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gstreamer/gst/wavparse/gstwavparse.c
Sebastian Dröge 233644df33 gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the element driving the pipeline is responsible f... 
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
2007-09-14 09:40:49 +00:00

2186 lines
64 KiB
C
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/* -*- 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, 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
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-wavparse
*
* <refsect2>
* <para>
* Parse a .wav file into raw or compressed audio.
* </para>
* <para>
* Wavparse supports both push and pull mode operations, making it possible to
* stream from a network source.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location=sine.wav ! 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 ! queue ! wavparse ! audioconvert ! alsasink
* </programlisting>
* Stream data from a network url.
* </para>
* </refsect2>
*
* Last reviewed on 2007-02-14 (0.10.6)
*/
/*
* TODO:
* http://replaygain.hydrogenaudio.org/file_format_wav.html
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include "gstwavparse.h"
#include "gst/riff/riff-ids.h"
#include "gst/riff/riff-media.h"
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
#define GST_CAT_DEFAULT (wavparse_debug)
static void gst_wavparse_dispose (GObject * object);
static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
gboolean active);
static gboolean gst_wavparse_send_event (GstElement * element,
GstEvent * event);
static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
GstStateChange transition);
static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
static gboolean gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
static void gst_wavparse_loop (GstPad * pad);
static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
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,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wav")
);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
GST_BOILERPLATE_FULL (GstWavParse, gst_wavparse, GstElement,
GST_TYPE_ELEMENT, DEBUG_INIT);
static void
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *src_template;
/* register pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template_factory));
src_template = gst_pad_template_new ("wavparse_src", GST_PAD_SRC,
GST_PAD_SOMETIMES, gst_riff_create_audio_template_caps ());
gst_element_class_add_pad_template (element_class, src_template);
gst_element_class_set_details (element_class, &gst_wavparse_details);
}
static void
gst_wavparse_class_init (GstWavParseClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *object_class;
gstelement_class = (GstElementClass *) klass;
object_class = (GObjectClass *) klass;
parent_class = g_type_class_peek_parent (klass);
object_class->dispose = gst_wavparse_dispose;
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
}
static void
gst_wavparse_dispose (GObject * object)
{
GstWavParse *wav;
GST_DEBUG ("WAV: Dispose");
wav = GST_WAVPARSE (object);
if (wav->adapter) {
g_object_unref (wav->adapter);
wav->adapter = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wavparse_reset (GstWavParse * wavparse)
{
wavparse->state = GST_WAVPARSE_START;
/* These will all be set correctly in the fmt chunk */
wavparse->depth = 0;
wavparse->rate = 0;
wavparse->width = 0;
wavparse->channels = 0;
wavparse->blockalign = 0;
wavparse->bps = 0;
wavparse->fact = 0;
wavparse->offset = 0;
wavparse->end_offset = 0;
wavparse->dataleft = 0;
wavparse->datasize = 0;
wavparse->datastart = 0;
wavparse->duration = 0;
wavparse->got_fmt = FALSE;
wavparse->first = TRUE;
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
wavparse->seek_event = NULL;
if (wavparse->adapter)
gst_adapter_clear (wavparse->adapter);
}
static void
gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
{
gst_wavparse_reset (wavparse);
/* sink */
wavparse->sinkpad =
gst_pad_new_from_static_template (&sink_template_factory, "sink");
gst_pad_set_activate_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
gst_pad_set_activatepull_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
/* src, will be created later */
wavparse->srcpad = NULL;
}
static void
gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
{
if (wavparse->srcpad) {
gst_element_remove_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
wavparse->srcpad = NULL;
}
}
static void
gst_wavparse_create_sourcepad (GstWavParse * wavparse)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavparse);
GstPadTemplate *src_template;
/* destroy previous one */
gst_wavparse_destroy_sourcepad (wavparse);
/* source */
src_template = gst_element_class_get_pad_template (klass, "wavparse_src");
wavparse->srcpad = gst_pad_new_from_template (src_template, "src");
gst_pad_use_fixed_caps (wavparse->srcpad);
gst_pad_set_query_type_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));
gst_pad_set_query_function (wavparse->srcpad,
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");
}
/* Compute (value * nom) % denom, avoiding overflow. This can be used
* to perform ceiling or rounding division together with
* gst_util_uint64_scale[_int]. */
#define uint64_scale_modulo(val, nom, denom) \
((val % denom) * (nom % denom) % denom)
/* Like gst_util_uint64_scale, but performs ceiling division. */
static guint64
uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
{
guint64 result = gst_util_uint64_scale (val, num, denom);
if (uint64_scale_modulo (val, num, denom) == 0)
return result;
else
return result + 1;
}
/* Like gst_util_uint64_scale, but performs ceiling division. */
static guint64
uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom)
{
guint64 result = gst_util_uint64_scale_int (val, num, denom);
if (uint64_scale_modulo (val, num, denom) == 0)
return result;
else
return result + 1;
}
#if 0
static void
gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
{
guint32 got_bytes;
GstByteStream *bs = wavparse->bs;
gst_riff_chunk *temp_chunk, chunk;
guint8 *tempdata;
struct _gst_riff_labl labl, *temp_labl;
struct _gst_riff_ltxt ltxt, *temp_ltxt;
struct _gst_riff_note note, *temp_note;
char *label_name;
GstProps *props;
GstPropsEntry *entry;
GstCaps *new_caps;
GList *caps = NULL;
props = wavparse->metadata->properties;
while (len > 0) {
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
if (got_bytes != sizeof (gst_riff_chunk)) {
return;
}
temp_chunk = (gst_riff_chunk *) tempdata;
chunk.id = GUINT32_FROM_LE (temp_chunk->id);
chunk.size = GUINT32_FROM_LE (temp_chunk->size);
if (chunk.size == 0) {
gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
len -= sizeof (gst_riff_chunk);
continue;
}
switch (chunk.id) {
case GST_RIFF_adtl_labl:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_labl));
if (got_bytes != sizeof (struct _gst_riff_labl)) {
return;
}
temp_labl = (struct _gst_riff_labl *) tempdata;
labl.id = GUINT32_FROM_LE (temp_labl->id);
labl.size = GUINT32_FROM_LE (temp_labl->size);
labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
len -= sizeof (struct _gst_riff_labl);
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
if (got_bytes != labl.size - 4) {
return;
}
label_name = (char *) tempdata;
gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
len -= (((labl.size - 4) + 1) & ~1);
new_caps = gst_caps_new ("label",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
"name", G_TYPE_STRING (label_name), NULL));
if (gst_props_get (props, "labels", &caps, NULL)) {
caps = g_list_append (caps, new_caps);
} else {
caps = g_list_append (NULL, new_caps);
entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
gst_props_add_entry (props, entry);
}
break;
case GST_RIFF_adtl_ltxt:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_ltxt));
if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
return;
}
temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
len -= sizeof (struct _gst_riff_ltxt);
if (ltxt.size - 20 > 0) {
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
if (got_bytes != ltxt.size - 20) {
return;
}
gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
len -= (((ltxt.size - 20) + 1) & ~1);
label_name = (char *) tempdata;
} else {
label_name = "";
}
new_caps = gst_caps_new ("ltxt",
"application/x-gst-metadata",
gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
"name", G_TYPE_STRING (label_name),
"length", G_TYPE_INT (ltxt.length), NULL));
if (gst_props_get (props, "ltxts", &caps, NULL)) {
caps = g_list_append (caps, new_caps);
} else {
caps = g_list_append (NULL, new_caps);
entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
gst_props_add_entry (props, entry);
}
break;
case GST_RIFF_adtl_note:
got_bytes =
gst_bytestream_peek_bytes (bs, &tempdata,
sizeof (struct _gst_riff_note));
if (got_bytes != sizeof (struct _gst_riff_note)) {
return;
}
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))
goto no_fmt;
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
if (wav->channels == 0)
goto no_channels;
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
if (wav->bps <= 0)
goto no_bps;
/* 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 == NULL)
goto no_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_CAST (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
return TRUE;
/* ERRORS */
no_fmt:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
("No FMT tag found"));
return FALSE;
}
no_channels:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain zero channels - invalid data"));
g_free (header);
return FALSE;
}
no_bps:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to bitrate of <= zero - invalid data"));
g_free (header);
return FALSE;
}
no_caps:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
return FALSE;
}
}
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 a 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_CAST (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, };
gint64 last_stop;
if (event) {
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
if (format != wav->segment.format) {
GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
gst_format_get_name (format),
gst_format_get_name (wav->segment.format));
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, cur,
&wav->segment.format, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, stop,
&wav->segment.format, &stop);
if (!res)
goto no_format;
format = wav->segment.format;
}
} else {
GST_DEBUG_OBJECT (wav, "doing seek without event");
flags = 0;
rate = 1.0;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
if (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);
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* save current position */
last_stop = wav->segment.last_stop;
GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (wav->bps > 0)
wav->offset =
uint64_ceiling_scale (seeksegment.last_stop, (guint64) wav->bps,
GST_SECOND);
else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
wav->offset =
uint64_ceiling_scale (seeksegment.last_stop, bps, GST_SECOND);
} else
wav->offset = seeksegment.last_stop;
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) {
if (wav->bps > 0)
wav->end_offset =
uint64_ceiling_scale (stop, (guint64) wav->bps, GST_SECOND);
else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
wav->end_offset = uint64_ceiling_scale (stop, bps, GST_SECOND);
} else
wav->end_offset = stop;
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);
/* this is the range of bytes we will use for playback */
wav->offset = MIN (wav->offset, wav->end_offset);
wav->dataleft = wav->end_offset - wav->offset;
GST_DEBUG_OBJECT (wav,
"seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
if (wav->srcpad) {
if (flush) {
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
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 previous last_stop. */
GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.accum, wav->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (wav->close_segment)
gst_event_unref (wav->close_segment);
wav->close_segment = gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.accum, wav->segment.last_stop, wav->segment.accum);
/* keep track of our last_stop */
seeksegment.accum = wav->segment.last_stop;
}
}
/* now we did the seek and can activate the new segment values */
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_start (GST_OBJECT_CAST (wav),
wav->segment.format, wav->segment.last_stop));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (wav, "Creating 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->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment =
gst_event_new_new_segment (FALSE, wav->segment.rate,
wav->segment.format, wav->segment.last_stop, stop,
wav->segment.last_stop);
/* mark discont if we are going to stream from another position. */
if (last_stop != wav->segment.last_stop) {
GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
wav->discont = TRUE;
}
/* and start the streaming task again */
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 */
negative_rate:
{
GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
return FALSE;
}
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;
}
}
/*
* gst_wavparse_calculate_duration:
* @wav Wavparse object
*
* Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
* fallback.
*
* Returns: %TRUE if duration is available.
*/
static gboolean
gst_wavparse_calculate_duration (GstWavParse * wav)
{
if (wav->duration > 0)
return TRUE;
if (wav->bps > 0) {
wav->duration =
uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps);
GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
} else if (wav->fact) {
wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate);
GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
}
return FALSE;
}
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;
gchar *codec_name = NULL;
GstEvent **event_p;
GstFormat bformat;
gint64 upstream_size = 0;
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;
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
buf = gst_adapter_take_buffer (wav->adapter, size);
} else {
if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (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 || tag == GST_RIFF_TAG_LIST) {
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
GST_FOURCC_ARGS (tag));
gst_buffer_unref (buf);
buf = NULL;
continue;
}
if (tag != GST_RIFF_TAG_fmt)
goto invalid_wav;
if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
&extra)))
goto parse_header_error;
buf = NULL; /* parse_strf_auds() took ownership of buffer */
/* do sanity checks of header fields */
if (header->channels == 0)
goto no_channels;
if (header->rate == 0)
goto no_rate;
GST_DEBUG_OBJECT (wav, "creating the caps");
/* 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);
if (!caps)
goto unknown_format;
/* do more sanity checks of header fields
* (these can be sanitized by gst_riff_create_audio_caps()
*/
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
wav->blockalign = header->blockalign;
wav->depth = header->size;
wav->av_bps = header->av_bps;
wav->vbr = FALSE;
g_free (header);
/* do format specific handling */
switch (wav->format) {
case GST_RIFF_WAVE_FORMAT_MPEGL12:
case GST_RIFF_WAVE_FORMAT_MPEGL3:
{
/* Note: workaround for mp2/mp3 embedded in wav, that relies on the
* bitrate inside the mpeg stream */
GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
wav->bps = 0;
break;
}
case GST_RIFF_WAVE_FORMAT_PCM:
if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0))
goto invalid_blockalign;
/* fall through */
default:
if (wav->av_bps > wav->blockalign * wav->rate)
goto invalid_bps;
/* use the configured bps */
wav->bps = wav->av_bps;
break;
}
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;
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, "av_bps = %u", (guint) wav->av_bps);
GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
/* bps can be 0 when we don't have a valid bitrate (mostly for compressed
* formats). This will make the element output a BYTE format segment and
* will not timestamp the outgoing buffers.
*/
GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
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;
}
}
bformat = GST_FORMAT_BYTES;
gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size);
GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
/* loop headers until we get data */
while (!gotdata) {
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_INFO_OBJECT (wav,
"Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
GST_FOURCC_ARGS (tag), wav->offset);
/* 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);
}
GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
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) {
#if 0
/* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
wav->bps =
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
(guint64) wav->fact);
GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps);
#endif
wav->vbr = TRUE;
}
if (gst_wavparse_calculate_duration (wav)) {
gst_segment_init (&wav->segment, GST_FORMAT_TIME);
gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, wav->duration);
} else {
/* no bitrate, let downstream peer do the math, we'll feed it bytes. */
gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize);
}
/* 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);
/* we just started, we are discont */
wav->discont = TRUE;
wav->state = GST_WAVPARSE_DATA;
return GST_FLOW_OK;
/* ERROR */
invalid_wav:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (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"));
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_rate:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream with sample_rate == 0 - invalid data"));
g_free (header);
g_free (codec_name);
return GST_FLOW_ERROR;
}
invalid_blockalign:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims blockalign = %u, which is more than %u - invalid data",
header->blockalign,
header->channels * (guint) ceil (header->size / 8.0)));
g_free (codec_name);
return GST_FLOW_ERROR;
}
invalid_bps:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims av_bsp = %u, which is more than %u - invalid data",
wav->av_bps, wav->blockalign * wav->rate));
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 (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 (header);
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;
/* _take flushes the data */
tmp = gst_adapter_take_buffer (wav->adapter, 12);
GST_DEBUG ("Parsing wav header");
if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (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 };
GST_DEBUG_OBJECT (wav, "adding src pad");
if (wav->caps) {
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_CAST (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
if (wav->close_segment) {
GST_DEBUG_OBJECT (wav, "Send close segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->close_segment);
wav->close_segment = NULL;
}
if (wav->start_segment) {
GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
}
if (wav->tags) {
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (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, duration;
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_adapter_take_buffer (wav->adapter, 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;
/* this will also push the segment events */
gst_wavparse_add_src_pad (wav, buf);
} else {
/* If we have a pending close/start segment, send it now. */
if (G_UNLIKELY (wav->close_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->close_segment);
wav->close_segment = NULL;
}
if (G_UNLIKELY (wav->start_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
}
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions in bytes */
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;
if (wav->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps);
next_timestamp =
uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
} else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps);
next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps);
duration = next_timestamp - timestamp;
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
* other positions and durations have unknown timestamp. */
if (pos == 0)
timestamp = 0;
else
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_BYTES, nextpos);
}
if ((pos > 0) && wav->vbr) {
/* don't set timestamps for VBR files if it's not the first buffer */
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
}
if (wav->discont) {
GST_DEBUG_OBJECT (wav, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
wav->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
/* 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 (timestamp), GST_TIME_ARGS (duration),
GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < wav->dataleft) {
wav->offset += obtained;
wav->dataleft -= obtained;
} else {
wav->offset += wav->dataleft;
wav->dataleft = 0;
}
/* 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");
return GST_FLOW_UNEXPECTED;
}
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_INFO_OBJECT (wav,
"Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
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_INFO_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_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_DATA;
GST_INFO_OBJECT (wav, "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:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
wav->segment_running = FALSE;
gst_pad_pause_task (pad);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* add pad before we perform EOS */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, NULL);
}
/* perform EOS logic */
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_CAST (wav),
gst_message_new_segment_done (GST_OBJECT_CAST (wav),
wav->segment.format, stop));
} else {
if (wav->srcpad != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
} else {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
(_("Internal data flow error.")),
("streaming task paused, reason %s (%d)", reason, ret));
if (wav->srcpad != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
}
return;
}
}
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_INFO_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
goto done;
if (wav->state != GST_WAVPARSE_HEADER)
break;
/* otherwise fall-through */
case GST_WAVPARSE_HEADER:
GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto done;
if (!wav->got_fmt || wav->datastart == 0)
break;
wav->state = GST_WAVPARSE_DATA;
GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
/* fall-through */
case GST_WAVPARSE_DATA:
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto done;
break;
default:
g_assert_not_reached ();
}
done:
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 (*dest_format == src_format) {
*dest_value = src_value;
return TRUE;
}
if ((wavparse->bps == 0) && !wavparse->fact)
goto no_bps_fact;
GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
gst_format_get_name (src_format), gst_format_get_name (*dest_format));
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / wavparse->bytes_per_sample;
/* make sure we end up on a sample boundary */
*dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
/* src_value + datastart = offset */
GST_INFO_OBJECT (wavparse,
"src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
wavparse->offset);
if (wavparse->bps > 0)
*dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
(guint64) wavparse->bps);
else {
guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
wavparse->rate, wavparse->fact);
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, bps);
}
break;
default:
res = FALSE;
goto done;
}
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 (src_value, GST_SECOND,
(guint64) wavparse->rate);
break;
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
if (wavparse->bps > 0)
*dest_value = gst_util_uint64_scale (src_value,
(guint64) wavparse->bps, GST_SECOND);
else {
guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
wavparse->rate, wavparse->fact);
*dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
}
/* make sure we end up on a sample boundary */
*dest_value -= *dest_value % wavparse->blockalign;
break;
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale (src_value,
(guint64) 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_bps_fact:
{
GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, 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,
GST_QUERY_SEEKING,
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;
/* this is not very precise, as we have pushed severla buffer upstream for prerolling */
curb = wav->offset - wav->datastart;
gst_query_parse_position (query, &format, NULL);
GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
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 duration;
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:{
if (gst_wavparse_calculate_duration (wav)) {
duration = wav->duration;
} else {
format = GST_FORMAT_BYTES;
duration = wav->datasize;
}
break;
}
default:
format = GST_FORMAT_BYTES;
duration = wav->datasize;
break;
}
gst_query_set_duration (query, format, duration);
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;
}
case GST_QUERY_SEEKING:{
GstFormat fmt;
gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
if (fmt == GST_FORMAT_TIME) {
gboolean seekable = TRUE;
if ((wav->bps == 0) && !wav->fact) {
seekable = FALSE;
} else if (!gst_wavparse_calculate_duration (wav)) {
seekable = FALSE;
}
gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
0, wav->duration);
res = TRUE;
}
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);
gst_event_unref (event);
break;
}
default:
res = gst_pad_push_event (wavparse->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_wavparse_sink_activate (GstPad * sinkpad)
{
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
gboolean res;
if (wav->adapter)
gst_object_unref (wav->adapter);
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:
gst_wavparse_destroy_sourcepad (wav);
gst_wavparse_reset (wav);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
gst_riff_init ();
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)
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