mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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bb6c23aa43
Original commit message from CVS: * configure.ac: Require core CVS. This is implicit in the -base CVS requirement already, so we might just well spell it out. Also, we do need at least 0.10.14 for gst_element_class_set_details_simple(). Make check for gmyth a bit more restrictive so things don't break if the next version changes API. * ext/alsaspdif/alsaspdifsink.c: Work around alsa alloca macros triggering 'always evaluates to true' warnings with gcc-4.2 and fix compilation with gcc-4.2. Also don't leak the device string. * ext/mpeg2enc/gstmpeg2enc.cc: * ext/soundtouch/gstpitch.cc: * gst/modplug/gstmodplug.cc: Fix compilation with g++4.2 and -Wall -Werror (also needs plugin define fix from core CVS). Fixes #462737.
710 lines
19 KiB
C++
710 lines
19 KiB
C++
/* GStreamer pitch controller element
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* Copyright (C) 2006 Wouter Paesen <wouter@blue-gate.be>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include <gst/gst.h>
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#include <gst/controller/gstcontroller.h>
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#include "gstpitch.hh"
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#include <math.h>
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#define FLOAT_SAMPLES
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#include <soundtouch/SoundTouch.h>
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/* wtf ?
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#ifdef G_PARAM_READWRITE
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# undef G_PARAM_READWRITE
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#endif
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#define G_PARAM_READWRITE ((GParamFlags)(G_PARAM_READABLE | G_PARAM_WRITABLE))
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*/
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GST_DEBUG_CATEGORY_STATIC (pitch_debug);
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#define GST_CAT_DEFAULT pitch_debug
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#define GST_PITCH_GET_PRIVATE(o) (o->priv)
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struct _GstPitchPrivate
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{
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gfloat stream_time_ratio;
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soundtouch::SoundTouch * st;
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};
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enum
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{
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ARG_0,
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ARG_RATE,
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ARG_TEMPO,
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ARG_PITCH
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};
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#define SUPPORTED_CAPS \
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GST_STATIC_CAPS( \
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"audio/x-raw-float, " \
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"rate = (int) [ 8000, 48000 ], " \
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"channels = (int) [ 1, 2 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32, " \
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"buffer-frames = (int) [ 0, MAX ]" \
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)
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static GstStaticPadTemplate gst_pitch_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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SUPPORTED_CAPS);
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static GstStaticPadTemplate gst_pitch_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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SUPPORTED_CAPS);
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static void gst_pitch_dispose (GObject * object);
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static void gst_pitch_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_pitch_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_pitch_chain (GstPad * pad, GstBuffer * buffer);
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static GstStateChangeReturn gst_pitch_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_pitch_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_pitch_src_event (GstPad * pad, GstEvent * event);
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static gboolean gst_pitch_src_query (GstPad * pad, GstQuery * query);
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static const GstQueryType *gst_pitch_get_query_types (GstPad * pad);
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GST_BOILERPLATE (GstPitch, gst_pitch, GstElement, GST_TYPE_ELEMENT);
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static void
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gst_pitch_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_pitch_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_pitch_sink_template));
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gst_element_class_set_details_simple (gstelement_class, "Pitch controller",
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"Filter/Converter/Audio", "Control the pitch of an audio stream",
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"Wouter Paesen <wouter@kangaroot.net>");
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}
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static void
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gst_pitch_class_init (GstPitchClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *element_class;
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gobject_class = G_OBJECT_CLASS (klass);
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element_class = GST_ELEMENT_CLASS (klass);
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gobject_class->set_property = gst_pitch_set_property;
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gobject_class->get_property = gst_pitch_get_property;
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pitch_dispose);
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_pitch_change_state);
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g_object_class_install_property (gobject_class, ARG_PITCH,
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g_param_spec_float ("pitch", "Pitch",
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"Audio stream pitch", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)));
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g_object_class_install_property (gobject_class, ARG_TEMPO,
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g_param_spec_float ("tempo", "Tempo",
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"Audio stream tempo", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)));
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g_object_class_install_property (gobject_class, ARG_RATE,
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g_param_spec_float ("rate", "Rate",
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"Audio stream rate", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)));
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g_type_class_add_private (gobject_class, sizeof (GstPitchPrivate));
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}
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static void
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gst_pitch_init (GstPitch * pitch, GstPitchClass * pitch_class)
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{
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pitch->priv =
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G_TYPE_INSTANCE_GET_PRIVATE ((pitch), GST_TYPE_PITCH, GstPitchPrivate);
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pitch->sinkpad =
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gst_pad_new_from_static_template (&gst_pitch_sink_template, "sink");
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gst_pad_set_chain_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pitch_chain));
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gst_pad_set_event_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pitch_sink_event));
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gst_pad_set_setcaps_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pitch_sink_setcaps));
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gst_pad_set_getcaps_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
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gst_element_add_pad (GST_ELEMENT (pitch), pitch->sinkpad);
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pitch->srcpad =
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gst_pad_new_from_static_template (&gst_pitch_src_template, "src");
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gst_pad_set_event_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_src_event));
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gst_pad_set_query_type_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_get_query_types));
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gst_pad_set_query_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_src_query));
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gst_pad_set_setcaps_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_sink_setcaps));
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gst_pad_set_getcaps_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
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gst_element_add_pad (GST_ELEMENT (pitch), pitch->srcpad);
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pitch->priv->st = new soundtouch::SoundTouch ();
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pitch->tempo = 1.0;
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pitch->rate = 1.0;
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pitch->pitch = 1.0;
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pitch->next_buffer_time = 0;
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pitch->next_buffer_offset = 0;
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pitch->priv->st->setRate (pitch->rate);
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pitch->priv->st->setTempo (pitch->tempo);
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pitch->priv->st->setPitch (pitch->pitch);
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pitch->priv->stream_time_ratio = 1.0;
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}
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static void
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gst_pitch_dispose (GObject * object)
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{
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GstPitch *pitch = GST_PITCH (object);
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if (pitch->priv->st) {
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delete (pitch->priv->st);
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pitch->priv->st = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_pitch_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstPitch *pitch = GST_PITCH (object);
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GST_OBJECT_LOCK (pitch);
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switch (prop_id) {
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case ARG_TEMPO:
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pitch->tempo = g_value_get_float (value);
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pitch->priv->stream_time_ratio = pitch->tempo * pitch->rate;
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pitch->priv->st->setTempo (pitch->tempo);
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break;
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case ARG_RATE:
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pitch->rate = g_value_get_float (value);
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pitch->priv->stream_time_ratio = pitch->tempo * pitch->rate;
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pitch->priv->st->setRate (pitch->rate);
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break;
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case ARG_PITCH:
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pitch->pitch = g_value_get_float (value);
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pitch->priv->st->setPitch (pitch->pitch);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (pitch);
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}
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static void
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gst_pitch_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstPitch *pitch = GST_PITCH (object);
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GST_OBJECT_LOCK (pitch);
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switch (prop_id) {
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case ARG_TEMPO:
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g_value_set_float (value, pitch->tempo);
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break;
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case ARG_RATE:
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g_value_set_float (value, pitch->rate);
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break;
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case ARG_PITCH:
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g_value_set_float (value, pitch->pitch);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (pitch);
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}
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static gboolean
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gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstPitch *pitch;
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GstPitchPrivate *priv;
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GstStructure *structure;
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GstPad *otherpad;
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gint rate, channels;
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pitch = GST_PITCH (GST_PAD_PARENT (pad));
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priv = GST_PITCH_GET_PRIVATE (pitch);
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otherpad = (pad == pitch->srcpad) ? pitch->sinkpad : pitch->srcpad;
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if (!gst_pad_set_caps (otherpad, caps))
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return FALSE;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &rate) ||
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!gst_structure_get_int (structure, "channels", &channels)) {
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return FALSE;
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}
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GST_OBJECT_LOCK (pitch);
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pitch->samplerate = rate;
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pitch->channels = channels;
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/* notify the soundtouch instance of this change */
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priv->st->setSampleRate (rate);
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priv->st->setChannels (channels);
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/* calculate sample size */
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pitch->sample_size = (sizeof (gfloat) * channels);
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pitch->sample_duration = gst_util_uint64_scale_int (GST_SECOND, 1, rate);
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GST_OBJECT_UNLOCK (pitch);
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return TRUE;
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}
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/* send a buffer out */
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static GstFlowReturn
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gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
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{
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gint samples;
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GST_BUFFER_TIMESTAMP (buffer) = pitch->next_buffer_time;
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pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
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samples = GST_BUFFER_OFFSET (buffer);
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GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
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pitch->next_buffer_offset += samples;
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GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
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gst_buffer_set_caps (buffer, GST_PAD_CAPS (pitch->srcpad));
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GST_LOG ("pushing buffer [%" GST_TIME_FORMAT "]-[%" GST_TIME_FORMAT
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"] (%d samples)", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
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GST_TIME_ARGS (pitch->next_buffer_time), samples);
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return gst_pad_push (pitch->srcpad, buffer);
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}
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/* extract a buffer from soundtouch */
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static GstBuffer *
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gst_pitch_prepare_buffer (GstPitch * pitch)
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{
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GstPitchPrivate *priv;
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guint samples;
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GstBuffer *buffer;
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priv = GST_PITCH_GET_PRIVATE (pitch);
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GST_LOG_OBJECT (pitch, "preparing buffer");
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samples = pitch->priv->st->numSamples ();
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if (samples == 0)
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return NULL;;
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buffer = gst_buffer_new_and_alloc (samples * pitch->sample_size);
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samples =
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priv->st->receiveSamples ((gfloat *) GST_BUFFER_DATA (buffer), samples);
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if (samples <= 0)
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return NULL;
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GST_BUFFER_DURATION (buffer) = samples * pitch->sample_duration;
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/* temporary store samples here, to avoid having to recalculate this */
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GST_BUFFER_OFFSET (buffer) = (gint64) samples;
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return buffer;
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}
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/* process the last samples, in a later stage we should make sure no more
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* samples are sent out here as strictly necessary, because soundtouch could
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* append zero samples, which could disturb looping. */
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static GstFlowReturn
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gst_pitch_flush_buffer (GstPitch * pitch, gboolean send)
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{
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GstBuffer *buffer;
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GST_DEBUG_OBJECT (pitch, "flushing buffer");
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if (pitch->next_buffer_offset == 0)
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return GST_FLOW_OK;
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pitch->priv->st->flush ();
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if (!send)
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return GST_FLOW_OK;
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buffer = gst_pitch_prepare_buffer (pitch);
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if (!buffer)
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return GST_FLOW_OK;
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return gst_pitch_forward_buffer (pitch, buffer);
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}
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static gboolean
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gst_pitch_src_event (GstPad * pad, GstEvent * event)
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{
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GstPitch *pitch;
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gboolean res;
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pitch = GST_PITCH (gst_pad_get_parent (pad));
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GST_DEBUG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_SEEK:{
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/* transform the event upstream, according to the playback rate */
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gdouble rate;
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GstFormat format;
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GstSeekFlags flags;
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GstSeekType cur_type, stop_type;
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gint64 cur, stop;
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gfloat stream_time_ratio;
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GST_OBJECT_LOCK (pitch);
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stream_time_ratio = pitch->priv->stream_time_ratio;
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GST_OBJECT_UNLOCK (pitch);
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gst_event_parse_seek (event, &rate, &format, &flags,
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&cur_type, &cur, &stop_type, &stop);
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cur = (gint64) (cur * stream_time_ratio);
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stop = (gint64) (stop * stream_time_ratio);
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gst_event_unref (event);
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event = gst_event_new_seek (rate, format, flags,
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cur_type, cur, stop_type, stop);
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res = gst_pad_event_default (pad, event);
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break;
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}
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default:
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res = gst_pad_event_default (pad, event);
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break;
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}
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gst_object_unref (pitch);
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return res;
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}
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/* generic convert function based on caps, no rate
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* used here
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*/
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static gboolean
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gst_pitch_convert (GstPitch * pitch,
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GstFormat src_format, gint64 src_value,
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GstFormat * dst_format, gint64 * dst_value)
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{
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gboolean res = TRUE;
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GstClockTime sample_duration;
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guint sample_size;
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g_return_val_if_fail (dst_format && dst_value, FALSE);
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GST_OBJECT_LOCK (pitch);
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sample_duration = pitch->sample_duration;
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sample_size = pitch->sample_size;
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GST_OBJECT_UNLOCK (pitch);
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if (sample_size == 0 || sample_duration == 0 ||
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sample_duration == GST_CLOCK_TIME_NONE) {
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return FALSE;
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}
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dst_format) {
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case GST_FORMAT_TIME:
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*dst_value = src_value / sample_size;
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*dst_value *= sample_duration;
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break;
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case GST_FORMAT_DEFAULT:
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*dst_value = src_value / sample_size;
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break;
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default:
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res = FALSE;
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break;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dst_format) {
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case GST_FORMAT_BYTES:
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*dst_value = src_value / sample_duration;
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*dst_value *= sample_size;
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break;
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case GST_FORMAT_DEFAULT:
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*dst_value = src_value / sample_duration;
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break;
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default:
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res = FALSE;
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break;
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}
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break;
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case GST_FORMAT_DEFAULT:
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switch (*dst_format) {
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case GST_FORMAT_BYTES:
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*dst_value = src_value * sample_size;
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break;
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|
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_FLUSH_STOP:
|
|
gst_pitch_flush_buffer (pitch, FALSE);
|
|
break;
|
|
case GST_EVENT_EOS:
|
|
gst_pitch_flush_buffer (pitch, TRUE);
|
|
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);
|
|
|
|
gst_object_sync_values (G_OBJECT (pitch), pitch->next_buffer_time);
|
|
|
|
/* 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_controller_init (NULL, NULL);
|
|
|
|
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_NAME, GST_PACKAGE_ORIGIN)
|