempo", "Tempo",
"Audio stream tempo", 0.1, 10.0, 1.0,
(GParamFlags) G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_RATE,
g_param_spec_float ("rate", "Rate",
"Audio stream rate", 0.1, 10.0, 1.0,
(GParamFlags) G_PARAM_READWRITE));
g_type_class_add_private (gobject_class, sizeof (GstPitchPrivate));
}
static void
gst_pitch_init (GstPitch * pitch, GstPitchClass * pitch_class)
{
pitch->priv =
G_TYPE_INSTANCE_GET_PRIVATE ((pitch), GST_TYPE_PITCH, GstPitchPrivate);
pitch->sinkpad =
gst_pad_new_from_static_template (&gst_pitch_sink_template, "sink");
gst_pad_set_chain_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pitch_chain));
gst_pad_set_event_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pitch_sink_event));
gst_pad_set_setcaps_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pitch_sink_setcaps));
gst_pad_set_getcaps_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
gst_element_add_pad (GST_ELEMENT (pitch), pitch->sinkpad);
pitch->srcpad =
gst_pad_new_from_static_template (&gst_pitch_src_template, "src");
gst_pad_set_event_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_src_event));
gst_pad_set_query_type_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_get_query_types));
gst_pad_set_query_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_src_query));
gst_pad_set_setcaps_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_sink_setcaps));
gst_pad_set_getcaps_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
gst_element_add_pad (GST_ELEMENT (pitch), pitch->srcpad);
pitch->priv->st = new soundtouch::SoundTouch ();
pitch->tempo = 1.0;
pitch->rate = 1.0;
pitch->pitch = 1.0;
pitch->next_buffer_time = 0;
pitch->next_buffer_offset = 0;
pitch->priv->st->setRate (pitch->rate);
pitch->priv->st->setTempo (pitch->tempo);
pitch->priv->st->setPitch (pitch->pitch);
pitch->priv->stream_time_ratio = 1.0;
}
static void
gst_pitch_dispose (GObject * object)
{
GstPitch *pitch = GST_PITCH (object);
if (pitch->priv->st) {
delete (pitch->priv->st);
pitch->priv->st = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_pitch_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstPitch *pitch = GST_PITCH (object);
GST_OBJECT_LOCK (pitch);
switch (prop_id) {
case ARG_TEMPO:
pitch->tempo = g_value_get_float (value);
pitch->priv->stream_time_ratio = pitch->tempo * pitch->rate;
pitch->priv->st->setTempo (pitch->tempo);
break;
case ARG_RATE:
pitch->rate = g_value_get_float (value);
pitch->priv->stream_time_ratio = pitch->tempo * pitch->rate;
pitch->priv->st->setRate (pitch->rate);
break;
case ARG_PITCH:
pitch->pitch = g_value_get_float (value);
pitch->priv->st->setPitch (pitch->pitch);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (pitch);
}
static void
gst_pitch_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstPitch *pitch = GST_PITCH (object);
GST_OBJECT_LOCK (pitch);
switch (prop_id) {
case ARG_TEMPO:
g_value_set_float (value, pitch->tempo);
break;
case ARG_RATE:
g_value_set_float (value, pitch->rate);
break;
case ARG_PITCH:
g_value_set_float (value, pitch->pitch);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (pitch);
}
static gboolean
gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstPitch *pitch;
GstPitchPrivate *priv;
GstStructure *structure;
GstPad *otherpad;
gint rate, channels;
pitch = GST_PITCH (GST_PAD_PARENT (pad));
priv = GST_PITCH_GET_PRIVATE (pitch);
otherpad = (pad == pitch->srcpad) ? pitch->sinkpad : pitch->srcpad;
if (!gst_pad_set_caps (otherpad, caps))
return FALSE;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate) ||
!gst_structure_get_int (structure, "channels", &channels)) {
return FALSE;
}
GST_OBJECT_LOCK (pitch);
pitch->samplerate = rate;
pitch->channels = channels;
/* notify the soundtouch instance of this change */
priv->st->setSampleRate (rate);
priv->st->setChannels (channels);
/* calculate sample size */
pitch->sample_size = (sizeof (gfloat) * channels);
pitch->sample_duration = gst_util_uint64_scale_int (GST_SECOND, 1, rate);
GST_OBJECT_UNLOCK (pitch);
return TRUE;
}
/* send a buffer out */
static GstFlowReturn
gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
{
gint samples;
GST_BUFFER_TIMESTAMP (buffer) = pitch->next_buffer_time;
pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
samples = GST_BUFFER_OFFSET (buffer);
GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
pitch->next_buffer_offset += samples;
GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
gst_buffer_set_caps (buffer, GST_PAD_CAPS (pitch->srcpad));
GST_LOG ("pushing buffer [%" GST_TIME_FORMAT "]-[%" GST_TIME_FORMAT
"] (%d samples)", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (pitch->next_buffer_time), samples);
return gst_pad_push (pitch->srcpad, buffer);
}
/* extract a buffer from soundtouch */
static GstBuffer *
gst_pitch_prepare_buffer (GstPitch * pitch)
{
GstPitchPrivate *priv;
guint samples;
GstBuffer *buffer;
priv = GST_PITCH_GET_PRIVATE (pitch);
GST_LOG_OBJECT (pitch, "preparing buffer");
samples = pitch->priv->st->numSamples ();
if (samples == 0)
return NULL;;
buffer = gst_buffer_new_and_alloc (samples * pitch->sample_size);
samples =
priv->st->receiveSamples ((gfloat *) GST_BUFFER_DATA (buffer), samples);
if (samples <= 0)
return NULL;
GST_BUFFER_DURATION (buffer) = samples * pitch->sample_duration;
/* temporary store samples here, to avoid having to recalculate this */
GST_BUFFER_OFFSET (buffer) = (gint64) samples;
return buffer;
}
/* process the last samples, in a later stage we should make sure no more
* samples are sent out here as strictly necessary, because soundtouch could
* append zero samples, which could disturb looping. */
static GstFlowReturn
gst_pitch_flush_buffer (GstPitch * pitch)
{
GstBuffer *buffer;
GST_DEBUG_OBJECT (pitch, "flushing buffer");
if (pitch->next_buffer_offset == 0)
return GST_FLOW_OK;
pitch->priv->st->flush ();
buffer = gst_pitch_prepare_buffer (pitch);
if (!buffer)
return GST_FLOW_OK;
return gst_pitch_forward_buffer (pitch, buffer);
}
static gboolean
gst_pitch_src_event (GstPad * pad, GstEvent * event)
{
GstPitch *pitch;
gboolean res;
pitch = GST_PITCH (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
/* transform the event upstream, according to the playback rate */
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gfloat stream_time_ratio;
GST_OBJECT_LOCK (pitch);
stream_time_ratio = pitch->priv->stream_time_ratio;
GST_OBJECT_UNLOCK (pitch);
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
cur = (gint64) (cur * stream_time_ratio);
stop = (gint64) (stop * stream_time_ratio);
gst_event_unref (event);
event = gst_event_new_seek (rate, format, flags,
cur_type, cur, stop_type, stop);
res = gst_pad_event_default (pad, event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (pitch);
return res;
}
/* generic convert function based on caps, no rate
* used here
*/
static gboolean
gst_pitch_convert (GstPitch * pitch,
GstFormat src_format, gint64 src_value,
GstFormat * dst_format, gint64 * dst_value)
{
gboolean res = TRUE;
GstClockTime sample_duration;
guint sample_size;
g_return_val_if_fail (dst_format && dst_value, FALSE);
GST_OBJECT_LOCK (pitch);
sample_duration = pitch->sample_duration;
sample_size = pitch->sample_size;
GST_OBJECT_UNLOCK (pitch);
if (sample_size == 0 || sample_duration == 0 ||
sample_duration == GST_CLOCK_TIME_NONE) {
return FALSE;
}
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dst_format) {
case GST_FORMAT_TIME:
*dst_value = src_value / sample_size;
*dst_value *= sample_duration;
break;
case GST_FORMAT_DEFAULT:
*dst_value = src_value / sample_size;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_TIME:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value = src_value / sample_duration;
*dst_value *= sample_size;
break;
case GST_FORMAT_DEFAULT:
*dst_value = src_value / sample_duration;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value = src_value * sample_size;
break;
case GST_FORMAT_TIME:
*dst_value = src_value * sample_duration;
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
return res;
}
static const GstQueryType *
gst_pitch_get_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
GST_QUERY_NONE
};
return types;
}
static gboolean
gst_pitch_src_query (GstPad * pad, GstQuery * query)
{
GstPitch *pitch;
gboolean res = FALSE;
gfloat stream_time_ratio;
gint64 next_buffer_offset;
pitch = GST_PITCH (gst_pad_get_parent (pad));
GST_LOG ("%s query", GST_QUERY_TYPE_NAME (query));
GST_OBJECT_LOCK (pitch);
stream_time_ratio = pitch->priv->stream_time_ratio;
next_buffer_offset = pitch->next_buffer_offset;
GST_OBJECT_UNLOCK (pitch);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:{
GstFormat format;
gint64 duration;
if (!gst_pad_query_default (pad, query)) {
GST_DEBUG_OBJECT (pitch, "upstream provided no duration");
break;
}
gst_query_parse_duration (query, &format, &duration);
if (format != GST_FORMAT_TIME && format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (pitch, "not TIME or DEFAULT format");
break;
}
GST_LOG_OBJECT (pitch, "upstream duration: %" G_GINT64_FORMAT, duration);
duration = (gint64) (duration / stream_time_ratio);
GST_LOG_OBJECT (pitch, "our duration: %" G_GINT64_FORMAT, duration);
gst_query_set_duration (query, format, duration);
res = TRUE;
break;
}
case GST_QUERY_POSITION:{
GstFormat dst_format;
gint64 dst_value;
gst_query_parse_position (query, &dst_format, &dst_value);
if (dst_format != GST_FORMAT_TIME && dst_format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (pitch, "not TIME or DEFAULT format");
break;
}
if (dst_format != GST_FORMAT_DEFAULT) {
res = gst_pitch_convert (pitch, GST_FORMAT_DEFAULT,
next_buffer_offset, &dst_format, &dst_value);
} else {
dst_value = next_buffer_offset;
res = TRUE;
}
if (res) {
GST_LOG_OBJECT (pitch, "our position: %" G_GINT64_FORMAT, dst_value);
gst_query_set_position (query, dst_format, dst_value);
}
break;
}
case GST_QUERY_CONVERT:{
GstFormat src_format, dst_format;
gint64 src_value, dst_value;
gst_query_parse_convert (query, &src_format, &src_value,
&dst_format, NULL);
res = gst_pitch_convert (pitch, src_format, src_value,
&dst_format, &dst_value);
if (res) {
gst_query_set_convert (query, src_format, src_value,
dst_format, dst_value);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (pitch);
return res;
}
static gboolean
gst_pitch_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstPitch *pitch;
pitch = GST_PITCH (gst_pad_get_parent (pad));
GST_LOG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
case GST_EVENT_EOS:
gst_pitch_flush_buffer (pitch);
break;
default:
break;
}
/* and forward it */
res = gst_pad_event_default (pad, event);
gst_object_unref (pitch);
return res;
}
static GstFlowReturn
gst_pitch_chain (GstPad * pad, GstBuffer * buffer)
{
GstPitch *pitch;
GstPitchPrivate *priv;
pitch = GST_PITCH (GST_PAD_PARENT (pad));
priv = GST_PITCH_GET_PRIVATE (pitch);
/* push the received samples on the soundtouch buffer */
GST_LOG_OBJECT (pitch, "incoming buffer (%d samples)",
(gint) (GST_BUFFER_SIZE (buffer) / pitch->sample_size));
priv->st->putSamples ((gfloat *) GST_BUFFER_DATA (buffer),
GST_BUFFER_SIZE (buffer) / pitch->sample_size);
gst_buffer_unref (buffer);
/* and try to extract some samples from the soundtouch buffer */
if (!priv->st->isEmpty ()) {
GstBuffer *out_buffer;
out_buffer = gst_pitch_prepare_buffer (pitch);
return gst_pitch_forward_buffer (pitch, out_buffer);
}
return GST_FLOW_OK;
}
static GstStateChangeReturn
gst_pitch_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPitch *pitch = GST_PITCH (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
pitch->next_buffer_time = 0;
pitch->next_buffer_offset = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = parent_class->change_state (element, transition);
if (ret != GST_STATE_CHANGE_SUCCESS)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (pitch_debug, "pitch", 0,
"audio pitch control element");
return gst_element_register (plugin, "pitch", GST_RANK_NONE, GST_TYPE_PITCH);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"soundtouch",
"Audio Pitch Controller",
plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)

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ext/soundtouch/gstpitch.hh Normal file
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/* GStreamer pitch controller element
* Copyright (C) 2006 Wouter Paesen <wouter@blue-gate.be>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifndef __GST_PITCH_H__
#define __GST_PITCH_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#define GST_TYPE_PITCH \
(gst_pitch_get_type())
#define GST_PITCH(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PITCH,GstPitch))
#define GST_PITCH_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PITCH,GstPitchClass))
#define GST_IS_PITCH(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PITCH))
#define GST_IS_PITCH_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PITCH))
typedef struct _GstPitch GstPitch;
typedef struct _GstPitchClass GstPitchClass;
typedef struct _GstPitchPrivate GstPitchPrivate;
struct _GstPitch
{
GstElement element;
GstPad *srcpad;
GstPad *sinkpad;
/* parameter values */
gfloat tempo; /* time stretch
* change the duration, without affecting the pitch
* > 1 makes the stream shorter
*/
gfloat rate; /* change playback rate
* resample
* > 1 makes the stream shorter
*/
gfloat pitch; /* change pitch
* change the pitch without affecting the
* duration, stream length doesn't change
*/
/* values extracted from caps */
gint samplerate; /* samplerate */
gint channels; /* number of audio channels */
gsize sample_size; /* number of bytes for a single sample */
GstClockTime sample_duration; /* time for 1 sample */
/* stream tracking */
GstClockTime next_buffer_time;
gint64 next_buffer_offset;
GstPitchPrivate *priv;
};
struct _GstPitchClass
{
GstElementClass parent_class;
};
GType gst_pitch_get_type (void);
G_END_DECLS
#endif /* __GST_PITCH_H__ */