Commit 562de2f6 authored by Stefan Kost's avatar Stefan Kost
Browse files

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)
parent da6b539b
2006-04-26 Stefan Kost <ensonic@users.sf.net>
* 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)
2006-04-26 Wim Taymans <wim@fluendo.com>
* ext/esd/esdsink.c:
......
/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
/* 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
......@@ -32,11 +32,18 @@
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink
* gst-launch filesrc location=sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Read a wav file and output to the soundcard using the ALSA element. The
* wav file is assumed to contain raw uncompressed samples.
* </para>
* <para>
* <programlisting>
* gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Stream data from
* </para>
*
* </refsect2>
*
* Last reviewed on 2006-03-03 (0.10.3)
......@@ -84,6 +91,12 @@ static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
static void gst_wavparse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static const GstElementDetails gst_wavparse_details =
GST_ELEMENT_DETAILS ("WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
GST_PAD_SINK,
......@@ -162,22 +175,13 @@ static void
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *templ;
static const GstElementDetails gst_wavparse_details =
GST_ELEMENT_DETAILS ("WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
gst_element_class_set_details (element_class, &gst_wavparse_details);
/* register src pads */
templ = gst_static_pad_template_get (&sink_template_factory);
gst_element_class_add_pad_template (element_class, templ);
gst_object_unref (templ);
templ = gst_static_pad_template_get (&src_template_factory);
gst_element_class_add_pad_template (element_class, templ);
gst_object_unref (templ);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template_factory));
gst_element_class_set_details (element_class, &gst_wavparse_details);
}
static void
......@@ -196,8 +200,6 @@ gst_wavparse_class_init (GstWavParseClass * klass)
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
}
......@@ -234,6 +236,7 @@ gst_wavparse_reset (GstWavParse * wavparse)
wavparse->datasize = 0;
wavparse->datastart = 0;
wavparse->got_fmt = FALSE;
wavparse->first = TRUE;
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
......@@ -258,6 +261,9 @@ gst_wavparse_init (GstWavParse * wavparse)
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
/* src, will be created later */
wavparse->srcpad = NULL;
}
static void
......@@ -285,6 +291,8 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse)
GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
gst_pad_set_event_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
GST_DEBUG_OBJECT (wavparse, "srcpad created");
}
static void
......@@ -301,6 +309,8 @@ gst_wavparse_get_property (GObject * object,
}
}
#if 0
static void
gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
......@@ -484,9 +494,7 @@ gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
g_object_notify (G_OBJECT (wavparse), "metadata");
}
#endif
#if 0
static void
gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
{
......@@ -549,49 +557,7 @@ gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
g_object_notify (G_OBJECT (wavparse), "metadata");
}
#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;
}
#if 0
/* Read 'fmt ' header */
static gboolean
gst_wavparse_fmt (GstWavParse * wav)
......@@ -749,6 +715,48 @@ gst_wavparse_other (GstWavParse * wav)
}
#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.
*
......@@ -800,7 +808,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
flush = flags & GST_SEEK_FLAG_FLUSH;
if (flush) {
if (flush && wav->srcpad) {
GST_DEBUG_OBJECT (wav, "sending flush start");
gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
} else {
......@@ -846,19 +854,21 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
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.start, wav->segment.last_stop);
gst_pad_push_event (wav->srcpad,
gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.start, wav->segment.last_stop, wav->segment.time));
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.start, wav->segment.last_stop);
gst_pad_push_event (wav->srcpad,
gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.start, wav->segment.last_stop, wav->segment.time));
}
}
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
......@@ -1080,7 +1090,6 @@ gst_wavparse_stream_headers (GstWavParse * wav)
/* loop headers until we get data */
while (!gotdata) {
if (wav->streaming) {
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
return GST_FLOW_OK;
......@@ -1322,7 +1331,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
#define MAX_BUFFER_SIZE 4096
static GstFlowReturn
gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
gst_wavparse_stream_data (GstWavParse * wav)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
......@@ -1331,8 +1340,9 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first)
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT,
wav->offset, wav->end_offset);
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)
......@@ -1350,8 +1360,12 @@ iterate_adapter:
"from the sinkpad", desired);
if (wav->streaming) {
if (gst_adapter_available (wav->adapter) < desired)
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);
......@@ -1362,14 +1376,15 @@ iterate_adapter:
goto pull_error;
}
obtained = GST_BUFFER_SIZE (buf);
/* 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 (first) {
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, buf);
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
......@@ -1400,16 +1415,17 @@ iterate_adapter:
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
} else {
GST_DEBUG ("Srcpad not linked!");
gst_buffer_unref (buf);
goto not_linked;
}
if (obtained < wav->dataleft) {
wav->dataleft -= obtained;
wav->offset += obtained;
//wav->offset += 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,
......@@ -1451,6 +1467,11 @@ push_error:
GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
return res;
}
not_linked:
{
GST_DEBUG_OBJECT (wav, "Srcpad not linked!");
return GST_FLOW_ERROR;
}
}
static void
......@@ -1459,8 +1480,11 @@ 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;
......@@ -1468,15 +1492,15 @@ gst_wavparse_loop (GstPad * pad)
/* 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:
......@@ -1505,24 +1529,30 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (wav, "adapter_push %" G_GINT64_FORMAT " 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;
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);
}
......
/* 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 {
......
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