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gst/wavparse/gstwavparse.*: correct partial implementation of push mode (from my last commit)
Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (plugin_init): * gst/wavparse/gstwavparse.h: correct partial implementation of push mode (from my last commit)
This commit is contained in:
Stefan Kost
parent
da6b539bbe
commit
562de2f6e0
3 changed files with 143 additions and 94 deletions
14
ChangeLog
14
ChangeLog
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@ -1,3 +1,17 @@
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2006-04-26 Stefan Kost <ensonic@users.sf.net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
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(gst_wavparse_class_init), (gst_wavparse_reset),
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(gst_wavparse_init), (gst_wavparse_create_sourcepad),
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(gst_wavparse_parse_adtl), (gst_wavparse_parse_cues),
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(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
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(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
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(gst_wavparse_stream_data), (gst_wavparse_loop),
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(gst_wavparse_chain), (plugin_init):
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* gst/wavparse/gstwavparse.h:
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correct partial implementation of push mode
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(from my last commit)
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2006-04-26 Wim Taymans <wim@fluendo.com>
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* ext/esd/esdsink.c:
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@ -1,7 +1,7 @@
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/* -*- 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.
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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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@ -32,11 +32,18 @@
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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 sine.wav ! wavparse ! audioconvert ! alsasink
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* gst-launch filesrc location=sine.wav ! queue ! 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 ! wavparse ! audioconvert ! alsasink
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* </programlisting>
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* Stream data from
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* </para>
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*
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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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@ -84,6 +91,12 @@ 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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@ -162,22 +175,13 @@ 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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GstPadTemplate *templ;
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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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gst_element_class_set_details (element_class, &gst_wavparse_details);
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/* register src pads */
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templ = gst_static_pad_template_get (&sink_template_factory);
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gst_element_class_add_pad_template (element_class, templ);
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gst_object_unref (templ);
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templ = gst_static_pad_template_get (&src_template_factory);
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gst_element_class_add_pad_template (element_class, templ);
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gst_object_unref (templ);
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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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@ -196,8 +200,6 @@ gst_wavparse_class_init (GstWavParseClass * klass)
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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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GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
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}
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@ -234,6 +236,7 @@ gst_wavparse_reset (GstWavParse * wavparse)
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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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@ -258,6 +261,9 @@ gst_wavparse_init (GstWavParse * wavparse)
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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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@ -285,6 +291,8 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse)
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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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@ -301,6 +309,8 @@ gst_wavparse_get_property (GObject * object,
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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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@ -484,9 +494,7 @@ gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
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g_object_notify (G_OBJECT (wavparse), "metadata");
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}
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#endif
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#if 0
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static void
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gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
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{
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@ -549,49 +557,7 @@ gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
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g_object_notify (G_OBJECT (wavparse), "metadata");
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}
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#endif
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static gboolean
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gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
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{
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guint32 doctype;
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if (!gst_riff_parse_file_header (element, buf, &doctype))
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return FALSE;
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if (doctype != GST_RIFF_RIFF_WAVE)
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goto not_wav;
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return TRUE;
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/* ERRORS */
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not_wav:
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{
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GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
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("File is not an WAVE file: %" GST_FOURCC_FORMAT,
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GST_FOURCC_ARGS (doctype)));
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_wavparse_stream_init (GstWavParse * wav)
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{
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GstFlowReturn res;
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GstBuffer *buf = NULL;
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if ((res = gst_pad_pull_range (wav->sinkpad,
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wav->offset, 12, &buf)) != GST_FLOW_OK)
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return res;
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else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
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return GST_FLOW_ERROR;
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wav->offset += 12;
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return GST_FLOW_OK;
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}
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#if 0
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/* Read 'fmt ' header */
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static gboolean
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gst_wavparse_fmt (GstWavParse * wav)
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@ -749,6 +715,48 @@ gst_wavparse_other (GstWavParse * wav)
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}
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#endif
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static gboolean
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gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
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{
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guint32 doctype;
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if (!gst_riff_parse_file_header (element, buf, &doctype))
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return FALSE;
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if (doctype != GST_RIFF_RIFF_WAVE)
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goto not_wav;
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return TRUE;
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/* ERRORS */
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not_wav:
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{
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GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
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("File is not an WAVE file: %" GST_FOURCC_FORMAT,
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GST_FOURCC_ARGS (doctype)));
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_wavparse_stream_init (GstWavParse * wav)
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{
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GstFlowReturn res;
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GstBuffer *buf = NULL;
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if ((res = gst_pad_pull_range (wav->sinkpad,
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wav->offset, 12, &buf)) != GST_FLOW_OK)
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return res;
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else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
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return GST_FLOW_ERROR;
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wav->offset += 12;
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return GST_FLOW_OK;
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}
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/* This function is used to perform seeks on the element in
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* pull mode.
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*
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@ -800,7 +808,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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flush = flags & GST_SEEK_FLAG_FLUSH;
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if (flush) {
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if (flush && wav->srcpad) {
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GST_DEBUG_OBJECT (wav, "sending flush start");
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gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
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} else {
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@ -846,6 +854,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
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/* prepare for streaming again */
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if (wav->srcpad) {
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if (flush) {
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GST_DEBUG_OBJECT (wav, "sending flush stop");
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gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
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@ -860,6 +869,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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wav->segment.rate, wav->segment.format,
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wav->segment.start, wav->segment.last_stop, wav->segment.time));
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}
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}
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memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
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@ -1080,7 +1090,6 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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/* loop headers until we get data */
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while (!gotdata) {
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if (wav->streaming) {
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if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
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return GST_FLOW_OK;
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@ -1322,7 +1331,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
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#define MAX_BUFFER_SIZE 4096
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static GstFlowReturn
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gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
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gst_wavparse_stream_data (GstWavParse * wav)
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{
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GstBuffer *buf = NULL;
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GstFlowReturn res = GST_FLOW_OK;
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@ -1331,8 +1340,9 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
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guint64 pos, nextpos;
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iterate_adapter:
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GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT,
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wav->offset, wav->end_offset);
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GST_LOG_OBJECT (wav,
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"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
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G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
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/* Get the next n bytes and output them */
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if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
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@ -1350,8 +1360,12 @@ iterate_adapter:
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"from the sinkpad", desired);
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if (wav->streaming) {
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if (gst_adapter_available (wav->adapter) < desired)
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guint avail = gst_adapter_available (wav->adapter);
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if (avail < desired) {
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GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
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return GST_FLOW_OK;
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}
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buf = gst_buffer_new ();
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GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired);
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@ -1362,14 +1376,15 @@ iterate_adapter:
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goto pull_error;
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}
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obtained = GST_BUFFER_SIZE (buf);
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/* first chunk of data? create the source pad. We do this only here so
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* we can detect broken .wav files with dts disguised as raw PCM (sigh) */
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if (first) {
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if (G_UNLIKELY (wav->first)) {
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wav->first = FALSE;
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gst_wavparse_add_src_pad (wav, buf);
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}
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obtained = GST_BUFFER_SIZE (buf);
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/* our positions */
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pos = wav->offset - wav->datastart;
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nextpos = pos + obtained;
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@ -1400,16 +1415,17 @@ iterate_adapter:
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if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
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goto push_error;
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} else {
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GST_DEBUG ("Srcpad not linked!");
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gst_buffer_unref (buf);
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goto not_linked;
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}
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if (obtained < wav->dataleft) {
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wav->dataleft -= obtained;
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wav->offset += obtained;
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//wav->offset += obtained;
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} else {
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wav->dataleft = 0;
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}
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wav->offset += obtained;
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/* Iterate until need more data, so adapter size won't grow */
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if (wav->streaming) {
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GST_LOG_OBJECT (wav,
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@ -1451,6 +1467,11 @@ push_error:
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GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
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return res;
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}
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not_linked:
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{
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GST_DEBUG_OBJECT (wav, "Srcpad not linked!");
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return GST_FLOW_ERROR;
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}
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}
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static void
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@ -1459,8 +1480,11 @@ gst_wavparse_loop (GstPad * pad)
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GstFlowReturn ret;
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GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
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GST_LOG_OBJECT (wav, "process data");
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switch (wav->state) {
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case GST_WAVPARSE_START:
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GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
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if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
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goto pause;
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|
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|
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@ -1468,15 +1492,15 @@ gst_wavparse_loop (GstPad * pad)
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/* fall-through */
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case GST_WAVPARSE_HEADER:
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GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
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if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
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goto pause;
|
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|
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wav->state = GST_WAVPARSE_DATA;
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if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
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goto pause;
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break;
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/* fall-through */
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case GST_WAVPARSE_DATA:
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if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
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if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
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goto pause;
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break;
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default:
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@ -1505,24 +1529,30 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
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GstFlowReturn ret;
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GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
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GST_LOG_OBJECT (wav, "adapter_push %" G_GINT64_FORMAT " bytes",
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GST_BUFFER_SIZE (buf));
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gst_adapter_push (wav->adapter, buf);
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|
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switch (wav->state) {
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case GST_WAVPARSE_START:
|
||||
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
|
||||
if ((ret = gst_wavparse_parse_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;
|
||||
if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK)
|
||||
goto pause;
|
||||
break;
|
||||
/* fall-through */
|
||||
|
||||
case GST_WAVPARSE_DATA:
|
||||
if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
|
||||
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
|
||||
goto pause;
|
||||
break;
|
||||
default:
|
||||
|
|
@ -1868,6 +1898,8 @@ 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);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
||||
* Copyright (C) <2006> Nokia Corporation.
|
||||
* Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
|
|
@ -103,6 +103,9 @@ struct _GstWavParse {
|
|||
/* configured segment, start/stop expressed in time */
|
||||
GstSegment segment;
|
||||
gboolean segment_running;
|
||||
|
||||
/* for late pad configuration */
|
||||
gboolean first;
|
||||
};
|
||||
|
||||
struct _GstWavParseClass {
|
||||
|
|
|
|||
Loading…
Reference in a new issue