From 2644d7178b5f8b141d20d52fb845e269c1dfc4a3 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 14 Feb 2007 09:55:47 +0000 Subject: [PATCH] gst/wavparse/gstwavparse.*: Update docs. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: Update docs. Use boilerplate. Various code cleanups. When the bitrate is not known (bps == 0 or compressed formats) let downstream element guestimate the duration and position and don't generate timestamps or durations. Fixes #405213. Fix EOS and ERROR conditions in chain mode, we just need to forward the error flowreturn upstream. --- ChangeLog | 23 +++ gst/wavparse/gstwavparse.c | 491 +++++++++++++++++++++++---------------------- gst/wavparse/gstwavparse.h | 7 +- 3 files changed, 281 insertions(+), 240 deletions(-) diff --git a/ChangeLog b/ChangeLog index 161ad53..f2360e2 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,26 @@ +2007-02-14 Wim Taymans,,, + + * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init), + (gst_wavparse_reset), (gst_wavparse_init), + (gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt), + (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), + (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), + (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), + (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), + (gst_wavparse_loop), (gst_wavparse_chain), + (gst_wavparse_pad_convert), (gst_wavparse_pad_query), + (gst_wavparse_srcpad_event), (gst_wavparse_change_state), + (plugin_init): + * gst/wavparse/gstwavparse.h: + Update docs. + Use boilerplate. + Various code cleanups. + When the bitrate is not known (bps == 0 or compressed formats) let + downstream element guestimate the duration and position and don't + generate timestamps or durations. Fixes #405213. + Fix EOS and ERROR conditions in chain mode, we just need to forward the + error flowreturn upstream. + 2007-02-13 Jan Schmidt * ext/gconf/Makefile.am: diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 72f0e94..1cef0e4 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -26,6 +26,10 @@ * * 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 * * @@ -38,11 +42,11 @@ * * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink * - * Stream data from + * Stream data from a network url. * * * - * Last reviewed on 2006-03-03 (0.10.3) + * Last reviewed on 2007-02-14 (0.10.6) */ /* @@ -63,9 +67,6 @@ GST_DEBUG_CATEGORY_STATIC (wavparse_debug); #define GST_CAT_DEFAULT (wavparse_debug) -static void gst_wavparse_base_init (gpointer g_class); -static void gst_wavparse_class_init (GstWavParseClass * klass); -static void gst_wavparse_init (GstWavParse * wavparse); static void gst_wavparse_dispose (GObject * object); static gboolean gst_wavparse_sink_activate (GstPad * sinkpad); @@ -73,20 +74,18 @@ static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active); static gboolean gst_wavparse_send_event (GstElement * element, GstEvent * event); -static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf); static GstStateChangeReturn gst_wavparse_change_state (GstElement * element, GstStateChange transition); -static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query); static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad); +static gboolean gst_wavparse_pad_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 void gst_wavparse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); static const GstElementDetails gst_wavparse_details = GST_ELEMENT_DETAILS ("WAV audio demuxer", @@ -141,33 +140,11 @@ static GstStaticPadTemplate src_template_factory = ); -static GstElementClass *parent_class = NULL; - -GType -gst_wavparse_get_type (void) -{ - static GType wavparse_type = 0; - - if (!wavparse_type) { - static const GTypeInfo wavparse_info = { - sizeof (GstWavParseClass), - gst_wavparse_base_init, - NULL, - (GClassInitFunc) gst_wavparse_class_init, - NULL, - NULL, - sizeof (GstWavParse), - 0, - (GInstanceInitFunc) gst_wavparse_init, - }; - - wavparse_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse", - &wavparse_info, 0); - } - return wavparse_type; -} +#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) @@ -193,7 +170,6 @@ gst_wavparse_class_init (GstWavParseClass * klass) parent_class = g_type_class_peek_parent (klass); - object_class->get_property = gst_wavparse_get_property; object_class->dispose = gst_wavparse_dispose; gstelement_class->change_state = gst_wavparse_change_state; @@ -217,7 +193,6 @@ gst_wavparse_dispose (GObject * object) G_OBJECT_CLASS (parent_class)->dispose (object); } - static void gst_wavparse_reset (GstWavParse * wavparse) { @@ -242,13 +217,12 @@ gst_wavparse_reset (GstWavParse * wavparse) if (wavparse->seek_event) gst_event_unref (wavparse->seek_event); wavparse->seek_event = NULL; - - /* we keep the segment info in time */ - gst_segment_init (&wavparse->segment, GST_FORMAT_TIME); + if (wavparse->adapter) + gst_adapter_clear (wavparse->adapter); } static void -gst_wavparse_init (GstWavParse * wavparse) +gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class) { gst_wavparse_reset (wavparse); @@ -261,7 +235,7 @@ gst_wavparse_init (GstWavParse * wavparse) GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull)); gst_pad_set_chain_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_chain)); - gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad); + gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad); /* src, will be created later */ wavparse->srcpad = NULL; @@ -271,7 +245,7 @@ static void gst_wavparse_destroy_sourcepad (GstWavParse * wavparse) { if (wavparse->srcpad) { - gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad); + gst_element_remove_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad); wavparse->srcpad = NULL; } } @@ -296,21 +270,6 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse) GST_DEBUG_OBJECT (wavparse, "srcpad created"); } -static void -gst_wavparse_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec) -{ - GstWavParse *wavparse; - - wavparse = GST_WAVPARSE (object); - - switch (prop_id) { - default: - break; - } -} - - #if 0 static void @@ -566,52 +525,68 @@ gst_wavparse_fmt (GstWavParse * wav) gst_riff_strf_auds *header = NULL; GstCaps *caps; - if (!gst_riff_read_strf_auds (wav, &header)) { - g_warning ("Not fmt"); - return FALSE; - } + 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) { - GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), - ("Stream claims to contain zero channels - invalid data")); - g_free (header); - return FALSE; - } + 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) { - GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), - ("Stream claims to bitrate of <= zero - invalid data")); - g_free (header); - return FALSE; - } + 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) { - gst_wavparse_create_sourcepad (wav); - gst_pad_use_fixed_caps (wav->srcpad); - gst_pad_set_active (wav->srcpad, TRUE); - gst_pad_set_caps (wav->srcpad, caps); - gst_caps_free (caps); - gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad); - gst_element_no_more_pads (GST_ELEMENT (wav)); - GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels); - } else { + 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; } - - return TRUE; } static gboolean @@ -717,8 +692,6 @@ gst_wavparse_other (GstWavParse * wav) } #endif - - static gboolean gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf) { @@ -736,7 +709,7 @@ gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf) not_wav: { GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL), - ("File is not an WAVE file: %" GST_FOURCC_FORMAT, + ("File is not a WAVE file: %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (doctype))); return FALSE; } @@ -751,7 +724,7 @@ gst_wavparse_stream_init (GstWavParse * wav) if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, 12, &buf)) != GST_FLOW_OK) return res; - else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf)) + else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf)) return GST_FLOW_ERROR; wav->offset += 12; @@ -781,17 +754,22 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) 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 != GST_FORMAT_TIME) { - GstFormat fmt; - - fmt = GST_FORMAT_TIME; + 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); @@ -809,34 +787,56 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) stop_type = GST_SEEK_TYPE_SET; } + /* get flush flag */ flush = flags & GST_SEEK_FLAG_FLUSH; - if (flush && wav->srcpad) { - GST_DEBUG_OBJECT (wav, "sending flush start"); - gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ()); + /* 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); } - if ((stop = seeksegment.stop) == GST_CLOCK_TIME_NONE) + /* 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)) { - wav->offset = - gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND); + /* 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 = + gst_util_uint64_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); @@ -848,7 +848,10 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) } if (stop_type != GST_SEEK_TYPE_NONE) { - wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND); + if (wav->bps) + wav->end_offset = gst_util_uint64_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); @@ -865,6 +868,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) 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; @@ -876,11 +880,12 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) /* 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 last_stop. */ + * 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); @@ -889,20 +894,23 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) 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 (wav), - gst_message_new_segment_start (GST_OBJECT (wav), + 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 send the newsegment */ - GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT + /* 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. */ @@ -913,6 +921,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) 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, @@ -924,6 +933,11 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) 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."); @@ -946,9 +960,8 @@ gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size) { const guint8 *data = NULL; - if (gst_adapter_available (wav->adapter) < 8) { + if (gst_adapter_available (wav->adapter) < 8) return FALSE; - } data = gst_adapter_peek (wav->adapter, 8); *tag = GST_READ_UINT32_LE (data); @@ -1027,14 +1040,12 @@ gst_wavparse_stream_headers (GstWavParse * wav) if (!gst_wavparse_peek_chunk (wav, &tag, &size)) return GST_FLOW_OK; - buf = gst_buffer_new (); - gst_buffer_ref (buf); gst_adapter_flush (wav->adapter, 8); wav->offset += 8; - GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size); - GST_BUFFER_SIZE (buf) = size; + + buf = gst_adapter_take_buffer (wav->adapter, size); } else { - if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad, + if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad, &wav->offset, &tag, &buf)) != GST_FLOW_OK) return res; } @@ -1055,14 +1066,15 @@ gst_wavparse_stream_headers (GstWavParse * wav) if (tag != GST_RIFF_TAG_fmt) goto invalid_wav; - if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra))) + 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_DATA (buf) = NULL; gst_buffer_unref (buf); + buf = NULL; } /* Note: gst_riff_create_audio_caps might need to fix values in @@ -1078,19 +1090,28 @@ gst_wavparse_stream_headers (GstWavParse * wav) wav->channels = header->channels; wav->blockalign = header->blockalign; wav->depth = header->size; - wav->bps = header->av_bps; + wav->av_bps = header->av_bps; g_free (header); if (wav->channels == 0) goto no_channels; - if (wav->bps == 0 && (wav->format == GST_RIFF_WAVE_FORMAT_MPEGL12 || - wav->format == GST_RIFF_WAVE_FORMAT_MPEGL3)) { - /* Note: ugly workaround for mp2/mp3 embedded in wav, that relies on the - * bitrate inside the mpeg stream */ - /* wav->bps = 1; */ - GST_INFO ("WAV file with bps==0 and format=mp2/3"); + /* 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; @@ -1105,11 +1126,17 @@ gst_wavparse_stream_headers (GstWavParse * wav) GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign); GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width); GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth); - GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps); + 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 @@ -1149,9 +1176,8 @@ gst_wavparse_stream_headers (GstWavParse * wav) 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 + /* 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 */ @@ -1235,13 +1261,18 @@ gst_wavparse_stream_headers (GstWavParse * wav) (guint64) wav->fact); GST_DEBUG_OBJECT (wav, "calculated bps : %d", wav->bps); } - if (wav->bps <= 0) - goto no_bitrate; - duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps); - GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT, - GST_TIME_ARGS (duration)); - gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration); + if (wav->bps > 0) { + duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps); + GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT, + GST_TIME_ARGS (duration)); + gst_segment_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 @@ -1252,12 +1283,13 @@ gst_wavparse_stream_headers (GstWavParse * wav) gst_event_replace (event_p, NULL); wav->state = GST_WAVPARSE_DATA; + return GST_FLOW_OK; /* ERROR */ invalid_wav: { - GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), + 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); @@ -1279,14 +1311,6 @@ no_channels: g_free (codec_name); return GST_FLOW_ERROR; } -no_bitrate: - { - GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), - ("Stream claims to have a bitrate of <= zero - invalid data")); - g_free (header); - g_free (codec_name); - return GST_FLOW_ERROR; - } no_bytes_per_sample: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), @@ -1311,7 +1335,6 @@ header_read_error: } } - /* * Read WAV file tag when streaming */ @@ -1323,10 +1346,10 @@ gst_wavparse_parse_stream_init (GstWavParse * wav) /* _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 (wav), tmp)) { + if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp)) return GST_FLOW_ERROR; - } wav->offset += 12; /* Go to next state */ @@ -1403,15 +1426,18 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) gst_pad_set_caps (wav->srcpad, wav->caps); gst_caps_replace (&wav->caps, NULL); - gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad); - gst_element_no_more_pads (GST_ELEMENT (wav)); + gst_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"); - gst_pad_push_event (wav->srcpad, wav->newsegment); - wav->newsegment = NULL; + 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 (wav), wav->srcpad, wav->tags); + gst_element_found_tags_for_pad (GST_ELEMENT_CAST (wav), wav->srcpad, + wav->tags); wav->tags = NULL; } } @@ -1424,7 +1450,7 @@ gst_wavparse_stream_data (GstWavParse * wav) GstBuffer *buf = NULL; GstFlowReturn res = GST_FLOW_OK; guint64 desired, obtained; - GstClockTime timestamp, next_timestamp; + GstClockTime timestamp, next_timestamp, duration; guint64 pos, nextpos; iterate_adapter: @@ -1478,7 +1504,7 @@ iterate_adapter: obtained = GST_BUFFER_SIZE (buf); - /* our positions */ + /* our positions in bytes */ pos = wav->offset - wav->datastart; nextpos = pos + obtained; @@ -1486,23 +1512,33 @@ iterate_adapter: GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample; GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample; - /* and timestamps, be carefull for overflows */ - timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps); - next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps); + if (wav->bps > 0) { + /* and timestamps if we have a bitrate, be carefull for overflows */ + timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps); + next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps); + duration = next_timestamp - timestamp; - GST_BUFFER_TIMESTAMP (buf) = timestamp; - GST_BUFFER_DURATION (buf) = next_timestamp - timestamp; + /* update current running segment position */ + gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp); + } 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); + } - /* update current running segment position */ - gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp); + 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 (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf)); + ", 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; @@ -1513,6 +1549,7 @@ iterate_adapter: wav->dataleft = 0; } wav->offset += obtained; + /* Iterate until need more data, so adapter size won't grow */ if (wav->streaming) { GST_LOG_OBJECT (wav, @@ -1520,32 +1557,13 @@ iterate_adapter: wav->end_offset); goto iterate_adapter; } - return res; /* ERROR */ found_eos: { GST_DEBUG_OBJECT (wav, "found EOS"); - /* we completed the segment */ - wav->segment_running = FALSE; - if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) { - GstClockTime stop; - - if ((stop = wav->segment.stop) == -1) - stop = wav->segment.duration; - - gst_element_post_message (GST_ELEMENT (wav), - gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME, - stop)); - } else { - if (G_UNLIKELY (wav->first)) { - wav->first = FALSE; - gst_wavparse_add_src_pad (wav, NULL); - } - gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); - } - return GST_FLOW_WRONG_STATE; + return GST_FLOW_UNEXPECTED; } pull_error: { @@ -1598,20 +1616,49 @@ gst_wavparse_loop (GstPad * pad) default: g_assert_not_reached (); } - return; /* ERRORS */ pause: - GST_LOG_OBJECT (wav, "pausing task %d", ret); - gst_pad_pause_task (wav->sinkpad); - if (GST_FLOW_IS_FATAL (ret)) { - /* for fatal errors we post an error message */ - GST_ELEMENT_ERROR (wav, STREAM, FAILED, - (_("Internal data stream error.")), - ("streaming stopped, reason %s", gst_flow_get_name (ret))); - if (wav->srcpad != NULL) - gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); + { + 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; } } @@ -1629,45 +1676,31 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf) case GST_WAVPARSE_START: GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK) - goto pause; + 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 pause; + goto done; if (!wav->got_fmt || wav->datastart == 0) break; wav->state = GST_WAVPARSE_DATA; - /* fall-through */ + /* fall-through */ case GST_WAVPARSE_DATA: if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) - goto pause; + goto done; break; default: g_assert_not_reached (); } - - return ret; - -pause: - GST_LOG_OBJECT (wav, "pausing task %d", ret); - gst_pad_pause_task (wav->sinkpad); - if (GST_FLOW_IS_FATAL (ret)) { - /* for fatal errors we post an error message */ - GST_ELEMENT_ERROR (wav, STREAM, FAILED, - (_("Internal data stream error.")), - ("streaming stopped, reason %s", gst_flow_get_name (ret))); - if (wav->srcpad != NULL) - gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); - } +done: return ret; } @@ -1697,8 +1730,10 @@ gst_wavparse_pad_convert (GstPad * pad, wavparse = GST_WAVPARSE (gst_pad_get_parent (pad)); - if (wavparse->bytes_per_sample == 0) - goto no_bytes_per_sample; + if (*dest_format == src_format) { + *dest_value = src_value; + return TRUE; + } if (wavparse->bps == 0) goto no_bps; @@ -1708,6 +1743,8 @@ gst_wavparse_pad_convert (GstPad * pad, 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 = @@ -1717,7 +1754,6 @@ gst_wavparse_pad_convert (GstPad * pad, res = FALSE; goto done; } - *dest_value -= *dest_value % wavparse->bytes_per_sample; break; case GST_FORMAT_DEFAULT: @@ -1764,14 +1800,6 @@ done: return res; /* ERRORS */ -no_bytes_per_sample: - { - GST_DEBUG_OBJECT (wavparse, - "bytes_per_sample 0, probably an mp3 - channels %d, width %d", - wavparse->channels, wavparse->width); - res = FALSE; - goto done; - } no_bps: { GST_DEBUG_OBJECT (wavparse, "bps 0, cannot convert"); @@ -1810,7 +1838,6 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query) gint64 curb; gint64 cur; GstFormat format; - gboolean res = TRUE; curb = wav->offset - wav->datastart; gst_query_parse_position (query, &format, NULL); @@ -1835,7 +1862,6 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query) gint64 endb; gint64 end; GstFormat format; - gboolean res = TRUE; endb = wav->datasize; gst_query_parse_duration (query, &format, NULL); @@ -1849,8 +1875,8 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query) gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb, &format, &end); } - } break; + } default: format = GST_FORMAT_BYTES; end = endb; @@ -1898,15 +1924,13 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) case GST_EVENT_SEEK: { res = gst_wavparse_perform_seek (wavparse, event); + gst_event_unref (event); break; } default: - res = FALSE; + res = gst_pad_push_event (wavparse->sinkpad, event); break; } - - gst_event_unref (event); - return res; } @@ -1974,17 +1998,10 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; - case GST_STATE_CHANGE_PAUSED_TO_READY:{ - GstEvent **event_p = &wav->seek_event; - + case GST_STATE_CHANGE_PAUSED_TO_READY: gst_wavparse_destroy_sourcepad (wav); - gst_event_replace (event_p, NULL); gst_wavparse_reset (wav); - if (wav->adapter) { - gst_adapter_clear (wav->adapter); - } break; - } case GST_STATE_CHANGE_READY_TO_NULL: break; default: @@ -1998,8 +2015,6 @@ plugin_init (GstPlugin * plugin) { gst_riff_init (); - GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser"); - return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY, GST_TYPE_WAVPARSE); } diff --git a/gst/wavparse/gstwavparse.h b/gst/wavparse/gstwavparse.h index b0a4dad..84debfb 100644 --- a/gst/wavparse/gstwavparse.h +++ b/gst/wavparse/gstwavparse.h @@ -76,13 +76,16 @@ struct _GstWavParse { /* useful audio data */ guint16 depth; - gint rate; + gint rate; guint16 channels; guint16 blockalign; guint16 width; - guint32 bps; + guint32 av_bps; guint32 fact; + /* real bps used or 0 when no bitrate is known */ + guint32 bps; + guint bytes_per_sample; /* position in data part */ -- 2.7.4