diff -r 4c282e7dd6d3 -r 5505e8908944 gst_plugins_good/gst/wavparse/gstwavparse.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/gst_plugins_good/gst/wavparse/gstwavparse.c Fri Jan 22 09:59:59 2010 +0200 @@ -0,0 +1,1983 @@ +/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * Copyright (C) <2006> Nokia Corporation, Stefan Kost . + * + * 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 + * + * + * + * Parse a .wav file into raw or compressed audio. + * + * + * This element currently only supports pull based scheduling. + * + * Example launch line + * + * + * gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink + * + * Read a wav file and output to the soundcard using the ALSA element. The + * wav file is assumed to contain raw uncompressed samples. + * + * + * + * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink + * + * Stream data from + * + * + * + * Last reviewed on 2006-03-03 (0.10.3) + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "string.h" +#include "gstwavparse.h" +#include +#include +#include + +#ifndef __SYMBIAN32__ +#include +#else +#include "gst/gst-i18n-plugin.h" +#endif + +#ifdef __SYMBIAN32__ +#include +#endif + +#ifndef G_MAXUINT32 +#define G_MAXUINT32 0xffffffff +#endif + +GST_DEBUG_CATEGORY_STATIC (wavparse_debug); +#define GST_CAT_DEFAULT (wavparse_debug) + +static void gst_wavparse_base_init (gpointer g_class); +static void gst_wavparse_class_init (GstWavParseClass * klass); +static void gst_wavparse_init (GstWavParse * wavparse); +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 GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf); +static GstStateChangeReturn gst_wavparse_change_state (GstElement * element, + GstStateChange transition); + +static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query); +static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad); +static gboolean gst_wavparse_pad_convert (GstPad * pad, + GstFormat src_format, + gint64 src_value, GstFormat * dest_format, gint64 * dest_value); + +static void gst_wavparse_loop (GstPad * pad); +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 "); + +static GstStaticPadTemplate sink_template_factory = +//GST_STATIC_PAD_TEMPLATE ("wavparse_sink", +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-wav") + ); + +/* the pad is marked a sometimes and is added to the element when the + * exact type is known. This makes it much easier for a static autoplugger + * to connect the right decoder when needed. + */ +static GstStaticPadTemplate src_template_factory = + // GST_STATIC_PAD_TEMPLATE ("wavparse_src", + GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("audio/x-raw-int, " + "endianness = (int) little_endian, " + "signed = (boolean) { true, false }, " + "width = (int) { 8, 16, 24, 32 }, " + "depth = (int) { 8, 16, 24, 32 }, " + "rate = (int) [ 8000, 96000 ], " + "channels = (int) [ 1, 8 ]; " + "audio/mpeg, " + "mpegversion = (int) 1, " + "layer = (int) [ 1, 3 ], " + "rate = (int) [ 8000, 48000 ], " + "channels = (int) [ 1, 2 ]; " + "audio/x-alaw, " + "rate = (int) [ 8000, 48000 ], " + "channels = (int) [ 1, 2 ]; " + "audio/x-mulaw, " + "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];" + "audio/x-adpcm, " + "layout = (string) microsoft, " + "block_align = (int) [ 1, 8192 ], " + "rate = (int) [ 8000, 48000 ], " + "channels = (int) [ 1, 2 ]; " + "audio/x-adpcm, " + "layout = (string) dvi, " + "block_align = (int) [ 1, 8192 ], " + "rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];" + "audio/x-vnd.sony.atrac3;" + "audio/x-dts;" "audio/x-wma, " "wmaversion = (int) [ 1, 2 ]") + ); + + +static GstElementClass *parent_class = NULL; + +GType +gst_wavparse_get_type (void) +{ + static GType wavparse_type = 0; + + if (!wavparse_type) { + static const GTypeInfo wavparse_info = { + sizeof (GstWavParseClass), + gst_wavparse_base_init, + NULL, + (GClassInitFunc) gst_wavparse_class_init, + NULL, + NULL, + sizeof (GstWavParse), + 0, + (GInstanceInitFunc) gst_wavparse_init, + }; + + wavparse_type = + g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse", + &wavparse_info, 0); + } + return wavparse_type; +} + + +static void +gst_wavparse_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + /* register src pads */ + 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 +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->get_property = gst_wavparse_get_property; + 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) +{ + #ifndef __SYMBIAN32__ + GST_DEBUG("WAV: Dispose\n"); + #endif + GstWavParse *wav = GST_WAVPARSE (object); + #ifdef __SYMBIAN32__ + GST_DEBUG("WAV: Dispose\n"); + #endif + + 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->offset = 0; + wavparse->end_offset = 0; + wavparse->dataleft = 0; + wavparse->datasize = 0; + wavparse->datastart = 0; + wavparse->got_fmt = FALSE; + wavparse->first = TRUE; + + if (wavparse->seek_event) + gst_event_unref (wavparse->seek_event); + wavparse->seek_event = NULL; + + /* we keep the segment info in time */ + gst_segment_init (&wavparse->segment, GST_FORMAT_TIME); +} + +static void +gst_wavparse_init (GstWavParse * wavparse) +{ + 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 (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 (wavparse), wavparse->srcpad); + wavparse->srcpad = NULL; + } +} + +static void +gst_wavparse_create_sourcepad (GstWavParse * wavparse) +{ + /* destroy previous one */ + gst_wavparse_destroy_sourcepad (wavparse); + + /* source */ + wavparse->srcpad = + gst_pad_new_from_static_template (&src_template_factory, "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"); +} + +static void +gst_wavparse_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec) +{ + GstWavParse *wavparse; + + wavparse = GST_WAVPARSE (object); + + switch (prop_id) { + default: + break; + } +} + + + +#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)) { + g_warning ("Not fmt"); + return FALSE; + } + + wav->format = header->format; + wav->rate = header->rate; + wav->channels = header->channels; + if (wav->channels == 0) { + GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), + ("Stream claims to contain zero channels - invalid data")); + g_free (header); + return FALSE; + } + wav->blockalign = header->blockalign; + wav->width = (header->blockalign * 8) / header->channels; + wav->depth = header->size; + wav->bps = header->av_bps; + if (wav->bps <= 0) { + GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), + ("Stream claims to bitrate of <= zero - invalid data")); + g_free (header); + return FALSE; + } + + /* Note: gst_riff_create_audio_caps might nedd to fix values in + * the header header depending on the format, so call it first */ + caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL); + + g_free (header); + + if (caps) { + gst_wavparse_create_sourcepad (wav); + gst_pad_use_fixed_caps (wav->srcpad); + gst_pad_set_active (wav->srcpad, TRUE); + gst_pad_set_caps (wav->srcpad, caps); + gst_caps_free (caps); + gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad); + gst_element_no_more_pads (GST_ELEMENT (wav)); + GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels); + } else { + GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL)); + return FALSE; + } + + return TRUE; +} + +static gboolean +gst_wavparse_other (GstWavParse * wav) +{ + guint32 tag, length; + + if (!gst_riff_peek_head (wav, &tag, &length, NULL)) { + GST_WARNING_OBJECT (wav, "could not peek head"); + return FALSE; + } + GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag, + (gchar *) & tag, length); + + switch (tag) { + case GST_RIFF_TAG_LIST: + if (!(tag = gst_riff_peek_list (wav))) { + GST_WARNING_OBJECT (wav, "could not peek list"); + return FALSE; + } + + switch (tag) { + case GST_RIFF_LIST_INFO: + if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) { + GST_WARNING_OBJECT (wav, "could not read list"); + return FALSE; + } + break; + + case GST_RIFF_LIST_adtl: + if (!gst_riff_read_skip (wav)) { + GST_WARNING_OBJECT (wav, "could not read skip"); + return FALSE; + } + break; + + default: + GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, + (gchar *) & tag); + if (!gst_riff_read_skip (wav)) { + GST_WARNING_OBJECT (wav, "could not read skip"); + return FALSE; + } + break; + } + + break; + + case GST_RIFF_TAG_data: + if (!gst_bytestream_flush (wav->bs, 8)) { + GST_WARNING_OBJECT (wav, "could not flush 8 bytes"); + return FALSE; + } + + GST_DEBUG_OBJECT (wav, "switching to data mode"); + wav->state = GST_WAVPARSE_DATA; + wav->datastart = gst_bytestream_tell (wav->bs); + if (length == 0) { + guint64 file_length; + + /* length is 0, data probably stretches to the end + * of file */ + GST_DEBUG_OBJECT (wav, "length is 0 trying to find length"); + /* get length of file */ + file_length = gst_bytestream_length (wav->bs); + if (file_length == -1) { + GST_DEBUG_OBJECT (wav, + "could not get file length, assuming data to eof"); + /* could not get length, assuming till eof */ + length = G_MAXUINT32; + } + if (file_length > G_MAXUINT32) { + GST_DEBUG_OBJECT (wav, "file length %lld, clipping to 32 bits"); + /* could not get length, assuming till eof */ + length = G_MAXUINT32; + } else { + GST_DEBUG_OBJECT (wav, "file length %lld, datalength", + file_length, length); + /* substract offset of datastart from length */ + length = file_length - wav->datastart; + GST_DEBUG_OBJECT (wav, "datalength %lld", length); + } + } + wav->datasize = (guint64) length; + break; + + case GST_RIFF_TAG_cue: + if (!gst_riff_read_skip (wav)) { + GST_WARNING_OBJECT (wav, "could not read skip"); + return FALSE; + } + break; + + default: + GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag); + if (!gst_riff_read_skip (wav)) + return FALSE; + break; + } + + return TRUE; +} +#endif + + + +static gboolean +gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf) +{ + guint32 doctype; + + if (!gst_riff_parse_file_header (element, buf, &doctype)) + return FALSE; + + if (doctype != GST_RIFF_RIFF_WAVE) + goto not_wav; + + return TRUE; + + /* ERRORS */ +not_wav: + + { + GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL), + ("File is not an WAVE file: %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (doctype))); + return FALSE; + } +} + +static GstFlowReturn +gst_wavparse_stream_init (GstWavParse * wav) +{ + GstFlowReturn res; + GstBuffer *buf = NULL; + + + if ((res = gst_pad_pull_range (wav->sinkpad, + wav->offset, 12, &buf)) != GST_FLOW_OK) + { + + return res; + } + else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf)) + return GST_FLOW_ERROR; + + wav->offset += 12; + + return GST_FLOW_OK; +} + +/* This function is used to perform seeks on the element in + * pull mode. + * + * It also works when event is NULL, in which case it will just + * start from the last configured segment. This technique is + * used when activating the element and to perform the seek in + * READY. + */ +static gboolean +gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) +{ + gboolean res; + gdouble rate; + GstEvent *newsegment; + GstFormat format; + GstSeekFlags flags; + GstSeekType cur_type, stop_type; + gint64 cur, stop; + gboolean flush; + gboolean update; + GstSegment seeksegment; + + if (event) { + GST_DEBUG_OBJECT (wav, "doing seek with event"); + + gst_event_parse_seek (event, &rate, &format, &flags, + &cur_type, &cur, &stop_type, &stop); + + /* we have to have a format as the segment format. Try to convert + * if not. */ + if (format != GST_FORMAT_TIME) { + GstFormat fmt; + + fmt = GST_FORMAT_TIME; + res = TRUE; + if (cur_type != GST_SEEK_TYPE_NONE) + res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur); + if (res && stop_type != GST_SEEK_TYPE_NONE) + res = gst_pad_query_convert (wav->srcpad, format, stop, &fmt, &stop); + if (!res) + goto no_format; + + format = fmt; + } + } else { + GST_DEBUG_OBJECT (wav, "doing seek without event"); + flags = 0; + } + + flush = flags & GST_SEEK_FLAG_FLUSH; + + if (flush && wav->srcpad) { + GST_DEBUG_OBJECT (wav, "sending flush start"); + gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ()); + } else { + gst_pad_pause_task (wav->sinkpad); + } + + GST_PAD_STREAM_LOCK (wav->sinkpad); + + /* copy segment, we need this because we still need the old + * segment when we close the current segment. */ + memcpy (&seeksegment, &wav->segment, sizeof (GstSegment)); + + if (event) { + GST_DEBUG_OBJECT (wav, "configuring seek"); + gst_segment_set_seek (&seeksegment, rate, format, flags, + cur_type, cur, stop_type, stop, &update); + } + + if ((stop = seeksegment.stop) == -1) + stop = seeksegment.duration; + + if (cur_type != GST_SEEK_TYPE_NONE) { + wav->offset = + gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND); + wav->offset -= wav->offset % wav->bytes_per_sample; + wav->offset += wav->datastart; + } + + if (stop != -1) { + wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND); + wav->end_offset += + wav->bytes_per_sample - (wav->end_offset % wav->bytes_per_sample); + wav->end_offset += wav->datastart; + } else { + wav->end_offset = wav->datasize + wav->datastart; + } + wav->offset = MIN (wav->offset, wav->end_offset); + wav->dataleft = wav->end_offset - wav->offset; + + GST_DEBUG_OBJECT (wav, + "seek: offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT ", segment %" + GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, wav->offset, wav->end_offset, + GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop)); + + /* prepare for streaming again */ + if (wav->srcpad) { + if (flush) { + GST_DEBUG_OBJECT (wav, "sending flush stop"); + gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ()); + } else if (wav->segment_running) { + /* we are running the current segment and doing a non-flushing seek, + * close the segment first based on the last_stop. */ + GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT + " to %" G_GINT64_FORMAT, wav->segment.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)); + + if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) { + gst_element_post_message (GST_ELEMENT (wav), + gst_message_new_segment_start (GST_OBJECT (wav), + wav->segment.format, wav->segment.last_stop)); + } + + /* now send the newsegment */ + GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT + " to %" G_GINT64_FORMAT, wav->segment.start, stop); + + newsegment = + gst_event_new_new_segment (FALSE, wav->segment.rate, + wav->segment.format, wav->segment.last_stop, stop, wav->segment.time); + + if (wav->srcpad) { + gst_pad_push_event (wav->srcpad, newsegment); + } else { + /* send later when we actually create the source pad */ + g_assert (wav->newsegment == NULL); + wav->newsegment = newsegment; + } + + wav->segment_running = TRUE; + if (!wav->streaming) { + gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop, + wav->sinkpad); + } + + GST_PAD_STREAM_UNLOCK (wav->sinkpad); + + return TRUE; + + /* ERRORS */ +no_format: + { + GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted."); + return FALSE; + } +} + +/* + * gst_wavparse_peek_chunk_info: + * @wav Wavparse object + * @tag holder for tag + * @size holder for tag size + * + * Peek next chunk info (tag and size) + * + * Returns: %TRUE when one chunk info has been got from the adapter + */ +static gboolean +gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size) +{ + const guint8 *data = NULL; + + if (gst_adapter_available (wav->adapter) < 8) { + return FALSE; + } + + GST_DEBUG ("Next chunk size is %d bytes", *size); + data = gst_adapter_peek (wav->adapter, 8); + *tag = GST_READ_UINT32_LE (data); + *size = GST_READ_UINT32_LE (data + 4); + + 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; + + gst_wavparse_peek_chunk_info (wav, tag, size); + GST_DEBUG ("Need to peek chunk of %d bytes", *size); + peek_size = (*size + 1) & ~1; + + if (gst_adapter_available (wav->adapter) >= (8 + peek_size)) { + return TRUE; + } else { + return FALSE; + } +} + +static gboolean +gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len) +{ + gboolean res = FALSE; + GstFormat fmt = GST_FORMAT_BYTES; + GstPad *peer; + + if ((peer = gst_pad_get_peer (wav->sinkpad))) { + res = gst_pad_query_duration (peer, &fmt, len); + gst_object_unref (peer); + } + + return res; +} + +static GstFlowReturn +gst_wavparse_stream_headers (GstWavParse * wav) +{ + GstFlowReturn res; + GstBuffer *buf; + gst_riff_strf_auds *header = NULL; + guint32 tag, size; + gboolean gotdata = FALSE; + GstCaps *caps; + gint64 duration; + gchar *codec_name = NULL; + GstEvent **event_p; + + + if (!wav->got_fmt) { + GstBuffer *extra; + + /* The header start with a 'fmt ' tag */ + + if (wav->streaming) { + if (!gst_wavparse_peek_chunk (wav, &tag, &size)) + return GST_FLOW_OK; + + buf = gst_buffer_new (); + gst_buffer_ref (buf); + gst_adapter_flush (wav->adapter, 8); + wav->offset += 8; + GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size); + GST_BUFFER_SIZE (buf) = size; + + } else { + if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad, + &wav->offset, &tag, &buf)) != GST_FLOW_OK) + return res; + } + + if (tag != GST_RIFF_TAG_fmt) + goto invalid_wav; + + if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra))) + goto parse_header_error; + + if (wav->streaming) { + gst_adapter_flush (wav->adapter, size); + wav->offset += size; + GST_BUFFER_DATA (buf) = NULL; + gst_buffer_unref (buf); + } + + /* Note: gst_riff_create_audio_caps might nedd to fix values in + * the header header depending on the format, so call it first */ + caps = + gst_riff_create_audio_caps (header->format, NULL, header, extra, + NULL, &codec_name); + + if (extra) + gst_buffer_unref (extra); + + wav->format = header->format; + wav->rate = header->rate; + wav->channels = header->channels; + + 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_bitrate; + + wav->bytes_per_sample = wav->channels * wav->width / 8; + if (wav->bytes_per_sample <= 0) + goto no_bytes_per_sample; + + g_free (header); + + if (!caps) + goto unknown_format; + + GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign); + GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width); + GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth); + GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps); + + /* 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; + } + + GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate, + wav->channels); + } + + /* 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); + } + + /* + 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:{ + gint64 upstream_size; + + GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size); + gotdata = TRUE; + if (wav->streaming) { + gst_adapter_flush (wav->adapter, 8); + } else { + gst_buffer_unref (buf); + } + wav->offset += 8; + wav->datastart = wav->offset; + /* file might be truncated */ + if (gst_wavparse_get_upstream_size (wav, &upstream_size)) { + size = MIN (size, (upstream_size - wav->datastart)); + } + wav->datasize = size; + wav->dataleft = size; + wav->end_offset = size + wav->datastart; + 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); + } + } + } + + GST_DEBUG_OBJECT (wav, "Finished parsing headers"); + + duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps); + GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT, + GST_TIME_ARGS (duration)); + gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration); + + /* now we have all the info to perform a pending seek if any, if no + * event, this will still do the right thing and it will also send + * the right newsegment event downstream. */ + gst_wavparse_perform_seek (wav, wav->seek_event); + /* remove pending event */ + event_p = &wav->seek_event; + gst_event_replace (event_p, NULL); + + wav->state = GST_WAVPARSE_DATA; + + return GST_FLOW_OK; + + /* ERROR */ +invalid_wav: + { + GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), + ("Invalid WAV header (no fmt at start): %" + GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag))); + g_free (codec_name); + + return GST_FLOW_ERROR; + } +parse_header_error: + { + GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), + ("Couldn't parse audio header")); + gst_buffer_unref (buf); + g_free (codec_name); + + return GST_FLOW_ERROR; + } +no_channels: + { + GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), + ("Stream claims to contain no channels - invalid data")); + g_free (header); + g_free (codec_name); + return GST_FLOW_ERROR; + } +no_bitrate: + { + GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), + ("Stream claims to have a bitrate of <= zero - invalid data")); + g_free (header); + g_free (codec_name); + return GST_FLOW_ERROR; + } +no_bytes_per_sample: + { + GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), + ("could not caluclate bytes per sample - invalid data")); + g_free (header); + g_free (codec_name); + return GST_FLOW_ERROR; + } +unknown_format: + { + GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), + ("No caps found for format 0x%x, %d channels, %d Hz", + wav->format, wav->channels, wav->rate)); + g_free (codec_name); + return GST_FLOW_ERROR; + } +header_read_error: + { + GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't read in header")); + g_free (codec_name); + return GST_FLOW_ERROR; + } +} + + +/* + * Read WAV file tag when streaming + */ +static GstFlowReturn +gst_wavparse_parse_stream_init (GstWavParse * wav) +{ + if (gst_adapter_available (wav->adapter) >= 12) { + GstBuffer *tmp = gst_buffer_new (); + + /* _take flushes the data */ + GST_BUFFER_DATA (tmp) = gst_adapter_take (wav->adapter, 12); + GST_BUFFER_SIZE (tmp) = 12; + + GST_DEBUG ("Parsing wav header"); + if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) { + return GST_FLOW_ERROR; + } + + wav->offset += 12; + /* Go to next state */ + wav->state = GST_WAVPARSE_HEADER; + } + return GST_FLOW_OK; +} + +/* handle an event sent directly to the element. + * + * This event can be sent either in the READY state or the + * >READY state. The only event of interest really is the seek + * event. + * + * In the READY state we can only store the event and try to + * respect it when going to PAUSED. We assume we are in the + * READY state when our parsing state != GST_WAVPARSE_DATA. + * + * When we are steaming, we can simply perform the seek right + * away. + */ +static gboolean +gst_wavparse_send_event (GstElement * element, GstEvent * event) +{ + GstWavParse *wav = GST_WAVPARSE (element); + gboolean res = FALSE; + GstEvent **event_p; + + GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_SEEK: + if (wav->state == GST_WAVPARSE_DATA) { + /* we can handle the seek directly when streaming data */ + res = gst_wavparse_perform_seek (wav, event); + } else { + GST_DEBUG_OBJECT (wav, "queuing seek for later"); + + event_p = &wav->seek_event; + gst_event_replace (event_p, event); + + /* we always return true */ + res = TRUE; + } + break; + default: + break; + } + gst_event_unref (event); + return res; +} + +static void +gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) +{ + GstStructure *s; + const guint8 dts_marker[] = { 0xFF, 0x1F, 0x00, 0xE8, 0xF1, 0x07 }; + + + s = gst_caps_get_structure (wav->caps, 0); + if (gst_structure_has_name (s, "audio/x-raw-int") && + GST_BUFFER_SIZE (buf) > 6 && + memcmp (GST_BUFFER_DATA (buf), dts_marker, 6) == 0) { + + GST_WARNING_OBJECT (wav, "Found DTS marker in file marked as raw PCM"); + gst_caps_unref (wav->caps); + wav->caps = gst_caps_from_string ("audio/x-dts"); + + gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, + GST_TAG_AUDIO_CODEC, "dts", NULL); + + } + + gst_wavparse_create_sourcepad (wav); + gst_pad_set_active (wav->srcpad, TRUE); + gst_pad_set_caps (wav->srcpad, wav->caps); + gst_caps_replace (&wav->caps, NULL); + + gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad); + + + gst_element_no_more_pads (GST_ELEMENT (wav)); + + GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad"); + gst_pad_push_event (wav->srcpad, wav->newsegment); + wav->newsegment = NULL; + + + if (wav->tags) { + gst_element_found_tags_for_pad (GST_ELEMENT (wav), wav->srcpad, wav->tags); + wav->tags = NULL; + } + +} + +#define MAX_BUFFER_SIZE 4096 + +static GstFlowReturn +gst_wavparse_stream_data (GstWavParse * wav) +{ + GstBuffer *buf = NULL; + GstFlowReturn res = GST_FLOW_OK; + guint64 desired, obtained; + GstClockTime timestamp, next_timestamp; + guint64 pos, nextpos; + + +iterate_adapter: + GST_LOG_OBJECT (wav, + "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %" + G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft); + + /* Get the next n bytes and output them */ + if (wav->dataleft == 0 || wav->dataleft < wav->blockalign) + goto found_eos; + + /* scale the amount of data by the segment rate so we get equal + * amounts of data regardless of the playback rate */ + desired = + MIN (gst_guint64_to_gdouble (wav->dataleft), + MAX_BUFFER_SIZE * ABS (wav->segment.rate)); + if (desired >= wav->blockalign && wav->blockalign > 0) + desired -= (desired % wav->blockalign); + + + GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data " + "from the sinkpad", desired); + + if (wav->streaming) { + guint avail = gst_adapter_available (wav->adapter); + + if (avail < desired) { + + GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail); + return GST_FLOW_OK; + } + + buf = gst_buffer_new (); + GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired); + GST_BUFFER_SIZE (buf) = desired; + + + } else { + if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, + desired, &buf)) != GST_FLOW_OK) + { + + goto pull_error; + } + + } + + /* first chunk of data? create the source pad. We do this only here so + * we can detect broken .wav files with dts disguised as raw PCM (sigh) */ + if (G_UNLIKELY (wav->first)) { + wav->first = FALSE; + + gst_wavparse_add_src_pad (wav, buf); + } + + obtained = GST_BUFFER_SIZE (buf); + + /* our positions */ + pos = wav->offset - wav->datastart; + nextpos = pos + obtained; + + /* update offsets, does not overflow. */ + GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample; + GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample; + + /* and timestamps, be carefull for overflows */ + timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps); + next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps); + + GST_BUFFER_TIMESTAMP (buf) = timestamp; + GST_BUFFER_DURATION (buf) = next_timestamp - timestamp; + + /* update current running segment position */ + gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp); + + /* don't forget to set the caps on the buffer */ + gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad)); + + GST_LOG_OBJECT (wav, + "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT + ", size:%u", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), + GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf)); + + + if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK) + { + + goto push_error; + } + + + if (obtained < wav->dataleft) { + wav->dataleft -= obtained; + } else { + wav->dataleft = 0; + } + wav->offset += obtained; + /* Iterate until need more data, so adapter size won't grow */ + if (wav->streaming) { + GST_LOG_OBJECT (wav, + "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset, + wav->end_offset); + + goto iterate_adapter; + } + + return res; + + /* ERROR */ +found_eos: + { + + GST_DEBUG_OBJECT (wav, "found EOS"); + /* we completed the segment */ + wav->segment_running = FALSE; + if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) { + GstClockTime stop; + + if ((stop = wav->segment.stop) == -1) + stop = wav->segment.duration; + + gst_element_post_message (GST_ELEMENT (wav), + gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME, + stop)); + + } else { + gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); + + } + return GST_FLOW_WRONG_STATE; + } +pull_error: + { + + GST_DEBUG_OBJECT (wav, "Error getting %" G_GINT64_FORMAT " bytes from the " + "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft); + return res; + } +push_error: + { + + GST_DEBUG_OBJECT (wav, "Error pushing on srcpad"); + return res; + } +} + +static void +gst_wavparse_loop (GstPad * pad) +{ + GstFlowReturn ret; + GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + + GST_LOG_OBJECT (wav, "process data"); + + switch (wav->state) { + case GST_WAVPARSE_START: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START"); + if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK) + goto pause; + + wav->state = GST_WAVPARSE_HEADER; + /* fall-through */ + + case GST_WAVPARSE_HEADER: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER"); + if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) + goto pause; + + wav->state = GST_WAVPARSE_DATA; + /* fall-through */ + + case GST_WAVPARSE_DATA: + if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) + { + + goto pause; + } + + break; + default: + g_assert_not_reached (); + } + + return; + + /* ERRORS */ +pause: + GST_LOG_OBJECT (wav, "pausing task %d", ret); + gst_pad_pause_task (wav->sinkpad); + if (GST_FLOW_IS_FATAL (ret)) { + + + /* for fatal errors we post an error message */ + GST_ELEMENT_ERROR (wav, STREAM, FAILED, + (_("Internal data stream error.")), + ("streaming stopped, reason %s", gst_flow_get_name (ret))); + if (wav->srcpad != NULL) + { + gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); + } + } + +} + +static GstFlowReturn +gst_wavparse_chain (GstPad * pad, GstBuffer * buf) +{ + GstFlowReturn ret; + GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + GST_LOG_OBJECT (wav, "adapter_push %" 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; + /* fall-through */ + + case GST_WAVPARSE_DATA: + if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) + { + + goto pause; + } + + break; + + default: + g_assert_not_reached (); + } + + return ret; + +pause: + + GST_LOG_OBJECT (wav, "pausing task %d", ret); + gst_pad_pause_task (wav->sinkpad); + if (GST_FLOW_IS_FATAL (ret)) { + /* for fatal errors we post an error message */ + + GST_ELEMENT_ERROR (wav, STREAM, FAILED, + (_("Internal data stream error.")), + ("streaming stopped, reason %s", gst_flow_get_name (ret))); + if (wav->srcpad != NULL) + { + gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); + + } + } + + return ret; +} + +#if 0 +/* convert and query stuff */ +static const GstFormat * +gst_wavparse_get_formats (GstPad * pad) +{ + static GstFormat formats[] = { + GST_FORMAT_TIME, + GST_FORMAT_BYTES, + GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */ + 0 + }; + + return formats; +} +#endif + +static gboolean +gst_wavparse_pad_convert (GstPad * pad, + GstFormat src_format, gint64 src_value, + GstFormat * dest_format, gint64 * dest_value) +{ + GstWavParse *wavparse; + gboolean res = TRUE; + + wavparse = GST_WAVPARSE (gst_pad_get_parent (pad)); + + if (wavparse->bytes_per_sample == 0) + goto no_bytes_per_sample; + + if (wavparse->bps == 0) + goto no_bps; + + switch (src_format) { + case GST_FORMAT_BYTES: + switch (*dest_format) { + case GST_FORMAT_DEFAULT: + *dest_value = src_value / wavparse->bytes_per_sample; + break; + case GST_FORMAT_TIME: + *dest_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->bps); + break; + default: + res = FALSE; + goto done; + } + *dest_value -= *dest_value % wavparse->bytes_per_sample; + break; + + case GST_FORMAT_DEFAULT: + switch (*dest_format) { + case GST_FORMAT_BYTES: + *dest_value = src_value * wavparse->bytes_per_sample; + break; + case GST_FORMAT_TIME: + *dest_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, wavparse->rate); + break; + default: + res = FALSE; + goto done; + } + break; + + case GST_FORMAT_TIME: + switch (*dest_format) { + case GST_FORMAT_BYTES: + /* make sure we end up on a sample boundary */ + *dest_value = + gst_util_uint64_scale_int (src_value, wavparse->bps, GST_SECOND); + *dest_value -= *dest_value % wavparse->blockalign; + break; + case GST_FORMAT_DEFAULT: + *dest_value = + gst_util_uint64_scale_int (src_value, wavparse->rate, GST_SECOND); + break; + default: + res = FALSE; + goto done; + } + break; + + default: + res = FALSE; + goto done; + } + +done: + gst_object_unref (wavparse); + + return res; + + /* ERRORS */ +no_bytes_per_sample: + { + GST_DEBUG_OBJECT (wavparse, + "bytes_per_sample 0, probably an mp3 - channels %d, width %d", + wavparse->channels, wavparse->width); + res = FALSE; + goto done; + } +no_bps: + { + GST_DEBUG_OBJECT (wavparse, "bps 0, cannot convert"); + res = FALSE; + goto done; + } +} + +static const GstQueryType * +gst_wavparse_get_query_types (GstPad * pad) +{ + static const GstQueryType types[] = { + GST_QUERY_POSITION, + GST_QUERY_DURATION, + GST_QUERY_CONVERT, + 0 + }; + + return types; +} + +/* handle queries for location and length in requested format */ +static gboolean +gst_wavparse_pad_query (GstPad * pad, GstQuery * query) +{ + gboolean res = TRUE; + GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + + /* only if we know */ + if (wav->state != GST_WAVPARSE_DATA) + return FALSE; + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_POSITION: + { + gint64 curb; + gint64 cur; + GstFormat format; + gboolean res = TRUE; + + curb = wav->offset - wav->datastart; + gst_query_parse_position (query, &format, NULL); + + switch (format) { + case GST_FORMAT_TIME: + res &= + gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb, + &format, &cur); + break; + default: + format = GST_FORMAT_BYTES; + cur = curb; + break; + } + if (res) + gst_query_set_position (query, format, cur); + break; + } + case GST_QUERY_DURATION: + { + gint64 endb; + gint64 end; + GstFormat format; + gboolean res = TRUE; + + endb = wav->datasize; + gst_query_parse_duration (query, &format, NULL); + + switch (format) { + case GST_FORMAT_TIME: + res &= + gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb, + &format, &end); + break; + default: + format = GST_FORMAT_BYTES; + end = endb; + break; + } + if (res) + gst_query_set_duration (query, format, end); + break; + } + case GST_QUERY_CONVERT: + { + gint64 srcvalue, dstvalue; + GstFormat srcformat, dstformat; + + gst_query_parse_convert (query, &srcformat, &srcvalue, + &dstformat, &dstvalue); + res &= + gst_wavparse_pad_convert (pad, srcformat, srcvalue, + &dstformat, &dstvalue); + if (res) + gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue); + break; + } + default: + res = gst_pad_query_default (pad, query); + break; + } + return res; +} + +static gboolean +gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) +{ + GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad)); + gboolean res = TRUE; + + GST_DEBUG_OBJECT (wavparse, "event %d, %s", GST_EVENT_TYPE (event), + GST_EVENT_TYPE_NAME (event)); + + /* can only handle events when we are in the data state */ + if (wavparse->state != GST_WAVPARSE_DATA) + return FALSE; + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_SEEK: + { + res = gst_wavparse_perform_seek (wavparse, event); + break; + } + default: + res = FALSE; + break; + } + + gst_event_unref (event); + + return res; +} + +static gboolean +gst_wavparse_sink_activate (GstPad * sinkpad) +{ + GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad)); + gboolean res; + + if (gst_pad_check_pull_range (sinkpad)) { + GST_DEBUG ("going to pull mode"); + wav->streaming = FALSE; + wav->adapter = NULL; + res = gst_pad_activate_pull (sinkpad, TRUE); + } else { + GST_DEBUG ("going to push (streaming) mode"); + wav->streaming = TRUE; + wav->adapter = gst_adapter_new (); + res = gst_pad_activate_push (sinkpad, TRUE); + } + gst_object_unref (wav); + return res; +} + + +static gboolean +gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active) +{ + GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad)); + + GST_DEBUG_OBJECT (wav, "activating pull"); + + if (active) { + /* if we have a scheduler we can start the task */ + wav->segment_running = TRUE; + gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad); + } else { + gst_pad_stop_task (sinkpad); + } + gst_object_unref (wav); + + return TRUE; +} + +static GstStateChangeReturn +gst_wavparse_change_state (GstElement * element, GstStateChange transition) +{ + GstStateChangeReturn ret; + GstWavParse *wav = GST_WAVPARSE (element); + + GST_DEBUG_OBJECT (wav, "changing state %s - %s", + gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)), + gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition))); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + gst_wavparse_reset (wav); + break; + case GST_STATE_CHANGE_PAUSED_TO_PLAYING: + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PLAYING_TO_PAUSED: + break; + case GST_STATE_CHANGE_PAUSED_TO_READY:{ + GstEvent **event_p = &wav->seek_event; + + gst_wavparse_destroy_sourcepad (wav); + gst_event_replace (event_p, NULL); + gst_wavparse_reset (wav); + if (wav->adapter) { + gst_adapter_clear (wav->adapter); + } + break; + } + case GST_STATE_CHANGE_READY_TO_NULL: + break; + default: + break; + } + return ret; +} + +static gboolean +plugin_init (GstPlugin * plugin) +{ + gst_riff_init (); + + GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser"); + + return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY, + GST_TYPE_WAVPARSE); +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "wavparse", + "Parse a .wav file into raw audio", + plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) + + +EXPORT_C GstPluginDesc* _GST_PLUGIN_DESC() +{ + return &gst_plugin_desc; +} +