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f65b61fd53
../gst/speed/gstspeed.c:523:39: error: variable 'base' set but not used [-Werror,-Wunused-but-set-variable] gint64 start_value, stop_value, base; ^ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
702 lines
19 KiB
C
702 lines
19 KiB
C
/* -*- c-basic-offset: 2 -*-
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* GStreamer
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* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
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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 Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-speed
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* @title: speed
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*
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* Plays an audio stream at a different speed (by resampling the audio).
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*
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* Do not use this element. Either use the 'pitch' element, or do a seek with
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* a non-1.0 rate parameter, this will have the same effect as using the speed
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* element (but relies on the decoder/demuxer to handle this correctly, also
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* requires a fairly up-to-date gst-plugins-base, as of February 2007).
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=test.ogg ! decodebin ! audioconvert ! speed speed=1.5 ! audioconvert ! audioresample ! autoaudiosink
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* ]| Plays an .ogg file at 1.5x speed.
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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 <string.h>
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#include <math.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "gstspeed.h"
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GST_DEBUG_CATEGORY_STATIC (speed_debug);
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#define GST_CAT_DEFAULT speed_debug
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enum
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{
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PROP_0,
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PROP_SPEED
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};
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/* assumption here: sizeof (gfloat) = 4 */
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#define GST_SPEED_AUDIO_CAPS \
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"audio/x-raw, " \
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"format = {" GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ]"
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static GstStaticPadTemplate gst_speed_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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GST_STATIC_CAPS (GST_SPEED_AUDIO_CAPS)
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);
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static GstStaticPadTemplate gst_speed_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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GST_STATIC_CAPS (GST_SPEED_AUDIO_CAPS)
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);
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static void speed_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void speed_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec);
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static gboolean speed_parse_caps (GstSpeed * filter, const GstCaps * caps);
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static GstFlowReturn speed_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buf);
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static GstStateChangeReturn speed_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean speed_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean speed_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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G_DEFINE_TYPE (GstSpeed, gst_speed, GST_TYPE_ELEMENT);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (speed, "speed", GST_RANK_NONE,
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GST_TYPE_SPEED, GST_DEBUG_CATEGORY_INIT (speed_debug, "speed", 0,
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"speed element");
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);
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static gboolean
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speed_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstSpeed *filter;
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gboolean ret;
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filter = GST_SPEED (gst_pad_get_parent (pad));
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ret = speed_parse_caps (filter, caps);
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gst_object_unref (filter);
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return ret;
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}
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static gboolean
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speed_parse_caps (GstSpeed * filter, const GstCaps * caps)
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{
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g_return_val_if_fail (filter != NULL, FALSE);
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g_return_val_if_fail (caps != NULL, FALSE);
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if (!gst_audio_info_from_caps (&filter->info, caps))
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return FALSE;
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return TRUE;
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}
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static gboolean
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speed_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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GstSpeed *filter;
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gboolean ret = FALSE;
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filter = GST_SPEED (parent);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_SEEK:{
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gdouble rate;
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GstFormat format;
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GstSeekFlags flags;
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GstSeekType start_type, stop_type;
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gint64 start, stop;
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gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
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&stop_type, &stop);
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gst_event_unref (event);
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if (format != GST_FORMAT_TIME) {
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GST_DEBUG_OBJECT (filter, "only support seeks in TIME format");
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break;
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}
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if (start_type != GST_SEEK_TYPE_NONE && start != -1) {
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start *= filter->speed;
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}
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if (stop_type != GST_SEEK_TYPE_NONE && stop != -1) {
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stop *= filter->speed;
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}
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event = gst_event_new_seek (rate, format, flags, start_type, start,
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stop_type, stop);
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GST_LOG ("sending seek event: %" GST_PTR_FORMAT,
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gst_event_get_structure (event));
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ret = gst_pad_send_event (GST_PAD_PEER (filter->sinkpad), event);
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break;
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}
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default:
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ret = gst_pad_event_default (pad, parent, event);
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break;
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}
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return ret;
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}
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static gboolean
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gst_speed_convert (GstSpeed * filter, GstFormat src_format, gint64 src_value,
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GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean ret = TRUE;
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guint scale = 1;
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if (src_format == *dest_format) {
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*dest_value = src_value;
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return TRUE;
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}
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_DEFAULT:
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if (GST_AUDIO_INFO_BPF (&filter->info) == 0) {
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ret = FALSE;
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break;
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}
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*dest_value = src_value / GST_AUDIO_INFO_BPF (&filter->info);
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break;
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case GST_FORMAT_TIME:
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{
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gint byterate =
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GST_AUDIO_INFO_BPF (&filter->info) *
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GST_AUDIO_INFO_RATE (&filter->info);
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if (byterate == 0) {
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ret = FALSE;
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break;
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}
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*dest_value = src_value * GST_SECOND / byterate;
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break;
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}
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default:
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ret = FALSE;
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}
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break;
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * GST_AUDIO_INFO_BPF (&filter->info);
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break;
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case GST_FORMAT_TIME:
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if (GST_AUDIO_INFO_RATE (&filter->info) == 0) {
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ret = FALSE;
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break;
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}
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*dest_value =
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src_value * GST_SECOND / GST_AUDIO_INFO_RATE (&filter->info);
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break;
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default:
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ret = FALSE;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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scale = GST_AUDIO_INFO_BPF (&filter->info);
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/* fallthrough */
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case GST_FORMAT_DEFAULT:
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*dest_value =
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src_value * scale * GST_AUDIO_INFO_RATE (&filter->info) /
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GST_SECOND;
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break;
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default:
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ret = FALSE;
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}
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break;
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default:
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ret = FALSE;
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}
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return ret;
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}
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static gboolean
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speed_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
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{
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gboolean ret = TRUE;
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GstSpeed *filter;
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filter = GST_SPEED (parent);
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:
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{
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GstFormat format;
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GstFormat rformat = GST_FORMAT_TIME;
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gint64 cur;
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GstFormat conv_format = GST_FORMAT_TIME;
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/* save requested format */
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gst_query_parse_position (query, &format, NULL);
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/* query peer for current position in time */
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gst_query_set_position (query, GST_FORMAT_TIME, -1);
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if (!gst_pad_peer_query_position (filter->sinkpad, rformat, &cur)) {
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GST_LOG_OBJECT (filter, "TIME query on peer pad failed, trying BYTES");
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rformat = GST_FORMAT_BYTES;
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if (!gst_pad_peer_query_position (filter->sinkpad, rformat, &cur)) {
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GST_LOG_OBJECT (filter, "BYTES query on peer pad failed too");
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goto error;
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}
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}
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if (rformat == GST_FORMAT_BYTES)
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GST_LOG_OBJECT (filter,
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"peer pad returned current=%" G_GINT64_FORMAT " bytes", cur);
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else if (rformat == GST_FORMAT_TIME)
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GST_LOG_OBJECT (filter, "peer pad returned time=%" G_GINT64_FORMAT,
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cur);
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/* convert to time format */
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if (!gst_speed_convert (filter, rformat, cur, &conv_format, &cur)) {
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ret = FALSE;
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break;
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}
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/* adjust for speed factor */
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cur /= filter->speed;
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/* convert to time format */
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if (!gst_speed_convert (filter, conv_format, cur, &format, &cur)) {
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ret = FALSE;
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break;
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}
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gst_query_set_position (query, format, cur);
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GST_LOG_OBJECT (filter,
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"position query: we return %" G_GUINT64_FORMAT " (format %u)", cur,
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format);
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break;
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}
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case GST_QUERY_DURATION:
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{
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GstFormat format;
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GstFormat rformat = GST_FORMAT_TIME;
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gint64 end;
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GstFormat conv_format = GST_FORMAT_TIME;
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/* save requested format */
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gst_query_parse_duration (query, &format, NULL);
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/* query peer for total length in time */
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gst_query_set_duration (query, GST_FORMAT_TIME, -1);
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if (!gst_pad_peer_query_duration (filter->sinkpad, rformat, &end)) {
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GST_LOG_OBJECT (filter, "TIME query on peer pad failed, trying BYTES");
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rformat = GST_FORMAT_BYTES;
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if (!gst_pad_peer_query_duration (filter->sinkpad, rformat, &end)) {
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GST_LOG_OBJECT (filter, "BYTES query on peer pad failed too");
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goto error;
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}
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}
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if (rformat == GST_FORMAT_BYTES)
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GST_LOG_OBJECT (filter,
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"peer pad returned total=%" G_GINT64_FORMAT " bytes", end);
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else if (rformat == GST_FORMAT_TIME)
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GST_LOG_OBJECT (filter, "peer pad returned time=%" G_GINT64_FORMAT,
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end);
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/* convert to time format */
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if (!gst_speed_convert (filter, rformat, end, &conv_format, &end)) {
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ret = FALSE;
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break;
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}
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/* adjust for speed factor */
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end /= filter->speed;
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/* convert to time format */
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if (!gst_speed_convert (filter, conv_format, end, &format, &end)) {
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ret = FALSE;
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break;
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}
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gst_query_set_duration (query, format, end);
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GST_LOG_OBJECT (filter,
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"duration query: we return %" G_GUINT64_FORMAT " (format %u)", end,
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format);
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break;
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}
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default:
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ret = FALSE;
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break;
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}
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return ret;
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error:
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gst_object_unref (filter);
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GST_DEBUG ("error handling query");
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return FALSE;
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}
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static void
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gst_speed_class_init (GstSpeedClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = speed_set_property;
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gobject_class->get_property = speed_get_property;
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gstelement_class->change_state = speed_change_state;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SPEED,
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g_param_spec_float ("speed", "speed", "speed",
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0.1f, 40.0, 1.0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class, "Speed",
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"Filter/Effect/Audio",
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"Set speed/pitch on audio/raw streams (resampler)",
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"Andy Wingo <apwingo@eos.ncsu.edu>, "
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"Tim-Philipp Müller <tim@centricular.net>");
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_speed_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_speed_sink_template);
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}
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static void
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gst_speed_init (GstSpeed * filter)
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{
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filter->sinkpad =
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gst_pad_new_from_static_template (&gst_speed_sink_template, "sink");
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gst_pad_set_chain_function (filter->sinkpad, speed_chain);
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gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
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gst_pad_set_event_function (filter->sinkpad, speed_sink_event);
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GST_PAD_SET_PROXY_CAPS (filter->sinkpad);
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filter->srcpad =
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gst_pad_new_from_static_template (&gst_speed_src_template, "src");
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gst_pad_set_query_function (filter->srcpad, speed_src_query);
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gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
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gst_pad_set_event_function (filter->srcpad, speed_src_event);
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GST_PAD_SET_PROXY_CAPS (filter->srcpad);
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filter->offset = 0;
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filter->timestamp = 0;
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}
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static inline guint
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speed_chain_int16 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf,
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guint c, guint in_samples)
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{
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gint16 *in_data, *out_data;
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gfloat interp, lower, i_float;
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guint i, j;
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GstMapInfo in_info, out_info;
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gst_buffer_map (in_buf, &in_info, GST_MAP_READ);
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gst_buffer_map (out_buf, &out_info, GST_MAP_WRITE);
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in_data = (gint16 *) in_info.data + c;
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out_data = (gint16 *) out_info.data + c;
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lower = in_data[0];
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i_float = 0.5 * (filter->speed - 1.0);
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i = (guint) ceil (i_float);
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j = 0;
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while (i < in_samples) {
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interp = i_float - floor (i_float);
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out_data[j * GST_AUDIO_INFO_CHANNELS (&filter->info)] =
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lower * (1 - interp) +
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in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)] * interp;
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lower = in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)];
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i_float += filter->speed;
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i = (guint) ceil (i_float);
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++j;
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}
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gst_buffer_unmap (in_buf, &in_info);
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gst_buffer_unmap (out_buf, &out_info);
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return j;
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}
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static inline guint
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speed_chain_float32 (GstSpeed * filter, GstBuffer * in_buf, GstBuffer * out_buf,
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guint c, guint in_samples)
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{
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gfloat *in_data, *out_data;
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gfloat interp, lower, i_float;
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guint i, j;
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GstMapInfo in_info, out_info;
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gst_buffer_map (in_buf, &in_info, GST_MAP_WRITE);
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gst_buffer_map (out_buf, &out_info, GST_MAP_WRITE);
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in_data = (gfloat *) in_info.data + c;
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out_data = (gfloat *) out_info.data + c;
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lower = in_data[0];
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i_float = 0.5 * (filter->speed - 1.0);
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i = (guint) ceil (i_float);
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j = 0;
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while (i < in_samples) {
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interp = i_float - floor (i_float);
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out_data[j * GST_AUDIO_INFO_CHANNELS (&filter->info)] =
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lower * (1 - interp) +
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in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)] * interp;
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lower = in_data[i * GST_AUDIO_INFO_CHANNELS (&filter->info)];
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i_float += filter->speed;
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i = (guint) ceil (i_float);
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++j;
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}
|
|
gst_buffer_unmap (in_buf, &in_info);
|
|
gst_buffer_unmap (out_buf, &out_info);
|
|
return j;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
speed_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstSpeed *filter = GST_SPEED (parent);
|
|
gboolean ret = FALSE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEGMENT:{
|
|
gdouble rate;
|
|
GstFormat format;
|
|
gint64 start_value, stop_value;
|
|
const GstSegment *segment;
|
|
GstSegment seg;
|
|
|
|
gst_event_parse_segment (event, &segment);
|
|
|
|
rate = segment->rate;
|
|
format = segment->format;
|
|
start_value = segment->start;
|
|
stop_value = segment->stop;
|
|
|
|
gst_event_unref (event);
|
|
|
|
if (format != GST_FORMAT_TIME) {
|
|
GST_WARNING_OBJECT (filter, "newsegment event not in TIME format!");
|
|
break;
|
|
}
|
|
|
|
g_assert (filter->speed > 0);
|
|
|
|
if (start_value >= 0)
|
|
start_value /= filter->speed;
|
|
if (stop_value >= 0)
|
|
stop_value /= filter->speed;
|
|
|
|
/* this would only really be correct if we clipped incoming data */
|
|
filter->timestamp = start_value;
|
|
|
|
/* set to NONE so it gets reset later based on the timestamp when we have
|
|
* the samplerate */
|
|
filter->offset = GST_BUFFER_OFFSET_NONE;
|
|
|
|
gst_segment_init (&seg, GST_FORMAT_TIME);
|
|
seg.rate = rate;
|
|
seg.start = start_value;
|
|
seg.stop = stop_value;
|
|
seg.time = segment->time;
|
|
ret = gst_pad_push_event (filter->srcpad, gst_event_new_segment (&seg));
|
|
|
|
break;
|
|
}
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
ret = speed_setcaps (pad, caps);
|
|
if (!ret) {
|
|
gst_event_unref (event);
|
|
return ret;
|
|
}
|
|
}
|
|
/* Fallthrough so that the caps event gets forwarded */
|
|
default:
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
speed_chain (GstPad * pad, GstObject * parent, GstBuffer * in_buf)
|
|
{
|
|
GstBuffer *out_buf;
|
|
GstSpeed *filter = GST_SPEED (parent);
|
|
guint c, in_samples, out_samples, out_size;
|
|
GstFlowReturn flow;
|
|
gsize size;
|
|
|
|
if (G_UNLIKELY (filter->offset == GST_BUFFER_OFFSET_NONE)) {
|
|
filter->offset = gst_util_uint64_scale_int (filter->timestamp,
|
|
GST_AUDIO_INFO_RATE (&filter->info), GST_SECOND);
|
|
}
|
|
|
|
/* buffersize has to be aligned to a frame */
|
|
out_size = ceil ((gfloat) gst_buffer_get_size (in_buf) / filter->speed);
|
|
out_size = ((out_size + GST_AUDIO_INFO_BPF (&filter->info) - 1) /
|
|
GST_AUDIO_INFO_BPF (&filter->info)) * GST_AUDIO_INFO_BPF (&filter->info);
|
|
|
|
out_buf = gst_buffer_new_and_alloc (out_size);
|
|
|
|
in_samples = gst_buffer_get_size (in_buf) /
|
|
GST_AUDIO_INFO_BPF (&filter->info);
|
|
|
|
out_samples = 0;
|
|
|
|
for (c = 0; c < GST_AUDIO_INFO_CHANNELS (&filter->info); ++c) {
|
|
if (GST_AUDIO_INFO_IS_INTEGER (&filter->info))
|
|
out_samples = speed_chain_int16 (filter, in_buf, out_buf, c, in_samples);
|
|
else
|
|
out_samples =
|
|
speed_chain_float32 (filter, in_buf, out_buf, c, in_samples);
|
|
}
|
|
|
|
size = out_samples * GST_AUDIO_INFO_BPF (&filter->info);
|
|
gst_buffer_set_size (out_buf, size);
|
|
|
|
GST_BUFFER_OFFSET (out_buf) = filter->offset;
|
|
GST_BUFFER_TIMESTAMP (out_buf) = filter->timestamp;
|
|
|
|
filter->offset += size / GST_AUDIO_INFO_BPF (&filter->info);
|
|
filter->timestamp = gst_util_uint64_scale_int (filter->offset, GST_SECOND,
|
|
GST_AUDIO_INFO_RATE (&filter->info));
|
|
|
|
/* make sure it's at least nominally a perfect stream */
|
|
GST_BUFFER_DURATION (out_buf) =
|
|
filter->timestamp - GST_BUFFER_TIMESTAMP (out_buf);
|
|
flow = gst_pad_push (filter->srcpad, out_buf);
|
|
|
|
if (G_UNLIKELY (flow != GST_FLOW_OK))
|
|
GST_DEBUG_OBJECT (filter, "flow: %s", gst_flow_get_name (flow));
|
|
|
|
gst_buffer_unref (in_buf);
|
|
return flow;
|
|
}
|
|
|
|
static void
|
|
speed_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstSpeed *filter = GST_SPEED (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SPEED:
|
|
filter->speed = g_value_get_float (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
}
|
|
|
|
static void
|
|
speed_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstSpeed *filter = GST_SPEED (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SPEED:
|
|
g_value_set_float (value, filter->speed);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
speed_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstSpeed *speed = GST_SPEED (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
speed->offset = GST_BUFFER_OFFSET_NONE;
|
|
speed->timestamp = 0;
|
|
gst_audio_info_init (&speed->info);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_ELEMENT_CLASS (gst_speed_parent_class)->change_state (element,
|
|
transition);
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
return GST_ELEMENT_REGISTER (speed, plugin);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
speed,
|
|
"Set speed/pitch on audio/raw streams (resampler)",
|
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|