/* -*- 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, 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-1.0 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-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink * ]| Stream data from a network url. * */ /* * TODO: * http://replaygain.hydrogenaudio.org/file_format_wav.html */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstwavparse.h" #include "gst/riff/riff-media.h" #include #include #include #ifdef GSTREAMER_LITE #include #endif // GSTREAMER_LITE GST_DEBUG_CATEGORY_STATIC (wavparse_debug); #define GST_CAT_DEFAULT (wavparse_debug) /* Data size chunk of RF64, * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */ #define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4') static void gst_wavparse_dispose (GObject * object); static gboolean gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent); static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent, GstPadMode mode, gboolean active); static gboolean gst_wavparse_send_event (GstElement * element, GstEvent * event); static GstStateChangeReturn gst_wavparse_change_state (GstElement * element, GstStateChange transition); #ifdef GSTREAMER_LITE static gboolean gst_wavparse_sink_query (GstPad * pad, GstObject *parent, GstQuery * query); #endif // GSTREAMER_LITE static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent, 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, GstObject * parent, GstBuffer * buf); static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static void gst_wavparse_loop (GstPad * pad); static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event); static void gst_wavparse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wavparse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); #define DEFAULT_IGNORE_LENGTH FALSE enum { PROP_0, PROP_IGNORE_LENGTH, }; static GstStaticPadTemplate sink_template_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wav") ); #define DEBUG_INIT \ GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser"); #define gst_wavparse_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT, DEBUG_INIT); typedef struct { /* Offset Size Description Value * 0x00 4 ID unique identification value * 0x04 4 Position play order position * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk * 0x0c 4 Chunk Start Byte Offset of Data Chunk * * 0x10 4 Block Start Byte Offset to sample of First Channel * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel */ guint32 id; guint32 position; guint32 data_chunk_id; guint32 chunk_start; guint32 block_start; guint32 sample_offset; } GstWavParseCue; typedef struct { /* Offset Size Description Value * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF * 0x0c Text */ guint32 cue_point_id; gchar *text; } GstWavParseLabl, GstWavParseNote; static void gst_wavparse_class_init (GstWavParseClass * klass) { GstElementClass *gstelement_class; GObjectClass *object_class; GstPadTemplate *src_template; gstelement_class = (GstElementClass *) klass; object_class = (GObjectClass *) klass; parent_class = g_type_class_peek_parent (klass); object_class->dispose = gst_wavparse_dispose; object_class->set_property = gst_wavparse_set_property; object_class->get_property = gst_wavparse_get_property; /** * GstWavParse:ignore-length: * * This selects whether the length found in a data chunk * should be ignored. This may be useful for streamed audio * where the length is unknown until the end of streaming, * and various software/hardware just puts some random value * in there and hopes it doesn't break too much. */ g_object_class_install_property (object_class, PROP_IGNORE_LENGTH, g_param_spec_boolean ("ignore-length", "Ignore length", "Ignore length from the Wave header", DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); gstelement_class->change_state = gst_wavparse_change_state; gstelement_class->send_event = gst_wavparse_send_event; /* register pads */ gst_element_class_add_static_pad_template (gstelement_class, &sink_template_factory); src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ()); gst_element_class_add_pad_template (gstelement_class, src_template); gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer", "Codec/Demuxer/Audio", "Parse a .wav file into raw audio", "Erik Walthinsen "); } static void gst_wavparse_notes_free (GstWavParseNote * note) { if (note) g_free (note->text); g_free (note); } static void gst_wavparse_labls_free (GstWavParseLabl * labl) { if (labl) g_free (labl->text); g_free (labl); } static void gst_wavparse_reset (GstWavParse * wav) { wav->state = GST_WAVPARSE_START; /* These will all be set correctly in the fmt chunk */ wav->depth = 0; wav->rate = 0; wav->width = 0; wav->channels = 0; wav->blockalign = 0; wav->bps = 0; wav->fact = 0; wav->offset = 0; wav->end_offset = 0; wav->dataleft = 0; wav->datasize = 0; wav->datastart = 0; wav->chunk_size = 0; wav->duration = 0; wav->got_fmt = FALSE; wav->first = TRUE; if (wav->seek_event) gst_event_unref (wav->seek_event); wav->seek_event = NULL; if (wav->adapter) { gst_adapter_clear (wav->adapter); g_object_unref (wav->adapter); wav->adapter = NULL; } if (wav->tags) gst_tag_list_unref (wav->tags); wav->tags = NULL; if (wav->toc) gst_toc_unref (wav->toc); wav->toc = NULL; if (wav->cues) g_list_free_full (wav->cues, g_free); wav->cues = NULL; if (wav->labls) g_list_free_full (wav->labls, (GDestroyNotify) gst_wavparse_labls_free); wav->labls = NULL; if (wav->notes) g_list_free_full (wav->notes, (GDestroyNotify) gst_wavparse_notes_free); wav->notes = NULL; if (wav->caps) gst_caps_unref (wav->caps); wav->caps = NULL; if (wav->start_segment) gst_event_unref (wav->start_segment); wav->start_segment = NULL; } static void gst_wavparse_dispose (GObject * object) { GstWavParse *wav = GST_WAVPARSE (object); GST_DEBUG_OBJECT (wav, "WAV: Dispose"); gst_wavparse_reset (wav); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_wavparse_init (GstWavParse * wavparse) { #ifdef GSTREAMER_LITE GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavparse); GstPadTemplate *src_template; #endif // GSTREAMER_LITE 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_activatemode_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode)); gst_pad_set_chain_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_chain)); gst_pad_set_event_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_sink_event)); #ifdef GSTREAMER_LITE gst_pad_set_query_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_sink_query)); #endif // GSTREAMER_LITE gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad); /* src */ wavparse->srcpad = gst_pad_new_from_template (gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src"); gst_pad_use_fixed_caps (wavparse->srcpad); 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_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad); } 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: 0x%" G_GINT32_MODIFIER "x", 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; } static gboolean gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos) { /* -1 always maps to -1 */ if (ts == -1) { *bytepos = -1; return TRUE; } /* 0 always maps to 0 */ if (ts == 0) { *bytepos = 0; return TRUE; } if (wav->bps > 0) { *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND); return TRUE; } else if (wav->fact) { guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact); *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND); return TRUE; } return FALSE; } /* This function is used to perform seeks on the element. * * It also works when event is NULL, in which case it will just * start from the last configured segment. This technique is * used when activating the element and to perform the seek in * READY. */ static gboolean gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) { gboolean res; gdouble rate; GstFormat format, bformat; GstSeekFlags flags; GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type; gint64 cur, stop, upstream_size; gboolean flush; gboolean update; GstSegment seeksegment = { 0, }; gint64 last_stop; guint32 seqnum = 0; if (event) { GST_DEBUG_OBJECT (wav, "doing seek with event"); gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); seqnum = gst_event_get_seqnum (event); /* no negative rates yet */ if (rate < 0.0) goto negative_rate; if (format != wav->segment.format) { GST_INFO_OBJECT (wav, "converting seek-event from %s to %s", gst_format_get_name (format), gst_format_get_name (wav->segment.format)); res = TRUE; if (cur_type != GST_SEEK_TYPE_NONE) res = gst_pad_query_convert (wav->srcpad, format, cur, wav->segment.format, &cur); if (res && stop_type != GST_SEEK_TYPE_NONE) res = gst_pad_query_convert (wav->srcpad, format, stop, wav->segment.format, &stop); if (!res) goto no_format; format = wav->segment.format; } } else { GST_DEBUG_OBJECT (wav, "doing seek without event"); flags = 0; rate = 1.0; cur_type = GST_SEEK_TYPE_SET; stop_type = GST_SEEK_TYPE_SET; } /* in push mode, we must delegate to upstream */ if (wav->streaming) { gboolean res = FALSE; /* if streaming not yet started; only prepare initial newsegment */ if (!event || wav->state != GST_WAVPARSE_DATA) { if (wav->start_segment) gst_event_unref (wav->start_segment); wav->start_segment = gst_event_new_segment (&wav->segment); res = TRUE; } else { /* convert seek positions to byte positions in data sections */ if (format == GST_FORMAT_TIME) { /* should not fail */ if (!gst_wavparse_time_to_bytepos (wav, cur, &cur)) goto no_position; if (!gst_wavparse_time_to_bytepos (wav, stop, &stop)) goto no_position; } /* mind sample boundary and header */ if (cur >= 0) { cur -= (cur % wav->bytes_per_sample); cur += wav->datastart; } if (stop >= 0) { stop -= (stop % wav->bytes_per_sample); stop += wav->datastart; } GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, " "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur, stop); /* BYTE seek event */ event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur, stop_type, stop); gst_event_set_seqnum (event, seqnum); res = gst_pad_push_event (wav->sinkpad, event); } return res; } /* 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) { GstEvent *fevent; GST_DEBUG_OBJECT (wav, "sending flush start"); fevent = gst_event_new_flush_start (); gst_event_set_seqnum (fevent, seqnum); gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent)); gst_pad_push_event (wav->srcpad, fevent); } else { gst_pad_pause_task (wav->sinkpad); } /* we should now be able to grab the streaming thread because we stopped it * with the above flush/pause code */ GST_PAD_STREAM_LOCK (wav->sinkpad); /* save current position */ last_stop = wav->segment.position; GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop); /* copy segment, we need this because we still need the old * segment when we close the current segment. */ memcpy (&seeksegment, &wav->segment, sizeof (GstSegment)); /* configure the seek parameters in the seeksegment. We will then have the * right values in the segment to perform the seek */ if (event) { GST_DEBUG_OBJECT (wav, "configuring seek"); gst_segment_do_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 (!gst_wavparse_time_to_bytepos (wav, seeksegment.position, (gint64 *) & wav->offset)) wav->offset = seeksegment.position; 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 (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset)) 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_peer_query_duration (wav->sinkpad, bformat, &upstream_size)) wav->end_offset = MIN (wav->end_offset, upstream_size); if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize) wav->end_offset = wav->datastart + wav->datasize; /* this is the range of bytes we will use for playback */ wav->offset = MIN (wav->offset, wav->end_offset); wav->dataleft = wav->end_offset - wav->offset; GST_DEBUG_OBJECT (wav, "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset, wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop)); /* prepare for streaming again */ if (flush) { GstEvent *fevent; /* if we sent a FLUSH_START, we now send a FLUSH_STOP */ GST_DEBUG_OBJECT (wav, "sending flush stop"); fevent = gst_event_new_flush_stop (TRUE); gst_event_set_seqnum (fevent, seqnum); gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent)); gst_pad_push_event (wav->srcpad, fevent); } /* 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.position)); } /* now create the newsegment */ GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT, wav->segment.position, stop); /* store the newsegment event so it can be sent from the streaming thread. */ if (wav->start_segment) gst_event_unref (wav->start_segment); wav->start_segment = gst_event_new_segment (&wav->segment); gst_event_set_seqnum (wav->start_segment, seqnum); /* mark discont if we are going to stream from another position. */ if (last_stop != wav->segment.position) { GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position"); wav->discont = TRUE; } /* and start the streaming task again */ if (!wav->streaming) { gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop, wav->sinkpad, NULL); } 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; } no_position: { GST_DEBUG_OBJECT (wav, "Could not determine byte position for desired time"); 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 the chunk info (header) is available */ 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_map (wav->adapter, 8); *tag = GST_READ_UINT32_LE (data); *size = GST_READ_UINT32_LE (data + 4); gst_adapter_unmap (wav->adapter); GST_DEBUG ("Next chunk size is %u 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 the full chunk is available */ 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; /* size 0 -> empty data buffer would surprise most callers, * large size -> do not bother trying to squeeze that into adapter, * so we throw poor man's exception, which can be caught if caller really * wants to handle 0 size chunk */ if (!(*size) || (*size) >= (1 << 30)) { GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT, *size, GST_FOURCC_ARGS (*tag)); /* chain should give up */ wav->abort_buffering = TRUE; return FALSE; } peek_size = (*size + 1) & ~1; available = gst_adapter_available (wav->adapter); if (available >= (8 + peek_size)) { return TRUE; } else { GST_LOG ("but only %u bytes available now", available); return FALSE; } } /* * gst_wavparse_calculate_duration: * @wav: wavparse object * * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a * fallback. * * Returns: %TRUE if duration is available. */ static gboolean gst_wavparse_calculate_duration (GstWavParse * wav) { if (wav->duration > 0) return TRUE; if (wav->bps > 0) { GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize); wav->duration = gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND, (guint64) wav->bps); GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT, GST_TIME_ARGS (wav->duration)); return TRUE; } else if (wav->fact) { wav->duration = gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate); GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT, GST_TIME_ARGS (wav->duration)); return TRUE; } return FALSE; } static gboolean gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag, guint32 size) { guint flush; if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) return FALSE; } GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)); flush = 8 + ((size + 1) & ~1); wav->offset += flush; if (wav->streaming) { gst_adapter_flush (wav->adapter, flush); } else { gst_buffer_unref (buf); } return TRUE; } /* * gst_wavparse_cue_chunk: * @wav GstWavParse object * @data holder for data * @size holder for data size * * Parse cue chunk from @data to wav->cues. * * Returns: %TRUE when cue chunk is available */ static gboolean gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size) { guint32 i, ncues; GList *cues = NULL; GstWavParseCue *cue; if (wav->cues) { GST_WARNING_OBJECT (wav, "found another cue's"); return TRUE; } ncues = GST_READ_UINT32_LE (data); if (size < 4 + ncues * 24) { GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues); return FALSE; } /* parse data */ data += 4; for (i = 0; i < ncues; i++) { cue = g_new0 (GstWavParseCue, 1); cue->id = GST_READ_UINT32_LE (data); cue->position = GST_READ_UINT32_LE (data + 4); cue->data_chunk_id = GST_READ_UINT32_LE (data + 8); cue->chunk_start = GST_READ_UINT32_LE (data + 12); cue->block_start = GST_READ_UINT32_LE (data + 16); cue->sample_offset = GST_READ_UINT32_LE (data + 20); cues = g_list_append (cues, cue); data += 24; } wav->cues = cues; return TRUE; } /* * gst_wavparse_labl_chunk: * @wav GstWavParse object * @data holder for data * @size holder for data size * * Parse labl from @data to wav->labls. * * Returns: %TRUE when labl chunk is available */ static gboolean gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size) { GstWavParseLabl *labl; if (size < 5) return FALSE; labl = g_new0 (GstWavParseLabl, 1); /* parse data */ data += 8; labl->cue_point_id = GST_READ_UINT32_LE (data); labl->text = g_memdup (data + 4, size - 4); wav->labls = g_list_append (wav->labls, labl); return TRUE; } /* * gst_wavparse_note_chunk: * @wav GstWavParse object * @data holder for data * @size holder for data size * * Parse note from @data to wav->notes. * * Returns: %TRUE when note chunk is available */ static gboolean gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size) { GstWavParseNote *note; if (size < 5) return FALSE; note = g_new0 (GstWavParseNote, 1); /* parse data */ data += 8; note->cue_point_id = GST_READ_UINT32_LE (data); note->text = g_memdup (data + 4, size - 4); wav->notes = g_list_append (wav->notes, note); return TRUE; } /* * gst_wavparse_smpl_chunk: * @wav GstWavParse object * @data holder for data * @size holder for data size * * Parse smpl chunk from @data. * * Returns: %TRUE when cue chunk is available */ static gboolean gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size) { guint32 note_number; /* manufacturer_id = GST_READ_UINT32_LE (data); product_id = GST_READ_UINT32_LE (data + 4); sample_period = GST_READ_UINT32_LE (data + 8); */ note_number = GST_READ_UINT32_LE (data + 12); /* pitch_fraction = GST_READ_UINT32_LE (data + 16); SMPTE_format = GST_READ_UINT32_LE (data + 20); SMPTE_offset = GST_READ_UINT32_LE (data + 24); num_sample_loops = GST_READ_UINT32_LE (data + 28); List of Sample Loops, 24 bytes each */ if (!wav->tags) wav->tags = gst_tag_list_new_empty (); gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL); return TRUE; } /* * gst_wavparse_adtl_chunk: * @wav GstWavParse object * @data holder for data * @size holder for data size * * Parse adtl from @data. * * Returns: %TRUE when adtl chunk is available */ static gboolean gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size) { guint32 ltag, lsize, offset = 0; while (size >= 8) { ltag = GST_READ_UINT32_LE (data + offset); lsize = GST_READ_UINT32_LE (data + offset + 4); if (lsize + 8 > size) { GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size); return FALSE; } switch (ltag) { case GST_RIFF_TAG_labl: gst_wavparse_labl_chunk (wav, data + offset, size); break; case GST_RIFF_TAG_note: gst_wavparse_note_chunk (wav, data + offset, size); break; default: GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (ltag)); GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize); break; } offset += 8 + GST_ROUND_UP_2 (lsize); size -= 8 + GST_ROUND_UP_2 (lsize); } return TRUE; } static GstTagList * gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id) { GstTagList *tags = NULL; GstTocEntry *entry = NULL; entry = gst_toc_find_entry (toc, id); if (entry != NULL) { tags = gst_toc_entry_get_tags (entry); if (tags == NULL) { tags = gst_tag_list_new_empty (); gst_toc_entry_set_tags (entry, tags); } } return tags; } /* * gst_wavparse_create_toc: * @wav GstWavParse object * * Create TOC from wav->cues and wav->labls. */ static gboolean gst_wavparse_create_toc (GstWavParse * wav) { gint64 start, stop; gchar *id; GList *list; GstWavParseCue *cue; GstWavParseLabl *labl; GstWavParseNote *note; GstTagList *tags; GstToc *toc; GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL; GST_OBJECT_LOCK (wav); if (wav->toc) { GST_OBJECT_UNLOCK (wav); GST_WARNING_OBJECT (wav, "found another TOC"); return FALSE; } if (!wav->cues) { GST_OBJECT_UNLOCK (wav); return TRUE; } /* FIXME: send CURRENT scope toc too */ toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL); /* add cue edition */ entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue"); gst_toc_entry_set_start_stop_times (entry, 0, wav->duration); gst_toc_append_entry (toc, entry); /* add tracks in cue edition */ list = wav->cues; while (list) { cue = list->data; prev_subentry = cur_subentry; /* previous track stop time = current track start time */ if (prev_subentry != NULL) { gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL); stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate); gst_toc_entry_set_start_stop_times (prev_subentry, start, stop); } id = g_strdup_printf ("%08x", cue->id); cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id); g_free (id); start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate); stop = wav->duration; gst_toc_entry_set_start_stop_times (cur_subentry, start, stop); gst_toc_entry_append_sub_entry (entry, cur_subentry); list = g_list_next (list); } /* add tags in tracks */ list = wav->labls; while (list) { labl = list->data; id = g_strdup_printf ("%08x", labl->cue_point_id); tags = gst_wavparse_get_tags_toc_entry (toc, id); g_free (id); if (tags != NULL) { gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text, NULL); } list = g_list_next (list); } list = wav->notes; while (list) { note = list->data; id = g_strdup_printf ("%08x", note->cue_point_id); tags = gst_wavparse_get_tags_toc_entry (toc, id); g_free (id); if (tags != NULL) { gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT, note->text, NULL); } list = g_list_next (list); } /* send data as TOC */ wav->toc = toc; /* send TOC event */ if (wav->toc) { GST_OBJECT_UNLOCK (wav); gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE)); } return TRUE; } #define MAX_BUFFER_SIZE 4096 static gboolean parse_ds64 (GstWavParse * wav, GstBuffer * buf) { GstMapInfo map; guint32 dataSizeLow, dataSizeHigh; guint32 sampleCountLow, sampleCountHigh; gst_buffer_map (buf, &map, GST_MAP_READ); dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4); dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4); sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4); sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4); gst_buffer_unmap (buf, &map); if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) { wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow; } if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) { wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow; } GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact); return TRUE; } static GstFlowReturn gst_wavparse_stream_headers (GstWavParse * wav) { GstFlowReturn res = GST_FLOW_OK; GstBuffer *buf = NULL; gst_riff_strf_auds *header = NULL; guint32 tag, size; gboolean gotdata = FALSE; GstCaps *caps = NULL; gchar *codec_name = NULL; gint64 upstream_size = 0; GstStructure *s; /* search for "_fmt" chunk, which must be before "data" */ while (!wav->got_fmt) { GstBuffer *extra; if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) return res; gst_adapter_flush (wav->adapter, 8); wav->offset += 8; if (size) { buf = gst_adapter_take_buffer (wav->adapter, size); if (size & 1) gst_adapter_flush (wav->adapter, 1); wav->offset += GST_ROUND_UP_2 (size); } else { buf = gst_buffer_new (); } } 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_RS64_TAG_DS64) { if (!parse_ds64 (wav, buf)) goto fail; else continue; } if (tag != GST_RIFF_TAG_fmt) { GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk", GST_FOURCC_ARGS (tag)); gst_buffer_unref (buf); buf = NULL; continue; } if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header, &extra))) goto parse_header_error; buf = NULL; /* parse_strf_auds() took ownership of buffer */ /* do sanity checks of header fields */ if (header->channels == 0) goto no_channels; if (header->rate == 0) goto no_rate; GST_DEBUG_OBJECT (wav, "creating the caps"); /* Note: gst_riff_create_audio_caps might need to fix values in * the header header depending on the format, so call it first */ /* FIXME: Need to handle the channel reorder map */ caps = gst_riff_create_audio_caps (header->format, NULL, header, extra, NULL, &codec_name, NULL); if (extra) gst_buffer_unref (extra); if (!caps) goto unknown_format; /* If we got raw audio from upstream, we remove the codec_data field, * which may have been added if the wav header included an extended * chunk. We want to keep it for non raw audio. */ s = gst_caps_get_structure (caps, 0); if (s && gst_structure_has_name (s, "audio/x-raw")) { gst_structure_remove_field (s, "codec_data"); } /* do more sanity checks of header fields * (these can be sanitized by gst_riff_create_audio_caps() */ wav->format = header->format; wav->rate = header->rate; wav->channels = header->channels; wav->blockalign = header->blockalign; wav->depth = header->bits_per_sample; wav->av_bps = header->av_bps; wav->vbr = FALSE; g_free (header); header = NULL; /* 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 %u to 0 for mp2/3", wav->av_bps); wav->bps = 0; break; } case GST_RIFF_WAVE_FORMAT_PCM: if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8)) goto invalid_blockalign; /* fall through */ default: if (wav->av_bps > wav->blockalign * wav->rate) goto invalid_bps; /* use the configured bps */ wav->bps = wav->av_bps; break; } wav->width = (wav->blockalign * 8) / wav->channels; wav->bytes_per_sample = wav->channels * wav->width / 8; if (wav->bytes_per_sample <= 0) goto no_bytes_per_sample; GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign); GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width); GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth); GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps); GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate); GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels); GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample); /* bps can be 0 when we don't have a valid bitrate (mostly for compressed * formats). This will make the element output a BYTE format segment and * will not timestamp the outgoing buffers. */ GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps); GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps); /* create pad later so we can sniff the first few bytes * of the real data and correct our caps if necessary */ gst_caps_replace (&wav->caps, caps); gst_caps_replace (&caps, NULL); wav->got_fmt = TRUE; if (wav->tags == NULL) wav->tags = gst_tag_list_new_empty (); { GstCaps *templ_caps = gst_pad_get_pad_template_caps (wav->sinkpad); gst_pb_utils_add_codec_description_to_tag_list (wav->tags, GST_TAG_CONTAINER_FORMAT, templ_caps); gst_caps_unref (templ_caps); } /* If bps is nonzero, then we do have a valid bitrate that can be * announced in a tag list. */ if (wav->bps) { guint bitrate = wav->bps * 8; gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, bitrate, NULL); } if (codec_name) { 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_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size); GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size); /* loop headers until we get data */ while (!gotdata) { if (wav->streaming) { if (!gst_wavparse_peek_chunk_info (wav, &tag, &size)) goto exit; } else { GstMapInfo map; buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, 8, &buf)) != GST_FLOW_OK) #ifdef GSTREAMER_LITE if (res == GST_FLOW_FLUSHING) goto exit; else #endif // GSTREAMER_LITE goto header_read_error; gst_buffer_map (buf, &map, GST_MAP_READ); tag = GST_READ_UINT32_LE (map.data); size = GST_READ_UINT32_LE (map.data + 4); gst_buffer_unmap (buf, &map); } GST_INFO_OBJECT (wav, "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %" G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size); /* Maximum valid size is INT_MAX */ if (size & 0x80000000) { GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff"); size = 0x7fffffff; } /* Clip to upstream size if known */ if (upstream_size > 0 && size + wav->offset > upstream_size) { GST_WARNING_OBJECT (wav, "Clipping chunk size to file size"); g_assert (upstream_size >= wav->offset); size = upstream_size - wav->offset; } /* wav is a st00pid format, we don't know for sure where data starts. * So we have to go bit by bit until we find the 'data' header */ switch (tag) { case GST_RIFF_TAG_data:{ guint64 size64; GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size); size64 = size; if (wav->ignore_length) { GST_DEBUG_OBJECT (wav, "Ignoring length"); size64 = 0; } if (wav->streaming) { gst_adapter_flush (wav->adapter, 8); gotdata = TRUE; } else { gst_buffer_unref (buf); } wav->offset += 8; wav->datastart = wav->offset; /* use size from ds64 chunk if available */ if (size64 == -1 && wav->datasize > 0) { GST_DEBUG_OBJECT (wav, "Using ds64 datasize"); size64 = wav->datasize; } wav->chunk_size = size64; /* If size is zero, then the data chunk probably actually extends to the end of the file */ if (size64 == 0 && upstream_size) { size64 = upstream_size - wav->datastart; } /* Or the file might be truncated */ else if (upstream_size) { size64 = MIN (size64, (upstream_size - wav->datastart)); } wav->datasize = size64; wav->dataleft = size64; wav->end_offset = size64 + wav->datastart; if (!wav->streaming) { /* We will continue parsing tags 'till end */ wav->offset += size64; } GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64); break; } case GST_RIFF_TAG_fact:{ if (wav->fact == 0 && wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 && wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) { const guint data_size = 4; GST_INFO_OBJECT (wav, "Have fact chunk"); if (size < data_size) { if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) { /* need more data */ goto exit; } GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk", data_size, size); break; } /* number of samples (for compressed formats) */ if (wav->streaming) { const guint8 *data = NULL; if (!gst_wavparse_peek_chunk (wav, &tag, &size)) { goto exit; } gst_adapter_flush (wav->adapter, 8); data = gst_adapter_map (wav->adapter, data_size); #ifdef GSTREAMER_LITE if (data == NULL) { goto header_read_error; } #endif // GSTREAMER_LITE wav->fact = GST_READ_UINT32_LE (data); gst_adapter_unmap (wav->adapter); gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size)); } else { gst_buffer_unref (buf); buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset + 8, data_size, &buf)) != GST_FLOW_OK) #ifdef GSTREAMER_LITE if (res == GST_FLOW_FLUSHING) goto exit; else #endif // GSTREAMER_LITE goto header_read_error; gst_buffer_extract (buf, 0, &wav->fact, 4); wav->fact = GUINT32_FROM_LE (wav->fact); gst_buffer_unref (buf); } GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact); wav->offset += 8 + GST_ROUND_UP_2 (size); break; } else { if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) { /* need more data */ goto exit; } } break; } case GST_RIFF_TAG_acid:{ const gst_riff_acid *acid = NULL; const guint data_size = sizeof (gst_riff_acid); gfloat tempo; GST_INFO_OBJECT (wav, "Have acid chunk"); if (size < data_size) { if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) { /* need more data */ goto exit; } GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk", data_size, size); break; } if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) { goto exit; } gst_adapter_flush (wav->adapter, 8); acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter, data_size); tempo = acid->tempo; gst_adapter_unmap (wav->adapter); } else { GstMapInfo map; gst_buffer_unref (buf); buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset + 8, size, &buf)) != GST_FLOW_OK) #ifdef GSTREAMER_LITE if (res == GST_FLOW_FLUSHING) goto exit; else #endif // GSTREAMER_LITE goto header_read_error; gst_buffer_map (buf, &map, GST_MAP_READ); acid = (const gst_riff_acid *) map.data; tempo = acid->tempo; gst_buffer_unmap (buf, &map); } /* send data as tags */ if (!wav->tags) wav->tags = gst_tag_list_new_empty (); gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, GST_TAG_BEATS_PER_MINUTE, tempo, NULL); size = GST_ROUND_UP_2 (size); if (wav->streaming) { gst_adapter_flush (wav->adapter, size); } else { gst_buffer_unref (buf); } wav->offset += 8 + size; break; } /* FIXME: all list tags after data are ignored in streaming mode */ case GST_RIFF_TAG_LIST:{ guint32 ltag; if (wav->streaming) { const guint8 *data = NULL; if (gst_adapter_available (wav->adapter) < 12) { goto exit; } data = gst_adapter_map (wav->adapter, 12); ltag = GST_READ_UINT32_LE (data + 8); gst_adapter_unmap (wav->adapter); } else { gst_buffer_unref (buf); buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, 12, &buf)) != GST_FLOW_OK) #ifdef GSTREAMER_LITE if (res == GST_FLOW_FLUSHING) goto exit; else #endif // GSTREAMER_LITE goto header_read_error; gst_buffer_extract (buf, 8, <ag, 4); ltag = GUINT32_FROM_LE (ltag); } switch (ltag) { case GST_RIFF_LIST_INFO:{ const gint data_size = size - 4; GstTagList *new; GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size); if (wav->streaming) { if (!gst_wavparse_peek_chunk (wav, &tag, &size)) { goto exit; } gst_adapter_flush (wav->adapter, 12); wav->offset += 12; if (data_size > 0) { buf = gst_adapter_take_buffer (wav->adapter, data_size); if (data_size & 1) gst_adapter_flush (wav->adapter, 1); } } else { wav->offset += 12; gst_buffer_unref (buf); buf = NULL; if (data_size > 0) { if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, data_size, &buf)) != GST_FLOW_OK) #ifdef GSTREAMER_LITE if (res == GST_FLOW_FLUSHING) goto exit; else #endif // GSTREAMER_LITE goto header_read_error; } } if (data_size > 0) { /* parse tags */ gst_riff_parse_info (GST_ELEMENT (wav), buf, &new); if (new) { GstTagList *old = wav->tags; wav->tags = gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE); if (old) gst_tag_list_unref (old); gst_tag_list_unref (new); } gst_buffer_unref (buf); wav->offset += GST_ROUND_UP_2 (data_size); } break; } case GST_RIFF_LIST_adtl:{ const gint data_size = size - 4; GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size); if (wav->streaming) { const guint8 *data = NULL; gst_adapter_flush (wav->adapter, 12); wav->offset += 12; data = gst_adapter_map (wav->adapter, data_size); gst_wavparse_adtl_chunk (wav, data, data_size); gst_adapter_unmap (wav->adapter); } else { GstMapInfo map; gst_buffer_unref (buf); buf = NULL; wav->offset += 12; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, data_size, &buf)) != GST_FLOW_OK) goto header_read_error; gst_buffer_map (buf, &map, GST_MAP_READ); gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data, data_size); gst_buffer_unmap (buf, &map); } wav->offset += GST_ROUND_UP_2 (data_size); break; } default: GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (ltag)); if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) /* need more data */ goto exit; break; } break; } case GST_RIFF_TAG_cue:{ const guint data_size = size; GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size); if (wav->streaming) { const guint8 *data = NULL; if (!gst_wavparse_peek_chunk (wav, &tag, &size)) { goto exit; } gst_adapter_flush (wav->adapter, 8); wav->offset += 8; data = gst_adapter_map (wav->adapter, data_size); if (!gst_wavparse_cue_chunk (wav, data, data_size)) { goto header_read_error; } gst_adapter_unmap (wav->adapter); } else { GstMapInfo map; wav->offset += 8; gst_buffer_unref (buf); buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, data_size, &buf)) != GST_FLOW_OK) goto header_read_error; gst_buffer_map (buf, &map, GST_MAP_READ); if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data, data_size)) { goto header_read_error; } gst_buffer_unmap (buf, &map); } size = GST_ROUND_UP_2 (size); if (wav->streaming) { gst_adapter_flush (wav->adapter, size); } else { gst_buffer_unref (buf); } size = GST_ROUND_UP_2 (size); wav->offset += size; break; } case GST_RIFF_TAG_smpl:{ const gint data_size = size; GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size); if (wav->streaming) { const guint8 *data = NULL; if (!gst_wavparse_peek_chunk (wav, &tag, &size)) { goto exit; } gst_adapter_flush (wav->adapter, 8); wav->offset += 8; data = gst_adapter_map (wav->adapter, data_size); if (!gst_wavparse_smpl_chunk (wav, data, data_size)) { goto header_read_error; } gst_adapter_unmap (wav->adapter); } else { GstMapInfo map; wav->offset += 8; gst_buffer_unref (buf); buf = NULL; if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, data_size, &buf)) != GST_FLOW_OK) goto header_read_error; gst_buffer_map (buf, &map, GST_MAP_READ); if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data, data_size)) { goto header_read_error; } gst_buffer_unmap (buf, &map); } size = GST_ROUND_UP_2 (size); if (wav->streaming) { gst_adapter_flush (wav->adapter, size); } else { gst_buffer_unref (buf); } size = GST_ROUND_UP_2 (size); wav->offset += size; break; } default: GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)); if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) /* need more data */ goto exit; break; } if (upstream_size && (wav->offset >= upstream_size)) { /* Now we are gone through the whole file */ gotdata = TRUE; } } GST_DEBUG_OBJECT (wav, "Finished parsing headers"); if (wav->bps <= 0 && wav->fact) { #if 0 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */ wav->bps = (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize, (guint64) wav->fact); GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps); #endif wav->vbr = TRUE; } if (gst_wavparse_calculate_duration (wav)) { gst_segment_init (&wav->segment, GST_FORMAT_TIME); if (!wav->ignore_length) wav->segment.duration = wav->duration; if (!wav->toc) gst_wavparse_create_toc (wav); } else { /* no bitrate, let downstream peer do the math, we'll feed it bytes. */ gst_segment_init (&wav->segment, GST_FORMAT_BYTES); if (!wav->ignore_length) wav->segment.duration = 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 */ gst_event_replace (&wav->seek_event, NULL); /* we just started, we are discont */ wav->discont = TRUE; wav->state = GST_WAVPARSE_DATA; /* determine reasonable max buffer size, * that is, buffers not too small either size or time wise * so we do not end up with too many of them */ /* var abuse */ if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size)) wav->max_buf_size = upstream_size; else wav->max_buf_size = 0; wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE); if (wav->blockalign > 0) wav->max_buf_size -= (wav->max_buf_size % wav->blockalign); GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size); return GST_FLOW_OK; /* ERROR */ exit: { g_free (codec_name); g_free (header); if (caps) gst_caps_unref (caps); return res; } fail: { res = GST_FLOW_ERROR; goto exit; } parse_header_error: { GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't parse audio header")); goto fail; } no_channels: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Stream claims to contain no channels - invalid data")); goto fail; } no_rate: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Stream with sample_rate == 0 - invalid data")); goto fail; } invalid_blockalign: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Stream claims blockalign = %u, which is more than %u - invalid data", wav->blockalign, wav->channels * ((wav->depth + 7) / 8))); goto fail; } invalid_bps: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Stream claims av_bsp = %u, which is more than %u - invalid data", wav->av_bps, wav->blockalign * wav->rate)); goto fail; } no_bytes_per_sample: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Could not caluclate bytes per sample - invalid data")); goto fail; } unknown_format: { GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), ("No caps found for format 0x%x, %u channels, %u Hz", wav->format, wav->channels, wav->rate)); goto fail; } header_read_error: { GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res))); goto fail; } } /* * 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; 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"); gst_event_replace (&wav->seek_event, event); /* we always return true */ res = TRUE; } break; default: break; } gst_event_unref (event); return res; } static gboolean gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob) { GstStructure *s; s = gst_caps_get_structure (caps, 0); if (!gst_structure_has_name (s, "audio/x-dts")) return FALSE; /* typefind behavior for DTS: * MAXIMUM: multiple frame syncs detected, certainly DTS * LIKELY: single frame sync at offset 0. Maybe DTS? * POSSIBLE: single frame sync, not at offset 0. Highly unlikely * to be DTS. */ if (prob > GST_TYPE_FIND_LIKELY) return TRUE; if (prob <= GST_TYPE_FIND_POSSIBLE) return FALSE; /* for maybe, check for at least a valid-looking rate and channels */ if (!gst_structure_has_field (s, "channels")) return FALSE; /* and for extra assurance we could also check the rate from the DTS frame * against the one in the wav header, but for now let's not do that */ return gst_structure_has_field (s, "rate"); } static GstTagList * gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope) { GstTagList *tags = NULL; GstEvent *ev; gint i; i = 0; while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) { gst_event_parse_tag (ev, &tags); if (tags != NULL && gst_tag_list_get_scope (tags) == scope) { tags = gst_tag_list_copy (tags); gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT); gst_event_unref (ev); break; } tags = NULL; gst_event_unref (ev); } return tags; } static void gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) { GstStructure *s; GstTagList *tags, *utags; GST_DEBUG_OBJECT (wav, "adding src pad"); g_assert (wav->caps != NULL); s = gst_caps_get_structure (wav->caps, 0); if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) { GstTypeFindProbability prob; GstCaps *tf_caps; tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob); if (tf_caps != NULL) { GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob); if (gst_wavparse_have_dts_caps (tf_caps, prob)) { GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM"); gst_caps_unref (wav->caps); wav->caps = tf_caps; gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, GST_TAG_AUDIO_CODEC, "dts", NULL); } else { GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream " "marked as raw PCM audio, but ignoring for now", tf_caps); gst_caps_unref (tf_caps); } } } gst_pad_set_caps (wav->srcpad, wav->caps); if (wav->start_segment) { GST_DEBUG_OBJECT (wav, "Send start segment event on newpad"); gst_pad_push_event (wav->srcpad, wav->start_segment); wav->start_segment = NULL; } /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now * that there'll be only one scope/type of tag list from upstream, if any */ utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL); if (utags == NULL) utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM); /* if there's a tag upstream it's probably been added to override the * tags from inside the wav header, so keep upstream tags if in doubt */ tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP); if (wav->tags != NULL) { gst_tag_list_unref (wav->tags); wav->tags = NULL; } if (utags != NULL) gst_tag_list_unref (utags); /* send tags downstream, if any */ if (tags != NULL) gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags)); } 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); if ((wav->dataleft == 0 || wav->dataleft < wav->blockalign)) { /* In case chunk size is not declared in the begining get size from the * file size directly */ if (wav->chunk_size == 0) { gint64 upstream_size = 0; /* Get the size of the file */ if (!gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size)) goto found_eos; if (upstream_size < wav->offset + wav->datastart) goto found_eos; /* If file has updated since the beggining continue reading the file */ wav->dataleft = upstream_size - wav->offset - wav->datastart; wav->end_offset = upstream_size; /* Get the next n bytes and output them, if we can */ if (wav->dataleft == 0 || wav->dataleft < wav->blockalign) goto found_eos; } else { 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), wav->max_buf_size * ABS (wav->segment.rate)); if (desired >= wav->blockalign && wav->blockalign > 0) desired -= (desired % wav->blockalign); #ifdef GSTREAMER_LITE if (desired == 0) { GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("Invalid stream")); return GST_FLOW_ERROR; } #endif // GSTREAMER_LITE 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); guint extra; /* flush some bytes if evil upstream sends segment that starts * before data or does is not send sample aligned segment */ if (G_LIKELY (wav->offset >= wav->datastart)) { extra = (wav->offset - wav->datastart) % wav->bytes_per_sample; } else { extra = wav->datastart - wav->offset; } if (G_UNLIKELY (extra)) { extra = wav->bytes_per_sample - extra; if (extra <= avail) { GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra); gst_adapter_flush (wav->adapter, extra); wav->offset += extra; wav->dataleft -= extra; goto iterate_adapter; } else { GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail); gst_adapter_clear (wav->adapter); wav->offset += avail; wav->dataleft -= avail; return GST_FLOW_OK; } } if (avail < desired) { GST_LOG_OBJECT (wav, "Got only %u 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; /* we may get a short buffer at the end of the file */ if (gst_buffer_get_size (buf) < desired) { gsize size = gst_buffer_get_size (buf); GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size); if (size >= wav->blockalign) { if (wav->blockalign > 0) { buf = gst_buffer_make_writable (buf); gst_buffer_resize (buf, 0, size - (size % wav->blockalign)); } } else { gst_buffer_unref (buf); goto found_eos; } } } obtained = gst_buffer_get_size (buf); /* our positions in bytes */ pos = wav->offset - wav->datastart; nextpos = pos + obtained; /* update offsets, does not overflow. */ buf = gst_buffer_make_writable (buf); GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample; GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample; /* first chunk of data? create the source pad. We do this only here so * we can detect broken .wav files with dts disguised as raw PCM (sigh) */ if (G_UNLIKELY (wav->first)) { wav->first = FALSE; /* this will also push the segment events */ gst_wavparse_add_src_pad (wav, buf); } else { /* If we have a pending start segment, send it now. */ if (G_UNLIKELY (wav->start_segment != NULL)) { gst_pad_push_event (wav->srcpad, wav->start_segment); wav->start_segment = NULL; } } if (wav->bps > 0) { /* and timestamps if we have a bitrate, be careful for overflows */ timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps); next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps); duration = next_timestamp - timestamp; /* update current running segment position */ if (G_LIKELY (next_timestamp >= wav->segment.start)) wav->segment.position = next_timestamp; } else if (wav->fact) { guint64 bps = gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact); /* and timestamps if we have a bitrate, be careful for overflows */ timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps); next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps); duration = next_timestamp - timestamp; } else { /* no bitrate, all we know is that the first sample has timestamp 0, all * other positions and durations have unknown timestamp. */ if (pos == 0) timestamp = 0; else timestamp = GST_CLOCK_TIME_NONE; duration = GST_CLOCK_TIME_NONE; /* update current running segment position with byte offset */ if (G_LIKELY (nextpos >= wav->segment.start)) wav->segment.position = nextpos; } if ((pos > 0) && wav->vbr) { /* don't set timestamps for VBR files if it's not the first buffer */ timestamp = GST_CLOCK_TIME_NONE; duration = GST_CLOCK_TIME_NONE; } if (wav->discont) { GST_DEBUG_OBJECT (wav, "marking DISCONT"); GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); wav->discont = FALSE; } GST_BUFFER_TIMESTAMP (buf) = timestamp; GST_BUFFER_DURATION (buf) = duration; GST_LOG_OBJECT (wav, "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration), gst_buffer_get_size (buf)); if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK) goto push_error; if (obtained < wav->dataleft) { wav->offset += obtained; wav->dataleft -= obtained; } else { wav->offset += wav->dataleft; wav->dataleft = 0; } /* Iterate until need more data, so adapter size won't grow */ if (wav->streaming) { GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset, wav->end_offset); goto iterate_adapter; } return res; /* ERROR */ found_eos: { GST_DEBUG_OBJECT (wav, "found EOS"); return GST_FLOW_EOS; } pull_error: { /* check if we got EOS */ if (res == GST_FLOW_EOS) goto found_eos; GST_WARNING_OBJECT (wav, "Error getting %" G_GINT64_FORMAT " bytes from the " "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft); return res; } push_error: { GST_INFO_OBJECT (wav, "Error pushing on srcpad %s:%s, reason %s, is linked? = %d", GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res), gst_pad_is_linked (wav->srcpad)); return res; } } static void gst_wavparse_loop (GstPad * pad) { GstFlowReturn ret; GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); GstEvent *event; gchar *stream_id; GST_LOG_OBJECT (wav, "process data"); switch (wav->state) { case GST_WAVPARSE_START: GST_INFO_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK) goto pause; stream_id = gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL); event = gst_event_new_stream_start (stream_id); gst_event_set_group_id (event, gst_util_group_id_next ()); gst_pad_push_event (wav->srcpad, event); g_free (stream_id); wav->state = GST_WAVPARSE_HEADER; /* fall-through */ case GST_WAVPARSE_HEADER: GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER"); if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) goto pause; wav->state = GST_WAVPARSE_DATA; GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA"); /* fall-through */ case GST_WAVPARSE_DATA: if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) goto pause; break; default: g_assert_not_reached (); } return; /* ERRORS */ pause: { const gchar *reason = gst_flow_get_name (ret); GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason); gst_pad_pause_task (pad); if (ret == GST_FLOW_EOS) { /* handle end-of-stream/segment */ /* so align our position with the end of it, if there is one * this ensures a subsequent will arrive at correct base/acc time */ if (wav->segment.format == GST_FORMAT_TIME) { if (wav->segment.rate > 0.0 && GST_CLOCK_TIME_IS_VALID (wav->segment.stop)) wav->segment.position = wav->segment.stop; else if (wav->segment.rate < 0.0) wav->segment.position = wav->segment.start; } if (wav->state == GST_WAVPARSE_START || !wav->caps) { GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL), ("No valid input found before end of stream")); gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); } else { /* 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)); gst_pad_push_event (wav->srcpad, gst_event_new_segment_done (wav->segment.format, stop)); } else { gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); } } } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) { /* for fatal errors we post an error message, post the error * first so the app knows about the error first. */ GST_ELEMENT_FLOW_ERROR (wav, ret); gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); } return; } } static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstFlowReturn ret; GstWavParse *wav = GST_WAVPARSE (parent); GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes", gst_buffer_get_size (buf)); gst_adapter_push (wav->adapter, buf); switch (wav->state) { case GST_WAVPARSE_START: GST_INFO_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK) goto done; if (wav->state != GST_WAVPARSE_HEADER) break; /* otherwise fall-through */ case GST_WAVPARSE_HEADER: GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER"); if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) goto done; if (!wav->got_fmt || wav->datastart == 0) break; wav->state = GST_WAVPARSE_DATA; GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA"); /* fall-through */ case GST_WAVPARSE_DATA: if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) wav->discont = TRUE; if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) goto done; break; default: g_return_val_if_reached (GST_FLOW_ERROR); } done: if (G_UNLIKELY (wav->abort_buffering)) { wav->abort_buffering = FALSE; ret = GST_FLOW_ERROR; /* sort of demux/parse error */ GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size")); } #ifdef GSTREAMER_LITE else if (G_UNLIKELY(ret != GST_FLOW_OK && ret != GST_FLOW_FLUSHING && ret != GST_FLOW_EOS)) { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (_("Internal data flow error.")), ("streaming task paused, reason %s (%d)", gst_flow_get_name (ret), ret)); } #endif // GSTREAMER_LITE return ret; } static GstFlowReturn gst_wavparse_flush_data (GstWavParse * wav) { GstFlowReturn ret = GST_FLOW_OK; guint av; if ((av = gst_adapter_available (wav->adapter)) > 0) { ret = gst_wavparse_stream_data (wav); } return ret; } static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstWavParse *wav = GST_WAVPARSE (parent); gboolean ret = TRUE; GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { /* discard, we'll come up with proper src caps */ gst_event_unref (event); break; } case GST_EVENT_SEGMENT: { gint64 start, stop, offset = 0, end_offset = -1; GstSegment segment; /* some debug output */ gst_event_copy_segment (event, &segment); GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT, &segment); if (wav->state != GST_WAVPARSE_DATA) { GST_DEBUG_OBJECT (wav, "still starting, eating event"); goto exit; } /* now we are either committed to TIME or BYTE format, * and we only expect a BYTE segment, e.g. following a seek */ if (segment.format == GST_FORMAT_BYTES) { /* handle (un)signed issues */ start = segment.start; stop = segment.stop; if (start > 0) { offset = start; start -= wav->datastart; start = MAX (start, 0); } if (stop > 0) { end_offset = stop; stop -= wav->datastart; stop = MAX (stop, 0); } if (wav->segment.format == GST_FORMAT_TIME) { guint64 bps = wav->bps; /* operating in format TIME, so we can convert */ if (!bps && wav->fact) bps = gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact); if (bps) { if (start >= 0) start = gst_util_uint64_scale_ceil (start, GST_SECOND, (guint64) wav->bps); if (stop >= 0) stop = gst_util_uint64_scale_ceil (stop, GST_SECOND, (guint64) wav->bps); } } } else { GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring"); goto exit; } segment.start = start; segment.stop = stop; /* accept upstream's notion of segment and distribute along */ segment.format = wav->segment.format; segment.time = segment.position = segment.start; segment.duration = wav->segment.duration; segment.base = gst_segment_to_running_time (&wav->segment, GST_FORMAT_TIME, wav->segment.position); gst_segment_copy_into (&segment, &wav->segment); /* also store the newsegment event for the streaming thread */ if (wav->start_segment) gst_event_unref (wav->start_segment); GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment); wav->start_segment = gst_event_new_segment (&segment); /* stream leftover data in current segment */ gst_wavparse_flush_data (wav); /* and set up streaming thread for next one */ wav->offset = offset; wav->end_offset = end_offset; if (wav->datasize > 0 && (wav->end_offset == -1 || wav->end_offset > wav->datastart + wav->datasize)) wav->end_offset = wav->datastart + wav->datasize; if (wav->end_offset != -1) { #ifdef GSTREAMER_LITE wav->dataleft = MIN(wav->datasize, wav->end_offset - wav->offset); #else wav->dataleft = wav->end_offset - wav->offset; #endif // GSTREAMER_LITE } else { /* infinity; upstream will EOS when done */ wav->dataleft = G_MAXUINT64; } exit: gst_event_unref (event); break; } #ifdef GSTREAMER_LITE case FX_EVENT_RANGE_READY: // This event appears only in pull mode during outrange seeking. ret = gst_pad_start_task (pad, (GstTaskFunction) gst_wavparse_loop, pad, NULL); gst_event_unref(event); break; #endif // GSTREAMER_LITE case GST_EVENT_EOS: if (wav->state == GST_WAVPARSE_START || !wav->caps) { GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL), ("No valid input found before end of stream")); } else { /* add pad if needed so EOS is seen downstream */ if (G_UNLIKELY (wav->first)) { wav->first = FALSE; gst_wavparse_add_src_pad (wav, NULL); } else { /* stream leftover data in current segment */ gst_wavparse_flush_data (wav); } } /* fall-through */ case GST_EVENT_FLUSH_STOP: { GstClockTime dur; if (wav->adapter) gst_adapter_clear (wav->adapter); wav->discont = TRUE; dur = wav->segment.duration; gst_segment_init (&wav->segment, wav->segment.format); wav->segment.duration = dur; /* fall-through */ } default: ret = gst_pad_event_default (wav->sinkpad, parent, event); break; } return ret; } #if 0 /* convert and query stuff */ static const GstFormat * gst_wavparse_get_formats (GstPad * pad) { static const 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_PARENT (pad)); if (*dest_format == src_format) { *dest_value = src_value; return TRUE; } if ((wavparse->bps == 0) && !wavparse->fact) goto no_bps_fact; GST_INFO_OBJECT (wavparse, "converting value from %s to %s", gst_format_get_name (src_format), gst_format_get_name (*dest_format)); switch (src_format) { case GST_FORMAT_BYTES: switch (*dest_format) { case GST_FORMAT_DEFAULT: *dest_value = src_value / wavparse->bytes_per_sample; /* make sure we end up on a sample boundary */ *dest_value -= *dest_value % wavparse->bytes_per_sample; break; case GST_FORMAT_TIME: /* src_value + datastart = offset */ GST_INFO_OBJECT (wavparse, "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value, wavparse->offset); if (wavparse->bps > 0) *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND, (guint64) wavparse->bps); else if (wavparse->fact) { guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize, wavparse->rate, wavparse->fact); *dest_value = gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps); } else { res = FALSE; } break; default: res = FALSE; goto done; } break; case GST_FORMAT_DEFAULT: switch (*dest_format) { case GST_FORMAT_BYTES: *dest_value = src_value * wavparse->bytes_per_sample; break; case GST_FORMAT_TIME: *dest_value = gst_util_uint64_scale (src_value, GST_SECOND, (guint64) wavparse->rate); break; default: res = FALSE; goto done; } break; case GST_FORMAT_TIME: switch (*dest_format) { case GST_FORMAT_BYTES: if (wavparse->bps > 0) *dest_value = gst_util_uint64_scale (src_value, (guint64) wavparse->bps, GST_SECOND); else { guint64 bps = gst_util_uint64_scale_int (wavparse->datasize, wavparse->rate, wavparse->fact); *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND); } /* make sure we end up on a sample boundary */ *dest_value -= *dest_value % wavparse->blockalign; break; case GST_FORMAT_DEFAULT: *dest_value = gst_util_uint64_scale (src_value, (guint64) wavparse->rate, GST_SECOND); break; default: res = FALSE; goto done; } break; default: res = FALSE; goto done; } done: return res; /* ERRORS */ no_bps_fact: { GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert"); res = FALSE; goto done; } } /* handle queries for location and length in requested format */ static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res = TRUE; GstWavParse *wav = GST_WAVPARSE (parent); /* only if we know */ if (wav->state != GST_WAVPARSE_DATA) { return FALSE; } GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { gint64 curb; gint64 cur; GstFormat format; /* this is not very precise, as we have pushed severla buffer upstream for prerolling */ curb = wav->offset - wav->datastart; gst_query_parse_position (query, &format, NULL); GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb); switch (format) { case GST_FORMAT_BYTES: format = GST_FORMAT_BYTES; cur = curb; break; default: res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb, &format, &cur); break; } if (res) gst_query_set_position (query, format, cur); break; } case GST_QUERY_DURATION: { gint64 duration = 0; GstFormat format; if (wav->ignore_length) { res = FALSE; break; } gst_query_parse_duration (query, &format, NULL); switch (format) { case GST_FORMAT_BYTES:{ format = GST_FORMAT_BYTES; duration = wav->datasize; break; } case GST_FORMAT_TIME: if ((res = gst_wavparse_calculate_duration (wav))) { duration = wav->duration; } break; default: res = FALSE; break; } if (res) gst_query_set_duration (query, format, duration); break; } case GST_QUERY_CONVERT: { gint64 srcvalue, dstvalue; GstFormat srcformat, dstformat; gst_query_parse_convert (query, &srcformat, &srcvalue, &dstformat, &dstvalue); res = gst_wavparse_pad_convert (pad, srcformat, srcvalue, &dstformat, &dstvalue); if (res) gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue); break; } case GST_QUERY_SEEKING:{ GstFormat fmt; gboolean seekable = FALSE; gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL); if (fmt == wav->segment.format) { if (wav->streaming) { GstQuery *q; q = gst_query_new_seeking (GST_FORMAT_BYTES); if ((res = gst_pad_peer_query (wav->sinkpad, q))) { gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL); GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable); } gst_query_unref (q); } else { GST_LOG_OBJECT (wav, "looping => seekable"); seekable = TRUE; res = TRUE; } } else if (fmt == GST_FORMAT_TIME) { res = TRUE; } if (res) { gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration); } break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } #ifdef GSTREAMER_LITE static gboolean gst_wavparse_sink_query (GstPad* pad, GstObject *parent, GstQuery* query) { gboolean result = TRUE; switch (GST_QUERY_TYPE(query)) { case GST_QUERY_CUSTOM: { const GstStructure *s = gst_query_get_structure(query); if (gst_structure_has_name(s, GETRANGE_QUERY_NAME)) gst_structure_set(s, GETRANGE_QUERY_SUPPORTS_FIELDNANE, GETRANGE_QUERY_SUPPORTS_FIELDTYPE, TRUE, NULL); break; } default: result = gst_pad_query_default(pad, parent, query); break; } return result; } #endif // GSTREAMER_LITE static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstWavParse *wavparse = GST_WAVPARSE (parent); gboolean res = FALSE; GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: /* can only handle events when we are in the data state */ if (wavparse->state == GST_WAVPARSE_DATA) { res = gst_wavparse_perform_seek (wavparse, event); } gst_event_unref (event); break; case GST_EVENT_TOC_SELECT: { char *uid = NULL; GstTocEntry *entry = NULL; GstEvent *seek_event; gint64 start_pos; if (!wavparse->toc) { GST_DEBUG_OBJECT (wavparse, "no TOC to select"); return FALSE; } else { gst_event_parse_toc_select (event, &uid); if (uid != NULL) { GST_OBJECT_LOCK (wavparse); entry = gst_toc_find_entry (wavparse->toc, uid); if (entry == NULL) { GST_OBJECT_UNLOCK (wavparse); GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s", uid); res = FALSE; } else { gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL); GST_OBJECT_UNLOCK (wavparse); seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1); res = gst_wavparse_perform_seek (wavparse, seek_event); gst_event_unref (seek_event); } g_free (uid); } else { GST_WARNING_OBJECT (wavparse, "received empty TOC select event"); res = FALSE; } } 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, GstObject * parent) { GstWavParse *wav = GST_WAVPARSE (parent); GstQuery *query; gboolean pull_mode; if (wav->adapter) { gst_adapter_clear (wav->adapter); g_object_unref (wav->adapter); wav->adapter = NULL; } query = gst_query_new_scheduling (); if (!gst_pad_peer_query (sinkpad, query)) { gst_query_unref (query); goto activate_push; } pull_mode = gst_query_has_scheduling_mode_with_flags (query, GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE); gst_query_unref (query); if (!pull_mode) goto activate_push; GST_DEBUG_OBJECT (sinkpad, "activating pull"); wav->streaming = FALSE; return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE); activate_push: { GST_DEBUG_OBJECT (sinkpad, "activating push"); wav->streaming = TRUE; wav->adapter = gst_adapter_new (); return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE); } } static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent, GstPadMode mode, gboolean active) { gboolean res; switch (mode) { case GST_PAD_MODE_PUSH: res = TRUE; break; case GST_PAD_MODE_PULL: if (active) { /* if we have a scheduler we can start the task */ res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad, NULL); } else { res = gst_pad_stop_task (sinkpad); } break; default: res = FALSE; break; } return res; } 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: #ifndef GSTREAMER_LITE gst_wavparse_destroy_sourcepad (wav); #endif // GSTREAMER_LITE gst_wavparse_reset (wav); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static void gst_wavparse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWavParse *self; g_return_if_fail (GST_IS_WAVPARSE (object)); self = GST_WAVPARSE (object); switch (prop_id) { case PROP_IGNORE_LENGTH: self->ignore_length = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec); } } static void gst_wavparse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWavParse *self; g_return_if_fail (GST_IS_WAVPARSE (object)); self = GST_WAVPARSE (object); switch (prop_id) { case PROP_IGNORE_LENGTH: g_value_set_boolean (value, self->ignore_length); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec); } } #ifdef GSTREAMER_LITE gboolean plugin_init_wavparse (GstPlugin * plugin) #else // GSTREAMER_LITE static gboolean plugin_init (GstPlugin * plugin) #endif // GSTREAMER_LITE { gst_riff_init (); return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY, GST_TYPE_WAVPARSE); } #ifndef GSTREAMER_LITE 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) #endif // GSTREAMER_LITE