/* -*- 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. * * * Wavparse supports both push and pull mode operations, making it possible to * stream from a network source. * * 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 a network url. * * * * 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 #include "gstwavparse.h" #include "gst/riff/riff-ids.h" #include "gst/riff/riff-media.h" #include 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 "); static GstStaticPadTemplate sink_template_factory = GST_STATIC_PAD_TEMPLATE ("wavparse_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_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/ms-gsm; " "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 ]") ); #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); /* 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->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->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) { /* 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"); } /* 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_int, but performs ceiling division. */ static guint64 uint64_ceiling_scale_int (guint64 val, gint num, gint 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, }; if (event) { GstFormat fmt; 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; fmt = wav->segment.format; /* we have to have a format as the segment format. Try to convert * if not. */ if (format != wav->segment.format) { res = TRUE; if (cur_type != GST_SEEK_TYPE_NONE) res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur); if (res && stop_type != GST_SEEK_TYPE_NONE) res = gst_pad_query_convert (wav->srcpad, format, stop, &fmt, &stop); if (!res) goto no_format; format = fmt; } } else { GST_DEBUG_OBJECT (wav, "doing seek without event"); flags = 0; cur_type = GST_SEEK_TYPE_SET; stop_type = GST_SEEK_TYPE_SET; } /* 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); /* 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) wav->offset = uint64_ceiling_scale_int (seeksegment.last_stop, wav->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) wav->end_offset = uint64_ceiling_scale_int (stop, wav->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: 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) { /* 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); gst_pad_push_event (wav->srcpad, gst_event_new_new_segment (TRUE, wav->segment.rate, wav->segment.format, wav->segment.accum, wav->segment.last_stop, wav->segment.accum)); /* 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->newsegment) gst_event_unref (wav->newsegment); wav->newsegment = gst_event_new_new_segment (FALSE, wav->segment.rate, wav->segment.format, wav->segment.last_stop, stop, wav->segment.last_stop); /* 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; } } static gboolean gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len) { gboolean res = FALSE; GstFormat fmt = GST_FORMAT_BYTES; GstPad *peer; if ((peer = gst_pad_get_peer (wav->sinkpad))) { res = gst_pad_query_duration (peer, &fmt, len); gst_object_unref (peer); } return res; } static GstFlowReturn gst_wavparse_stream_headers (GstWavParse * wav) { GstFlowReturn res; GstBuffer *buf; gst_riff_strf_auds *header = NULL; guint32 tag, size; gboolean gotdata = FALSE; GstCaps *caps; gint64 duration; gchar *codec_name = NULL; GstEvent **event_p; while (!wav->got_fmt) { GstBuffer *extra; /* The header starts with a 'fmt ' tag */ if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) return GST_FLOW_OK; 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) { GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk", GST_FOURCC_ARGS (tag)); if (wav->streaming) { gst_adapter_flush (wav->adapter, size); wav->offset += size; } gst_buffer_unref (buf); buf = NULL; continue; } if (tag != GST_RIFF_TAG_fmt) goto invalid_wav; if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header, &extra))) goto parse_header_error; if (wav->streaming) { gst_adapter_flush (wav->adapter, size); wav->offset += size; gst_buffer_unref (buf); buf = NULL; } /* Note: gst_riff_create_audio_caps might need to fix values in * the header header depending on the format, so call it first */ caps = gst_riff_create_audio_caps (header->format, NULL, header, extra, NULL, &codec_name); if (extra) gst_buffer_unref (extra); wav->format = header->format; wav->rate = header->rate; wav->channels = header->channels; wav->blockalign = header->blockalign; wav->depth = header->size; wav->av_bps = header->av_bps; g_free (header); if (wav->channels == 0) goto no_channels; /* 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; } default: /* 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; 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, "av_bps = %u", (guint) wav->av_bps); GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate); GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels); GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample); /* 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; } } /* loop headers until we get data */ while (!gotdata) { gint64 upstream_size = 0; if (wav->streaming) { if (!gst_wavparse_peek_chunk_info (wav, &tag, &size)) return GST_FLOW_OK; } else { if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, 8, &buf)) != GST_FLOW_OK) goto header_read_error; tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf)); size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4); } gst_wavparse_get_upstream_size (wav, &upstream_size); /* wav is a st00pid format, we don't know for sure where data starts. * So we have to go bit by bit until we find the 'data' header */ switch (tag) { /* TODO : Implement the various cases */ case GST_RIFF_TAG_data:{ GstFormat fmt; GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size); if (wav->streaming) { gst_adapter_flush (wav->adapter, 8); gotdata = TRUE; } else { gst_buffer_unref (buf); } wav->offset += 8; wav->datastart = wav->offset; /* file might be truncated */ fmt = GST_FORMAT_BYTES; if (upstream_size) { size = MIN (size, (upstream_size - wav->datastart)); } wav->datasize = (guint64) size; wav->dataleft = (guint64) size; wav->end_offset = size + wav->datastart; if (!wav->streaming) { /* We will continue parsing tags 'till end */ wav->offset += size; } GST_DEBUG_OBJECT (wav, "datasize = %d", size); break; } case GST_RIFF_TAG_fact:{ /* number of samples (for compressed formats) */ if (wav->streaming) { const guint8 *data = NULL; if (gst_adapter_available (wav->adapter) < 8 + 4) { return GST_FLOW_OK; } gst_adapter_flush (wav->adapter, 8); data = gst_adapter_peek (wav->adapter, 4); wav->fact = GST_READ_UINT32_LE (data); gst_adapter_flush (wav->adapter, 4); } else { gst_buffer_unref (buf); if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset + 8, 4, &buf)) != GST_FLOW_OK) goto header_read_error; wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf)); gst_buffer_unref (buf); } wav->offset += 8 + 4; break; } default: if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) return GST_FLOW_OK; } GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)); wav->offset += 8 + ((size + 1) & ~1); if (wav->streaming) { gst_adapter_flush (wav->adapter, 8 + ((size + 1) & ~1)); } else { gst_buffer_unref (buf); } } if (upstream_size && (wav->offset >= upstream_size)) { /* Now we are gone through the whole file */ gotdata = TRUE; } } GST_DEBUG_OBJECT (wav, "Finished parsing headers"); if (wav->bps <= 0 && wav->fact) { wav->bps = (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize, (guint64) wav->fact); GST_DEBUG_OBJECT (wav, "calculated bps : %d", wav->bps); } if (wav->bps > 0) { duration = uint64_ceiling_scale_int (wav->datasize, GST_SECOND, wav->bps); GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT, GST_TIME_ARGS (duration)); gst_segment_init (&wav->segment, GST_FORMAT_TIME); gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, 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); 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")); 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_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; /* _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 }; 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)); GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad"); if (wav->newsegment) { gst_pad_push_event (wav->srcpad, wav->newsegment); wav->newsegment = 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; gst_wavparse_add_src_pad (wav, buf); } /* If we have a pending newsegment send it now. */ if (G_UNLIKELY (wav->newsegment != NULL)) { gst_pad_push_event (wav->srcpad, wav->newsegment); wav->newsegment = NULL; } obtained = GST_BUFFER_SIZE (buf); /* our positions 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 carefull for overflows */ timestamp = uint64_ceiling_scale_int (pos, GST_SECOND, wav->bps); next_timestamp = uint64_ceiling_scale_int (nextpos, GST_SECOND, wav->bps); duration = next_timestamp - timestamp; /* update current running segment position */ gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp); } else { /* no bitrate, don't timestamp */ timestamp = GST_CLOCK_TIME_NONE; next_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); } 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->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"); 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_WARNING_OBJECT (wav, "Error pushing on srcpad %p, is linked? = %d", wav->srcpad, gst_pad_is_linked (wav->srcpad)); return res; } } static void gst_wavparse_loop (GstPad * pad) { GstFlowReturn ret; GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); GST_LOG_OBJECT (wav, "process data"); switch (wav->state) { case GST_WAVPARSE_START: GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK) goto pause; wav->state = GST_WAVPARSE_HEADER; /* fall-through */ case GST_WAVPARSE_HEADER: GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER"); if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) goto pause; wav->state = GST_WAVPARSE_DATA; /* fall-through */ case GST_WAVPARSE_DATA: if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) goto pause; break; default: g_assert_not_reached (); } return; /* ERRORS */ pause: { 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_DEBUG_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_DEBUG_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; /* 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) 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; /* make sure we end up on a sample boundary */ *dest_value -= *dest_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; } 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_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; 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; endb = wav->datasize; gst_query_parse_duration (query, &format, NULL); switch (format) { case GST_FORMAT_TIME:{ if (wav->fact) { end = GST_SECOND * wav->fact / wav->rate; } else { res &= gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb, &format, &end); } break; } default: format = GST_FORMAT_BYTES; end = endb; break; } if (res) gst_query_set_duration (query, format, end); break; } case GST_QUERY_CONVERT: { gint64 srcvalue, dstvalue; GstFormat srcformat, dstformat; gst_query_parse_convert (query, &srcformat, &srcvalue, &dstformat, &dstvalue); res &= gst_wavparse_pad_convert (pad, srcformat, srcvalue, &dstformat, &dstvalue); if (res) gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue); break; } default: res = gst_pad_query_default (pad, query); break; } return res; } static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) { GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad)); gboolean res = TRUE; GST_DEBUG_OBJECT (wavparse, "event %d, %s", GST_EVENT_TYPE (event), GST_EVENT_TYPE_NAME (event)); /* can only handle events when we are in the data state */ if (wavparse->state != GST_WAVPARSE_DATA) return FALSE; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: { res = gst_wavparse_perform_seek (wavparse, event); 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)