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Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_picture_frame): We require a -base more recent than 0.10.9, so it's safe to use GST_TYPE_TAG_IMAGE_TYPE unconditionally now. * ext/dv/gstdvdec.c: (gst_dvdec_sink_event): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event): Use _newsegment_full() now that we depend on a recent enough core. * gst/wavparse/gstwavparse.c: Remove cruft that we don't need any longer now that we depend on a recent enough -base.
2015 lines
57 KiB
C
2015 lines
57 KiB
C
/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-wavparse
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*
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* <refsect2>
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* <para>
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* Parse a .wav file into raw or compressed audio.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
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* </programlisting>
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* Read a wav file and output to the soundcard using the ALSA element. The
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* wav file is assumed to contain raw uncompressed samples.
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* </para>
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* <para>
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* <programlisting>
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* gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
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* </programlisting>
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* Stream data from
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-03-03 (0.10.3)
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*/
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/*
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* TODO:
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* http://replaygain.hydrogenaudio.org/file_format_wav.html
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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 "gstwavparse.h"
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#include "gst/riff/riff-ids.h"
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#include "gst/riff/riff-media.h"
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#include <gst/gst-i18n-plugin.h>
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GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
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#define GST_CAT_DEFAULT (wavparse_debug)
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static void gst_wavparse_base_init (gpointer g_class);
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static void gst_wavparse_class_init (GstWavParseClass * klass);
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static void gst_wavparse_init (GstWavParse * wavparse);
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static void gst_wavparse_dispose (GObject * object);
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static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
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static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
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gboolean active);
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static gboolean gst_wavparse_send_event (GstElement * element,
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GstEvent * event);
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static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
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static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
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static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
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static gboolean gst_wavparse_pad_convert (GstPad * pad,
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GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
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static void gst_wavparse_loop (GstPad * pad);
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static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
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static void gst_wavparse_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static const GstElementDetails gst_wavparse_details =
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GST_ELEMENT_DETAILS ("WAV audio demuxer",
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"Codec/Demuxer/Audio",
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"Parse a .wav file into raw audio",
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"Erik Walthinsen <omega@cse.ogi.edu>");
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static GstStaticPadTemplate sink_template_factory =
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GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-wav")
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);
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/* the pad is marked a sometimes and is added to the element when the
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* exact type is known. This makes it much easier for a static autoplugger
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* to connect the right decoder when needed.
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*/
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static GstStaticPadTemplate src_template_factory =
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GST_STATIC_PAD_TEMPLATE ("wavparse_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) little_endian, "
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"signed = (boolean) { true, false }, "
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"width = (int) { 8, 16, 24, 32 }, "
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"depth = (int) { 8, 16, 24, 32 }, "
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"rate = (int) [ 8000, 96000 ], "
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"channels = (int) [ 1, 8 ]; "
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"audio/ms-gsm; "
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"audio/mpeg, "
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"mpegversion = (int) 1, "
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"layer = (int) [ 1, 3 ], "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-alaw, "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-mulaw, "
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"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
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"audio/x-adpcm, "
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"layout = (string) microsoft, "
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"block_align = (int) [ 1, 8192 ], "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-adpcm, "
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"layout = (string) dvi, "
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"block_align = (int) [ 1, 8192 ], "
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"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
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"audio/x-vnd.sony.atrac3;"
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"audio/x-dts;" "audio/x-wma, " "wmaversion = (int) [ 1, 2 ]")
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);
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static GstElementClass *parent_class = NULL;
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GType
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gst_wavparse_get_type (void)
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{
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static GType wavparse_type = 0;
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if (!wavparse_type) {
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static const GTypeInfo wavparse_info = {
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sizeof (GstWavParseClass),
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gst_wavparse_base_init,
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NULL,
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(GClassInitFunc) gst_wavparse_class_init,
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NULL,
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NULL,
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sizeof (GstWavParse),
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0,
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(GInstanceInitFunc) gst_wavparse_init,
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};
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wavparse_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse",
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&wavparse_info, 0);
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}
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return wavparse_type;
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}
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static void
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gst_wavparse_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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/* register src pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template_factory));
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gst_element_class_set_details (element_class, &gst_wavparse_details);
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}
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static void
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gst_wavparse_class_init (GstWavParseClass * klass)
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{
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GstElementClass *gstelement_class;
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GObjectClass *object_class;
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gstelement_class = (GstElementClass *) klass;
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object_class = (GObjectClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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object_class->get_property = gst_wavparse_get_property;
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object_class->dispose = gst_wavparse_dispose;
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gstelement_class->change_state = gst_wavparse_change_state;
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gstelement_class->send_event = gst_wavparse_send_event;
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}
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static void
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gst_wavparse_dispose (GObject * object)
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{
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GstWavParse *wav;
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GST_DEBUG ("WAV: Dispose\n");
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wav = GST_WAVPARSE (object);
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if (wav->adapter) {
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g_object_unref (wav->adapter);
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wav->adapter = 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_wavparse_reset (GstWavParse * wavparse)
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{
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wavparse->state = GST_WAVPARSE_START;
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/* These will all be set correctly in the fmt chunk */
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wavparse->depth = 0;
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wavparse->rate = 0;
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wavparse->width = 0;
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wavparse->channels = 0;
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wavparse->blockalign = 0;
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wavparse->bps = 0;
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wavparse->fact = 0;
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wavparse->offset = 0;
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wavparse->end_offset = 0;
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wavparse->dataleft = 0;
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wavparse->datasize = 0;
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wavparse->datastart = 0;
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wavparse->got_fmt = FALSE;
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wavparse->first = TRUE;
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if (wavparse->seek_event)
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gst_event_unref (wavparse->seek_event);
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wavparse->seek_event = NULL;
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/* we keep the segment info in time */
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gst_segment_init (&wavparse->segment, GST_FORMAT_TIME);
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}
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static void
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gst_wavparse_init (GstWavParse * wavparse)
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{
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gst_wavparse_reset (wavparse);
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/* sink */
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wavparse->sinkpad =
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gst_pad_new_from_static_template (&sink_template_factory, "sink");
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gst_pad_set_activate_function (wavparse->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
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gst_pad_set_activatepull_function (wavparse->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
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gst_pad_set_chain_function (wavparse->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_chain));
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
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/* src, will be created later */
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wavparse->srcpad = NULL;
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}
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static void
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gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
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{
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if (wavparse->srcpad) {
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gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
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wavparse->srcpad = NULL;
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}
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}
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static void
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gst_wavparse_create_sourcepad (GstWavParse * wavparse)
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{
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/* destroy previous one */
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gst_wavparse_destroy_sourcepad (wavparse);
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/* source */
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wavparse->srcpad =
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gst_pad_new_from_static_template (&src_template_factory, "src");
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gst_pad_use_fixed_caps (wavparse->srcpad);
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gst_pad_set_query_type_function (wavparse->srcpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));
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gst_pad_set_query_function (wavparse->srcpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
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gst_pad_set_event_function (wavparse->srcpad,
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GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
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GST_DEBUG_OBJECT (wavparse, "srcpad created");
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}
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static void
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gst_wavparse_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstWavParse *wavparse;
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wavparse = GST_WAVPARSE (object);
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switch (prop_id) {
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default:
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break;
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}
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}
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#if 0
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static void
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gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
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{
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guint32 got_bytes;
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GstByteStream *bs = wavparse->bs;
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gst_riff_chunk *temp_chunk, chunk;
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guint8 *tempdata;
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struct _gst_riff_labl labl, *temp_labl;
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struct _gst_riff_ltxt ltxt, *temp_ltxt;
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struct _gst_riff_note note, *temp_note;
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char *label_name;
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GstProps *props;
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GstPropsEntry *entry;
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GstCaps *new_caps;
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GList *caps = NULL;
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props = wavparse->metadata->properties;
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while (len > 0) {
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
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if (got_bytes != sizeof (gst_riff_chunk)) {
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return;
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}
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temp_chunk = (gst_riff_chunk *) tempdata;
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chunk.id = GUINT32_FROM_LE (temp_chunk->id);
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chunk.size = GUINT32_FROM_LE (temp_chunk->size);
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if (chunk.size == 0) {
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gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
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len -= sizeof (gst_riff_chunk);
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continue;
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}
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switch (chunk.id) {
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case GST_RIFF_adtl_labl:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_labl));
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if (got_bytes != sizeof (struct _gst_riff_labl)) {
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return;
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}
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temp_labl = (struct _gst_riff_labl *) tempdata;
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labl.id = GUINT32_FROM_LE (temp_labl->id);
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labl.size = GUINT32_FROM_LE (temp_labl->size);
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labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
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gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
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len -= sizeof (struct _gst_riff_labl);
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got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
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if (got_bytes != labl.size - 4) {
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return;
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}
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label_name = (char *) tempdata;
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gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
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len -= (((labl.size - 4) + 1) & ~1);
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new_caps = gst_caps_new ("label",
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"application/x-gst-metadata",
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gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
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"name", G_TYPE_STRING (label_name), NULL));
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if (gst_props_get (props, "labels", &caps, NULL)) {
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caps = g_list_append (caps, new_caps);
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} else {
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caps = g_list_append (NULL, new_caps);
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entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
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gst_props_add_entry (props, entry);
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}
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break;
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case GST_RIFF_adtl_ltxt:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_ltxt));
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if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
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return;
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}
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temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
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ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
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ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
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ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
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ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
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ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
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ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
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ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
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ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
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ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
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gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
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len -= sizeof (struct _gst_riff_ltxt);
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if (ltxt.size - 20 > 0) {
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got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
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if (got_bytes != ltxt.size - 20) {
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return;
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}
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gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
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len -= (((ltxt.size - 20) + 1) & ~1);
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label_name = (char *) tempdata;
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} else {
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label_name = "";
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}
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new_caps = gst_caps_new ("ltxt",
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"application/x-gst-metadata",
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gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
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"name", G_TYPE_STRING (label_name),
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"length", G_TYPE_INT (ltxt.length), NULL));
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if (gst_props_get (props, "ltxts", &caps, NULL)) {
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caps = g_list_append (caps, new_caps);
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} else {
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caps = g_list_append (NULL, new_caps);
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entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
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gst_props_add_entry (props, entry);
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}
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break;
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case GST_RIFF_adtl_note:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_note));
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if (got_bytes != sizeof (struct _gst_riff_note)) {
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return;
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}
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|
|
temp_note = (struct _gst_riff_note *) tempdata;
|
|
note.id = GUINT32_FROM_LE (temp_note->id);
|
|
note.size = GUINT32_FROM_LE (temp_note->size);
|
|
note.identifier = GUINT32_FROM_LE (temp_note->identifier);
|
|
|
|
gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
|
|
len -= sizeof (struct _gst_riff_note);
|
|
|
|
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
|
|
if (got_bytes != note.size - 4) {
|
|
return;
|
|
}
|
|
|
|
gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
|
|
len -= (((note.size - 4) + 1) & ~1);
|
|
|
|
label_name = (char *) tempdata;
|
|
|
|
new_caps = gst_caps_new ("note",
|
|
"application/x-gst-metadata",
|
|
gst_props_new ("identifier", G_TYPE_INT (note.identifier),
|
|
"name", G_TYPE_STRING (label_name), NULL));
|
|
|
|
if (gst_props_get (props, "notes", &caps, NULL)) {
|
|
caps = g_list_append (caps, new_caps);
|
|
} else {
|
|
caps = g_list_append (NULL, new_caps);
|
|
|
|
entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
|
|
gst_props_add_entry (props, entry);
|
|
}
|
|
|
|
break;
|
|
|
|
default:
|
|
g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
|
|
GST_FOURCC_ARGS (chunk.id));
|
|
return;
|
|
}
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (wavparse), "metadata");
|
|
}
|
|
|
|
static void
|
|
gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
|
|
{
|
|
guint32 got_bytes;
|
|
GstByteStream *bs = wavparse->bs;
|
|
struct _gst_riff_cue *temp_cue, cue;
|
|
struct _gst_riff_cuepoints *points;
|
|
guint8 *tempdata;
|
|
int i;
|
|
GList *cues = NULL;
|
|
GstPropsEntry *entry;
|
|
|
|
while (len > 0) {
|
|
int required;
|
|
|
|
got_bytes =
|
|
gst_bytestream_peek_bytes (bs, &tempdata,
|
|
sizeof (struct _gst_riff_cue));
|
|
temp_cue = (struct _gst_riff_cue *) tempdata;
|
|
|
|
/* fixup for our big endian friends */
|
|
cue.id = GUINT32_FROM_LE (temp_cue->id);
|
|
cue.size = GUINT32_FROM_LE (temp_cue->size);
|
|
cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
|
|
|
|
gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
|
|
if (got_bytes != sizeof (struct _gst_riff_cue)) {
|
|
return;
|
|
}
|
|
|
|
len -= sizeof (struct _gst_riff_cue);
|
|
|
|
/* -4 because cue.size contains the cuepoints size
|
|
and we've already flushed that out of the system */
|
|
required = cue.size - 4;
|
|
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
|
|
gst_bytestream_flush (bs, ((required) + 1) & ~1);
|
|
if (got_bytes != required) {
|
|
return;
|
|
}
|
|
|
|
len -= (((cue.size - 4) + 1) & ~1);
|
|
|
|
/* now we have an array of struct _gst_riff_cuepoints in tempdata */
|
|
points = (struct _gst_riff_cuepoints *) tempdata;
|
|
|
|
for (i = 0; i < cue.cuepoints; i++) {
|
|
GstCaps *caps;
|
|
|
|
caps = gst_caps_new ("cues",
|
|
"application/x-gst-metadata",
|
|
gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
|
|
"position", G_TYPE_INT (points[i].offset), NULL));
|
|
cues = g_list_append (cues, caps);
|
|
}
|
|
|
|
entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
|
|
gst_props_add_entry (wavparse->metadata->properties, entry);
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (wavparse), "metadata");
|
|
}
|
|
|
|
/* Read 'fmt ' header */
|
|
static gboolean
|
|
gst_wavparse_fmt (GstWavParse * wav)
|
|
{
|
|
gst_riff_strf_auds *header = NULL;
|
|
GstCaps *caps;
|
|
|
|
if (!gst_riff_read_strf_auds (wav, &header)) {
|
|
g_warning ("Not fmt");
|
|
return FALSE;
|
|
}
|
|
|
|
wav->format = header->format;
|
|
wav->rate = header->rate;
|
|
wav->channels = header->channels;
|
|
if (wav->channels == 0) {
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
|
|
("Stream claims to contain zero channels - invalid data"));
|
|
g_free (header);
|
|
return FALSE;
|
|
}
|
|
wav->blockalign = header->blockalign;
|
|
wav->width = (header->blockalign * 8) / header->channels;
|
|
wav->depth = header->size;
|
|
wav->bps = header->av_bps;
|
|
if (wav->bps <= 0) {
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
|
|
("Stream claims to bitrate of <= zero - invalid data"));
|
|
g_free (header);
|
|
return FALSE;
|
|
}
|
|
|
|
/* Note: gst_riff_create_audio_caps might need to fix values in
|
|
* the header header depending on the format, so call it first */
|
|
caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
|
|
|
|
g_free (header);
|
|
|
|
if (caps) {
|
|
gst_wavparse_create_sourcepad (wav);
|
|
gst_pad_use_fixed_caps (wav->srcpad);
|
|
gst_pad_set_active (wav->srcpad, TRUE);
|
|
gst_pad_set_caps (wav->srcpad, caps);
|
|
gst_caps_free (caps);
|
|
gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
|
|
gst_element_no_more_pads (GST_ELEMENT (wav));
|
|
GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
|
|
} else {
|
|
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_other (GstWavParse * wav)
|
|
{
|
|
guint32 tag, length;
|
|
|
|
if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
|
|
GST_WARNING_OBJECT (wav, "could not peek head");
|
|
return FALSE;
|
|
}
|
|
GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
|
|
(gchar *) & tag, length);
|
|
|
|
switch (tag) {
|
|
case GST_RIFF_TAG_LIST:
|
|
if (!(tag = gst_riff_peek_list (wav))) {
|
|
GST_WARNING_OBJECT (wav, "could not peek list");
|
|
return FALSE;
|
|
}
|
|
|
|
switch (tag) {
|
|
case GST_RIFF_LIST_INFO:
|
|
if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
|
|
GST_WARNING_OBJECT (wav, "could not read list");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
case GST_RIFF_LIST_adtl:
|
|
if (!gst_riff_read_skip (wav)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
|
|
(gchar *) & tag);
|
|
if (!gst_riff_read_skip (wav)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
}
|
|
|
|
break;
|
|
|
|
case GST_RIFF_TAG_data:
|
|
if (!gst_bytestream_flush (wav->bs, 8)) {
|
|
GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (wav, "switching to data mode");
|
|
wav->state = GST_WAVPARSE_DATA;
|
|
wav->datastart = gst_bytestream_tell (wav->bs);
|
|
if (length == 0) {
|
|
guint64 file_length;
|
|
|
|
/* length is 0, data probably stretches to the end
|
|
* of file */
|
|
GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
|
|
/* get length of file */
|
|
file_length = gst_bytestream_length (wav->bs);
|
|
if (file_length == -1) {
|
|
GST_DEBUG_OBJECT (wav,
|
|
"could not get file length, assuming data to eof");
|
|
/* could not get length, assuming till eof */
|
|
length = G_MAXUINT32;
|
|
}
|
|
if (file_length > G_MAXUINT32) {
|
|
GST_DEBUG_OBJECT (wav, "file length %lld, clipping to 32 bits");
|
|
/* could not get length, assuming till eof */
|
|
length = G_MAXUINT32;
|
|
} else {
|
|
GST_DEBUG_OBJECT (wav, "file length %lld, datalength",
|
|
file_length, length);
|
|
/* substract offset of datastart from length */
|
|
length = file_length - wav->datastart;
|
|
GST_DEBUG_OBJECT (wav, "datalength %lld", length);
|
|
}
|
|
}
|
|
wav->datasize = (guint64) length;
|
|
GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
|
|
break;
|
|
|
|
case GST_RIFF_TAG_cue:
|
|
if (!gst_riff_read_skip (wav)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
|
|
if (!gst_riff_read_skip (wav))
|
|
return FALSE;
|
|
break;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
#endif
|
|
|
|
|
|
|
|
static gboolean
|
|
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
|
|
{
|
|
guint32 doctype;
|
|
|
|
if (!gst_riff_parse_file_header (element, buf, &doctype))
|
|
return FALSE;
|
|
|
|
if (doctype != GST_RIFF_RIFF_WAVE)
|
|
goto not_wav;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
not_wav:
|
|
{
|
|
GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
|
|
("File is not an WAVE file: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (doctype)));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_wavparse_stream_init (GstWavParse * wav)
|
|
{
|
|
GstFlowReturn res;
|
|
GstBuffer *buf = NULL;
|
|
|
|
if ((res = gst_pad_pull_range (wav->sinkpad,
|
|
wav->offset, 12, &buf)) != GST_FLOW_OK)
|
|
return res;
|
|
else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
|
|
return GST_FLOW_ERROR;
|
|
|
|
wav->offset += 12;
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/* This function is used to perform seeks on the element in
|
|
* pull mode.
|
|
*
|
|
* It also works when event is NULL, in which case it will just
|
|
* start from the last configured segment. This technique is
|
|
* used when activating the element and to perform the seek in
|
|
* READY.
|
|
*/
|
|
static gboolean
|
|
gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
|
|
{
|
|
gboolean res;
|
|
gdouble rate;
|
|
GstFormat format, bformat;
|
|
GstSeekFlags flags;
|
|
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
|
|
gint64 cur, stop, upstream_size;
|
|
gboolean flush;
|
|
gboolean update;
|
|
GstSegment seeksegment = { 0, };
|
|
|
|
if (event) {
|
|
GST_DEBUG_OBJECT (wav, "doing seek with event");
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags,
|
|
&cur_type, &cur, &stop_type, &stop);
|
|
|
|
/* we have to have a format as the segment format. Try to convert
|
|
* if not. */
|
|
if (format != GST_FORMAT_TIME) {
|
|
GstFormat fmt;
|
|
|
|
fmt = GST_FORMAT_TIME;
|
|
res = TRUE;
|
|
if (cur_type != GST_SEEK_TYPE_NONE)
|
|
res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur);
|
|
if (res && stop_type != GST_SEEK_TYPE_NONE)
|
|
res = gst_pad_query_convert (wav->srcpad, format, stop, &fmt, &stop);
|
|
if (!res)
|
|
goto no_format;
|
|
|
|
format = fmt;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (wav, "doing seek without event");
|
|
flags = 0;
|
|
cur_type = GST_SEEK_TYPE_SET;
|
|
stop_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
flush = flags & GST_SEEK_FLAG_FLUSH;
|
|
|
|
if (flush && wav->srcpad) {
|
|
GST_DEBUG_OBJECT (wav, "sending flush start");
|
|
gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
|
|
} else {
|
|
gst_pad_pause_task (wav->sinkpad);
|
|
}
|
|
|
|
GST_PAD_STREAM_LOCK (wav->sinkpad);
|
|
|
|
/* copy segment, we need this because we still need the old
|
|
* segment when we close the current segment. */
|
|
memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
|
|
|
|
if (event) {
|
|
GST_DEBUG_OBJECT (wav, "configuring seek");
|
|
gst_segment_set_seek (&seeksegment, rate, format, flags,
|
|
cur_type, cur, stop_type, stop, &update);
|
|
}
|
|
|
|
if ((stop = seeksegment.stop) == GST_CLOCK_TIME_NONE)
|
|
stop = seeksegment.duration;
|
|
|
|
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
|
|
if ((cur_type != GST_SEEK_TYPE_NONE)) {
|
|
wav->offset =
|
|
gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);
|
|
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
|
|
wav->offset -= (wav->offset % wav->bytes_per_sample);
|
|
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
|
|
wav->offset += wav->datastart;
|
|
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
|
|
} else {
|
|
GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
|
|
wav->offset);
|
|
}
|
|
|
|
if (stop_type != GST_SEEK_TYPE_NONE) {
|
|
wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
|
|
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
|
|
wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
|
|
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
|
|
wav->end_offset += wav->datastart;
|
|
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
|
|
} else {
|
|
GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
|
|
wav->end_offset);
|
|
}
|
|
|
|
/* make sure filesize is not exceeded due to rounding errors or so,
|
|
* same precaution as in _stream_headers */
|
|
bformat = GST_FORMAT_BYTES;
|
|
if (gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size))
|
|
wav->end_offset = MIN (wav->end_offset, upstream_size);
|
|
|
|
wav->offset = MIN (wav->offset, wav->end_offset);
|
|
wav->dataleft = wav->end_offset - wav->offset;
|
|
|
|
GST_DEBUG_OBJECT (wav,
|
|
"seek: offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT ", segment %"
|
|
GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, wav->offset, wav->end_offset,
|
|
GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
|
|
|
|
/* prepare for streaming again */
|
|
if (wav->srcpad) {
|
|
if (flush) {
|
|
GST_DEBUG_OBJECT (wav, "sending flush stop");
|
|
gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
|
|
} else if (wav->segment_running) {
|
|
/* we are running the current segment and doing a non-flushing seek,
|
|
* close the segment first based on the last_stop. */
|
|
GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, wav->segment.accum, wav->segment.last_stop);
|
|
|
|
gst_pad_push_event (wav->srcpad,
|
|
gst_event_new_new_segment (TRUE,
|
|
wav->segment.rate, wav->segment.format,
|
|
wav->segment.accum, wav->segment.last_stop, wav->segment.accum));
|
|
|
|
seeksegment.accum = wav->segment.last_stop;
|
|
}
|
|
}
|
|
|
|
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
|
|
|
|
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
gst_element_post_message (GST_ELEMENT (wav),
|
|
gst_message_new_segment_start (GST_OBJECT (wav),
|
|
wav->segment.format, wav->segment.last_stop));
|
|
}
|
|
|
|
/* now send the newsegment */
|
|
GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, wav->segment.last_stop, stop);
|
|
|
|
/* store the newsegment event so it can be sent from the streaming thread. */
|
|
if (wav->newsegment)
|
|
gst_event_unref (wav->newsegment);
|
|
wav->newsegment =
|
|
gst_event_new_new_segment (FALSE, wav->segment.rate,
|
|
wav->segment.format, wav->segment.last_stop, stop,
|
|
wav->segment.last_stop);
|
|
|
|
wav->segment_running = TRUE;
|
|
if (!wav->streaming) {
|
|
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
|
|
wav->sinkpad);
|
|
}
|
|
|
|
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_format:
|
|
{
|
|
GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* gst_wavparse_peek_chunk_info:
|
|
* @wav Wavparse object
|
|
* @tag holder for tag
|
|
* @size holder for tag size
|
|
*
|
|
* Peek next chunk info (tag and size)
|
|
*
|
|
* Returns: %TRUE when one chunk info has been got from the adapter
|
|
*/
|
|
static gboolean
|
|
gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
|
|
{
|
|
const guint8 *data = NULL;
|
|
|
|
if (gst_adapter_available (wav->adapter) < 8) {
|
|
return FALSE;
|
|
}
|
|
|
|
data = gst_adapter_peek (wav->adapter, 8);
|
|
*tag = GST_READ_UINT32_LE (data);
|
|
*size = GST_READ_UINT32_LE (data + 4);
|
|
|
|
GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
|
|
GST_FOURCC_ARGS (*tag));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/*
|
|
* gst_wavparse_peek_chunk:
|
|
* @wav Wavparse object
|
|
* @tag holder for tag
|
|
* @size holder for tag size
|
|
*
|
|
* Peek enough data for one full chunk
|
|
*
|
|
* Returns: %TRUE when one chunk has been got
|
|
*/
|
|
static gboolean
|
|
gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
|
|
{
|
|
guint32 peek_size = 0;
|
|
guint available;
|
|
|
|
if (!gst_wavparse_peek_chunk_info (wav, tag, size))
|
|
return FALSE;
|
|
|
|
GST_DEBUG ("Need to peek chunk of %d bytes", *size);
|
|
peek_size = (*size + 1) & ~1;
|
|
|
|
available = gst_adapter_available (wav->adapter);
|
|
if (available >= (8 + peek_size)) {
|
|
return TRUE;
|
|
} else {
|
|
GST_LOG ("but only %u bytes available now", available);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstFormat fmt = GST_FORMAT_BYTES;
|
|
GstPad *peer;
|
|
|
|
if ((peer = gst_pad_get_peer (wav->sinkpad))) {
|
|
res = gst_pad_query_duration (peer, &fmt, len);
|
|
gst_object_unref (peer);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_wavparse_stream_headers (GstWavParse * wav)
|
|
{
|
|
GstFlowReturn res;
|
|
GstBuffer *buf;
|
|
gst_riff_strf_auds *header = NULL;
|
|
guint32 tag, size;
|
|
gboolean gotdata = FALSE;
|
|
GstCaps *caps;
|
|
gint64 duration;
|
|
gchar *codec_name = NULL;
|
|
GstEvent **event_p;
|
|
|
|
while (!wav->got_fmt) {
|
|
GstBuffer *extra;
|
|
|
|
/* The header starts with a 'fmt ' tag */
|
|
if (wav->streaming) {
|
|
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
|
|
return GST_FLOW_OK;
|
|
|
|
buf = gst_buffer_new ();
|
|
gst_buffer_ref (buf);
|
|
gst_adapter_flush (wav->adapter, 8);
|
|
wav->offset += 8;
|
|
GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size);
|
|
GST_BUFFER_SIZE (buf) = size;
|
|
} else {
|
|
if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,
|
|
&wav->offset, &tag, &buf)) != GST_FLOW_OK)
|
|
return res;
|
|
}
|
|
|
|
if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_bext ||
|
|
tag == GST_RIFF_TAG_BEXT) {
|
|
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
|
|
GST_FOURCC_ARGS (tag));
|
|
if (wav->streaming) {
|
|
gst_adapter_flush (wav->adapter, size);
|
|
wav->offset += size;
|
|
}
|
|
gst_buffer_unref (buf);
|
|
buf = NULL;
|
|
continue;
|
|
}
|
|
|
|
if (tag != GST_RIFF_TAG_fmt)
|
|
goto invalid_wav;
|
|
|
|
if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))
|
|
goto parse_header_error;
|
|
|
|
if (wav->streaming) {
|
|
gst_adapter_flush (wav->adapter, size);
|
|
wav->offset += size;
|
|
GST_BUFFER_DATA (buf) = NULL;
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* Note: gst_riff_create_audio_caps might need to fix values in
|
|
* the header header depending on the format, so call it first */
|
|
caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
|
|
NULL, &codec_name);
|
|
|
|
if (extra)
|
|
gst_buffer_unref (extra);
|
|
|
|
wav->format = header->format;
|
|
wav->rate = header->rate;
|
|
wav->channels = header->channels;
|
|
wav->blockalign = header->blockalign;
|
|
wav->depth = header->size;
|
|
wav->bps = header->av_bps;
|
|
|
|
g_free (header);
|
|
|
|
if (wav->channels == 0)
|
|
goto no_channels;
|
|
|
|
if (wav->bps == 0 && (wav->format == GST_RIFF_WAVE_FORMAT_MPEGL12 ||
|
|
wav->format == GST_RIFF_WAVE_FORMAT_MPEGL3)) {
|
|
/* Note: ugly workaround for mp2/mp3 embedded in wav, that relies on the
|
|
* bitrate inside the mpeg stream */
|
|
/* wav->bps = 1; */
|
|
GST_INFO ("WAV file with bps==0 and format=mp2/3");
|
|
}
|
|
|
|
wav->width = (wav->blockalign * 8) / wav->channels;
|
|
wav->bytes_per_sample = wav->channels * wav->width / 8;
|
|
|
|
if (wav->bytes_per_sample <= 0)
|
|
goto no_bytes_per_sample;
|
|
|
|
if (!caps)
|
|
goto unknown_format;
|
|
|
|
GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
|
|
GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
|
|
GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
|
|
GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
|
|
GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate);
|
|
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
|
|
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
|
|
|
|
GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
|
|
|
|
/* create pad later so we can sniff the first few bytes
|
|
* of the real data and correct our caps if necessary */
|
|
gst_caps_replace (&wav->caps, caps);
|
|
gst_caps_replace (&caps, NULL);
|
|
|
|
wav->got_fmt = TRUE;
|
|
|
|
if (codec_name) {
|
|
wav->tags = gst_tag_list_new ();
|
|
|
|
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_AUDIO_CODEC, codec_name, NULL);
|
|
|
|
g_free (codec_name);
|
|
codec_name = NULL;
|
|
}
|
|
|
|
}
|
|
|
|
/* loop headers until we get data */
|
|
while (!gotdata) {
|
|
gint64 upstream_size = 0;
|
|
|
|
if (wav->streaming) {
|
|
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
|
|
return GST_FLOW_OK;
|
|
} else {
|
|
if ((res =
|
|
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
|
|
&buf)) != GST_FLOW_OK)
|
|
goto header_read_error;
|
|
tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
|
|
size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
|
|
}
|
|
|
|
gst_wavparse_get_upstream_size (wav, &upstream_size);
|
|
|
|
/*
|
|
wav is a st00pid format, we don't know for sure where data starts.
|
|
So we have to go bit by bit until we find the 'data' header
|
|
*/
|
|
switch (tag) {
|
|
/* TODO : Implement the various cases */
|
|
case GST_RIFF_TAG_data:{
|
|
GstFormat fmt;
|
|
|
|
GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
|
|
if (wav->streaming) {
|
|
gst_adapter_flush (wav->adapter, 8);
|
|
gotdata = TRUE;
|
|
} else {
|
|
gst_buffer_unref (buf);
|
|
}
|
|
wav->offset += 8;
|
|
wav->datastart = wav->offset;
|
|
/* file might be truncated */
|
|
fmt = GST_FORMAT_BYTES;
|
|
if (upstream_size) {
|
|
size = MIN (size, (upstream_size - wav->datastart));
|
|
}
|
|
wav->datasize = (guint64) size;
|
|
wav->dataleft = (guint64) size;
|
|
wav->end_offset = size + wav->datastart;
|
|
if (!wav->streaming) {
|
|
/* We will continue parsing tags 'till end */
|
|
wav->offset += size;
|
|
}
|
|
GST_DEBUG_OBJECT (wav, "datasize = %d", size);
|
|
break;
|
|
}
|
|
case GST_RIFF_TAG_fact:{
|
|
/* number of samples (for compressed formats) */
|
|
if (wav->streaming) {
|
|
const guint8 *data = NULL;
|
|
|
|
if (gst_adapter_available (wav->adapter) < 8 + 4) {
|
|
return GST_FLOW_OK;
|
|
}
|
|
gst_adapter_flush (wav->adapter, 8);
|
|
data = gst_adapter_peek (wav->adapter, 4);
|
|
wav->fact = GST_READ_UINT32_LE (data);
|
|
gst_adapter_flush (wav->adapter, 4);
|
|
} else {
|
|
gst_buffer_unref (buf);
|
|
if ((res =
|
|
gst_pad_pull_range (wav->sinkpad, wav->offset + 8, 4,
|
|
&buf)) != GST_FLOW_OK)
|
|
goto header_read_error;
|
|
wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
|
|
gst_buffer_unref (buf);
|
|
}
|
|
wav->offset += 8 + 4;
|
|
break;
|
|
}
|
|
default:
|
|
if (wav->streaming) {
|
|
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
|
|
return GST_FLOW_OK;
|
|
}
|
|
GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (tag));
|
|
wav->offset += 8 + ((size + 1) & ~1);
|
|
if (wav->streaming) {
|
|
gst_adapter_flush (wav->adapter, 8 + ((size + 1) & ~1));
|
|
} else {
|
|
gst_buffer_unref (buf);
|
|
}
|
|
}
|
|
|
|
if (upstream_size && (wav->offset >= upstream_size)) {
|
|
/* Now we are gone through the whole file */
|
|
gotdata = TRUE;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (wav, "Finished parsing headers");
|
|
|
|
if (wav->bps <= 0 && wav->fact) {
|
|
wav->bps =
|
|
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
|
|
(guint64) wav->fact);
|
|
GST_DEBUG_OBJECT (wav, "calculated bps : %d", wav->bps);
|
|
}
|
|
if (wav->bps <= 0)
|
|
goto no_bitrate;
|
|
|
|
duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
|
|
GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (duration));
|
|
gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration);
|
|
|
|
/* now we have all the info to perform a pending seek if any, if no
|
|
* event, this will still do the right thing and it will also send
|
|
* the right newsegment event downstream. */
|
|
gst_wavparse_perform_seek (wav, wav->seek_event);
|
|
/* remove pending event */
|
|
event_p = &wav->seek_event;
|
|
gst_event_replace (event_p, NULL);
|
|
|
|
wav->state = GST_WAVPARSE_DATA;
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERROR */
|
|
invalid_wav:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
|
|
("Invalid WAV header (no fmt at start): %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
parse_header_error:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
|
|
("Couldn't parse audio header"));
|
|
gst_buffer_unref (buf);
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
no_channels:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
|
|
("Stream claims to contain no channels - invalid data"));
|
|
g_free (header);
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
no_bitrate:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
|
|
("Stream claims to have a bitrate of <= zero - invalid data"));
|
|
g_free (header);
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
no_bytes_per_sample:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
|
|
("could not caluclate bytes per sample - invalid data"));
|
|
g_free (header);
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
unknown_format:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
|
|
("No caps found for format 0x%x, %d channels, %d Hz",
|
|
wav->format, wav->channels, wav->rate));
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
header_read_error:
|
|
{
|
|
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't read in header"));
|
|
g_free (codec_name);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
|
|
/*
|
|
* Read WAV file tag when streaming
|
|
*/
|
|
static GstFlowReturn
|
|
gst_wavparse_parse_stream_init (GstWavParse * wav)
|
|
{
|
|
if (gst_adapter_available (wav->adapter) >= 12) {
|
|
GstBuffer *tmp = gst_buffer_new ();
|
|
|
|
/* _take flushes the data */
|
|
GST_BUFFER_DATA (tmp) = gst_adapter_take (wav->adapter, 12);
|
|
GST_BUFFER_SIZE (tmp) = 12;
|
|
|
|
GST_DEBUG ("Parsing wav header");
|
|
if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) {
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
wav->offset += 12;
|
|
/* Go to next state */
|
|
wav->state = GST_WAVPARSE_HEADER;
|
|
}
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/* handle an event sent directly to the element.
|
|
*
|
|
* This event can be sent either in the READY state or the
|
|
* >READY state. The only event of interest really is the seek
|
|
* event.
|
|
*
|
|
* In the READY state we can only store the event and try to
|
|
* respect it when going to PAUSED. We assume we are in the
|
|
* READY state when our parsing state != GST_WAVPARSE_DATA.
|
|
*
|
|
* When we are steaming, we can simply perform the seek right
|
|
* away.
|
|
*/
|
|
static gboolean
|
|
gst_wavparse_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstWavParse *wav = GST_WAVPARSE (element);
|
|
gboolean res = FALSE;
|
|
GstEvent **event_p;
|
|
|
|
GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
if (wav->state == GST_WAVPARSE_DATA) {
|
|
/* we can handle the seek directly when streaming data */
|
|
res = gst_wavparse_perform_seek (wav, event);
|
|
} else {
|
|
GST_DEBUG_OBJECT (wav, "queuing seek for later");
|
|
|
|
event_p = &wav->seek_event;
|
|
gst_event_replace (event_p, event);
|
|
|
|
/* we always return true */
|
|
res = TRUE;
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
gst_event_unref (event);
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
|
|
{
|
|
GstStructure *s;
|
|
const guint8 dts_marker[] = { 0xFF, 0x1F, 0x00, 0xE8, 0xF1, 0x07 };
|
|
|
|
s = gst_caps_get_structure (wav->caps, 0);
|
|
if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf &&
|
|
GST_BUFFER_SIZE (buf) > 6 &&
|
|
memcmp (GST_BUFFER_DATA (buf), dts_marker, 6) == 0) {
|
|
|
|
GST_WARNING_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
|
|
gst_caps_unref (wav->caps);
|
|
wav->caps = gst_caps_from_string ("audio/x-dts");
|
|
|
|
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_AUDIO_CODEC, "dts", NULL);
|
|
}
|
|
|
|
gst_wavparse_create_sourcepad (wav);
|
|
gst_pad_set_active (wav->srcpad, TRUE);
|
|
gst_pad_set_caps (wav->srcpad, wav->caps);
|
|
gst_caps_replace (&wav->caps, NULL);
|
|
|
|
gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
|
|
gst_element_no_more_pads (GST_ELEMENT (wav));
|
|
|
|
GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad");
|
|
gst_pad_push_event (wav->srcpad, wav->newsegment);
|
|
wav->newsegment = NULL;
|
|
|
|
if (wav->tags) {
|
|
gst_element_found_tags_for_pad (GST_ELEMENT (wav), wav->srcpad, wav->tags);
|
|
wav->tags = NULL;
|
|
}
|
|
}
|
|
|
|
#define MAX_BUFFER_SIZE 4096
|
|
|
|
static GstFlowReturn
|
|
gst_wavparse_stream_data (GstWavParse * wav)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn res = GST_FLOW_OK;
|
|
guint64 desired, obtained;
|
|
GstClockTime timestamp, next_timestamp;
|
|
guint64 pos, nextpos;
|
|
|
|
iterate_adapter:
|
|
GST_LOG_OBJECT (wav,
|
|
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
|
|
G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
|
|
|
|
/* Get the next n bytes and output them */
|
|
if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
|
|
goto found_eos;
|
|
|
|
/* scale the amount of data by the segment rate so we get equal
|
|
* amounts of data regardless of the playback rate */
|
|
desired =
|
|
MIN (gst_guint64_to_gdouble (wav->dataleft),
|
|
MAX_BUFFER_SIZE * wav->segment.abs_rate);
|
|
|
|
if (desired >= wav->blockalign && wav->blockalign > 0)
|
|
desired -= (desired % wav->blockalign);
|
|
|
|
GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
|
|
"from the sinkpad", desired);
|
|
|
|
if (wav->streaming) {
|
|
guint avail = gst_adapter_available (wav->adapter);
|
|
|
|
if (avail < desired) {
|
|
GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired);
|
|
GST_BUFFER_SIZE (buf) = desired;
|
|
} else {
|
|
if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
|
|
desired, &buf)) != GST_FLOW_OK)
|
|
goto pull_error;
|
|
}
|
|
|
|
/* first chunk of data? create the source pad. We do this only here so
|
|
* we can detect broken .wav files with dts disguised as raw PCM (sigh) */
|
|
if (G_UNLIKELY (wav->first)) {
|
|
wav->first = FALSE;
|
|
gst_wavparse_add_src_pad (wav, buf);
|
|
}
|
|
|
|
/* If we have a pending newsegment send it now. */
|
|
if (G_UNLIKELY (wav->newsegment != NULL)) {
|
|
gst_pad_push_event (wav->srcpad, wav->newsegment);
|
|
wav->newsegment = NULL;
|
|
}
|
|
|
|
obtained = GST_BUFFER_SIZE (buf);
|
|
|
|
/* our positions */
|
|
pos = wav->offset - wav->datastart;
|
|
nextpos = pos + obtained;
|
|
|
|
/* update offsets, does not overflow. */
|
|
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
|
|
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
|
|
|
|
/* and timestamps, be carefull for overflows */
|
|
timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps);
|
|
next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps);
|
|
|
|
GST_BUFFER_TIMESTAMP (buf) = timestamp;
|
|
GST_BUFFER_DURATION (buf) = next_timestamp - timestamp;
|
|
|
|
/* update current running segment position */
|
|
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
|
|
|
|
/* don't forget to set the caps on the buffer */
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
|
|
|
|
GST_LOG_OBJECT (wav,
|
|
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
|
|
", size:%u", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf));
|
|
|
|
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
|
|
goto push_error;
|
|
|
|
if (obtained < wav->dataleft) {
|
|
wav->dataleft -= obtained;
|
|
} else {
|
|
wav->dataleft = 0;
|
|
}
|
|
wav->offset += obtained;
|
|
/* Iterate until need more data, so adapter size won't grow */
|
|
if (wav->streaming) {
|
|
GST_LOG_OBJECT (wav,
|
|
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
|
|
wav->end_offset);
|
|
goto iterate_adapter;
|
|
}
|
|
|
|
return res;
|
|
|
|
/* ERROR */
|
|
found_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (wav, "found EOS");
|
|
/* we completed the segment */
|
|
wav->segment_running = FALSE;
|
|
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
GstClockTime stop;
|
|
|
|
if ((stop = wav->segment.stop) == -1)
|
|
stop = wav->segment.duration;
|
|
|
|
gst_element_post_message (GST_ELEMENT (wav),
|
|
gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
|
|
stop));
|
|
} else {
|
|
if (G_UNLIKELY (wav->first)) {
|
|
wav->first = FALSE;
|
|
gst_wavparse_add_src_pad (wav, NULL);
|
|
}
|
|
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
|
|
}
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
pull_error:
|
|
{
|
|
/* check if we got EOS */
|
|
if (res == GST_FLOW_UNEXPECTED)
|
|
goto found_eos;
|
|
|
|
GST_WARNING_OBJECT (wav,
|
|
"Error getting %" G_GINT64_FORMAT " bytes from the "
|
|
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
|
|
return res;
|
|
}
|
|
push_error:
|
|
{
|
|
GST_WARNING_OBJECT (wav, "Error pushing on srcpad %p, is linked? = %d",
|
|
wav->srcpad, gst_pad_is_linked (wav->srcpad));
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_wavparse_loop (GstPad * pad)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
|
|
GST_LOG_OBJECT (wav, "process data");
|
|
|
|
switch (wav->state) {
|
|
case GST_WAVPARSE_START:
|
|
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
|
|
if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
wav->state = GST_WAVPARSE_HEADER;
|
|
/* fall-through */
|
|
|
|
case GST_WAVPARSE_HEADER:
|
|
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
|
|
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
wav->state = GST_WAVPARSE_DATA;
|
|
/* fall-through */
|
|
|
|
case GST_WAVPARSE_DATA:
|
|
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
pause:
|
|
GST_LOG_OBJECT (wav, "pausing task %d", ret);
|
|
gst_pad_pause_task (wav->sinkpad);
|
|
if (GST_FLOW_IS_FATAL (ret)) {
|
|
/* for fatal errors we post an error message */
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
|
|
(_("Internal data stream error.")),
|
|
("streaming stopped, reason %s", gst_flow_get_name (ret)));
|
|
if (wav->srcpad != NULL)
|
|
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
|
|
GST_LOG_OBJECT (wav, "adapter_push %u bytes", GST_BUFFER_SIZE (buf));
|
|
|
|
gst_adapter_push (wav->adapter, buf);
|
|
|
|
switch (wav->state) {
|
|
case GST_WAVPARSE_START:
|
|
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
|
|
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
if (wav->state != GST_WAVPARSE_HEADER)
|
|
break;
|
|
|
|
/* otherwise fall-through */
|
|
|
|
case GST_WAVPARSE_HEADER:
|
|
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
|
|
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
|
|
if (!wav->got_fmt || wav->datastart == 0)
|
|
break;
|
|
|
|
wav->state = GST_WAVPARSE_DATA;
|
|
/* fall-through */
|
|
|
|
case GST_WAVPARSE_DATA:
|
|
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
|
|
goto pause;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
return ret;
|
|
|
|
pause:
|
|
GST_LOG_OBJECT (wav, "pausing task %d", ret);
|
|
gst_pad_pause_task (wav->sinkpad);
|
|
if (GST_FLOW_IS_FATAL (ret)) {
|
|
/* for fatal errors we post an error message */
|
|
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
|
|
(_("Internal data stream error.")),
|
|
("streaming stopped, reason %s", gst_flow_get_name (ret)));
|
|
if (wav->srcpad != NULL)
|
|
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
#if 0
|
|
/* convert and query stuff */
|
|
static const GstFormat *
|
|
gst_wavparse_get_formats (GstPad * pad)
|
|
{
|
|
static GstFormat formats[] = {
|
|
GST_FORMAT_TIME,
|
|
GST_FORMAT_BYTES,
|
|
GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
|
|
0
|
|
};
|
|
|
|
return formats;
|
|
}
|
|
#endif
|
|
|
|
static gboolean
|
|
gst_wavparse_pad_convert (GstPad * pad,
|
|
GstFormat src_format, gint64 src_value,
|
|
GstFormat * dest_format, gint64 * dest_value)
|
|
{
|
|
GstWavParse *wavparse;
|
|
gboolean res = TRUE;
|
|
|
|
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
|
|
|
|
if (wavparse->bytes_per_sample == 0)
|
|
goto no_bytes_per_sample;
|
|
|
|
if (wavparse->bps == 0)
|
|
goto no_bps;
|
|
|
|
switch (src_format) {
|
|
case GST_FORMAT_BYTES:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value = src_value / wavparse->bytes_per_sample;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value =
|
|
gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->bps);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
*dest_value -= *dest_value % wavparse->bytes_per_sample;
|
|
break;
|
|
|
|
case GST_FORMAT_DEFAULT:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
*dest_value = src_value * wavparse->bytes_per_sample;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value =
|
|
gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->rate);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
break;
|
|
|
|
case GST_FORMAT_TIME:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
/* make sure we end up on a sample boundary */
|
|
*dest_value =
|
|
gst_util_uint64_scale_int (src_value, wavparse->bps, GST_SECOND);
|
|
*dest_value -= *dest_value % wavparse->blockalign;
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value =
|
|
gst_util_uint64_scale_int (src_value, wavparse->rate, GST_SECOND);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
done:
|
|
gst_object_unref (wavparse);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_bytes_per_sample:
|
|
{
|
|
GST_DEBUG_OBJECT (wavparse,
|
|
"bytes_per_sample 0, probably an mp3 - channels %d, width %d",
|
|
wavparse->channels, wavparse->width);
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
no_bps:
|
|
{
|
|
GST_DEBUG_OBJECT (wavparse, "bps 0, cannot convert");
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_wavparse_get_query_types (GstPad * pad)
|
|
{
|
|
static const GstQueryType types[] = {
|
|
GST_QUERY_POSITION,
|
|
GST_QUERY_DURATION,
|
|
GST_QUERY_CONVERT,
|
|
0
|
|
};
|
|
|
|
return types;
|
|
}
|
|
|
|
/* handle queries for location and length in requested format */
|
|
static gboolean
|
|
gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
|
|
/* only if we know */
|
|
if (wav->state != GST_WAVPARSE_DATA)
|
|
return FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
gint64 curb;
|
|
gint64 cur;
|
|
GstFormat format;
|
|
gboolean res = TRUE;
|
|
|
|
curb = wav->offset - wav->datastart;
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_TIME:
|
|
res &=
|
|
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
|
|
&format, &cur);
|
|
break;
|
|
default:
|
|
format = GST_FORMAT_BYTES;
|
|
cur = curb;
|
|
break;
|
|
}
|
|
if (res)
|
|
gst_query_set_position (query, format, cur);
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
gint64 endb;
|
|
gint64 end;
|
|
GstFormat format;
|
|
gboolean res = TRUE;
|
|
|
|
endb = wav->datasize;
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_TIME:{
|
|
if (wav->fact) {
|
|
end = GST_SECOND * wav->fact / wav->rate;
|
|
} else {
|
|
res &=
|
|
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
|
|
&format, &end);
|
|
}
|
|
}
|
|
break;
|
|
default:
|
|
format = GST_FORMAT_BYTES;
|
|
end = endb;
|
|
break;
|
|
}
|
|
if (res)
|
|
gst_query_set_duration (query, format, end);
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
gint64 srcvalue, dstvalue;
|
|
GstFormat srcformat, dstformat;
|
|
|
|
gst_query_parse_convert (query, &srcformat, &srcvalue,
|
|
&dstformat, &dstvalue);
|
|
res &=
|
|
gst_wavparse_pad_convert (pad, srcformat, srcvalue,
|
|
&dstformat, &dstvalue);
|
|
if (res)
|
|
gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
gboolean res = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (wavparse, "event %d, %s", GST_EVENT_TYPE (event),
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
/* can only handle events when we are in the data state */
|
|
if (wavparse->state != GST_WAVPARSE_DATA)
|
|
return FALSE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
res = gst_wavparse_perform_seek (wavparse, event);
|
|
break;
|
|
}
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_sink_activate (GstPad * sinkpad)
|
|
{
|
|
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
|
|
gboolean res;
|
|
|
|
if (gst_pad_check_pull_range (sinkpad)) {
|
|
GST_DEBUG ("going to pull mode");
|
|
wav->streaming = FALSE;
|
|
wav->adapter = NULL;
|
|
res = gst_pad_activate_pull (sinkpad, TRUE);
|
|
} else {
|
|
GST_DEBUG ("going to push (streaming) mode");
|
|
wav->streaming = TRUE;
|
|
wav->adapter = gst_adapter_new ();
|
|
res = gst_pad_activate_push (sinkpad, TRUE);
|
|
}
|
|
gst_object_unref (wav);
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
|
|
{
|
|
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
|
|
|
|
GST_DEBUG_OBJECT (wav, "activating pull");
|
|
|
|
if (active) {
|
|
/* if we have a scheduler we can start the task */
|
|
wav->segment_running = TRUE;
|
|
gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad);
|
|
} else {
|
|
gst_pad_stop_task (sinkpad);
|
|
}
|
|
gst_object_unref (wav);
|
|
|
|
return TRUE;
|
|
};
|
|
|
|
static GstStateChangeReturn
|
|
gst_wavparse_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstWavParse *wav = GST_WAVPARSE (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_wavparse_reset (wav);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:{
|
|
GstEvent **event_p = &wav->seek_event;
|
|
|
|
gst_wavparse_destroy_sourcepad (wav);
|
|
gst_event_replace (event_p, NULL);
|
|
gst_wavparse_reset (wav);
|
|
if (wav->adapter) {
|
|
gst_adapter_clear (wav->adapter);
|
|
}
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
gst_riff_init ();
|
|
|
|
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
|
|
|
|
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
|
|
GST_TYPE_WAVPARSE);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"wavparse",
|
|
"Parse a .wav file into raw audio",
|
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|