diff options
author | Stefan Kost <ensonic@users.sourceforge.net> | 2006-04-25 21:39:46 +0000 |
---|---|---|
committer | Stefan Kost <ensonic@users.sourceforge.net> | 2006-04-25 21:39:46 +0000 |
commit | 27f2c9b2555be2fda77179d16cd0c19f0ee37cfa (patch) | |
tree | 1a989bb0850065730f707065795f39911c7fe125 /gst/wavparse | |
parent | 55aed72d3cdd61e82bb2838f7462c4a2bd75d4f6 (diff) |
Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
Diffstat (limited to 'gst/wavparse')
-rw-r--r-- | gst/wavparse/gstwavparse.c | 402 | ||||
-rw-r--r-- | gst/wavparse/gstwavparse.h | 7 |
2 files changed, 333 insertions, 76 deletions
diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 27b2964e..87fdd5a4 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -1,6 +1,7 @@ /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */ /* GStreamer * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) <2006> Nokia Corporation. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -51,18 +52,24 @@ #include "gst/riff/riff-media.h" #include <gst/gst-i18n-plugin.h> +#ifndef G_MAXUINT32 +#define G_MAXUINT32 0xffffffff +#endif + GST_DEBUG_CATEGORY_STATIC (wavparse_debug); #define GST_CAT_DEFAULT (wavparse_debug) static void gst_wavparse_base_init (gpointer g_class); static void gst_wavparse_class_init (GstWavParseClass * klass); static void gst_wavparse_init (GstWavParse * wavparse); +static void gst_wavparse_dispose (GObject * object); static gboolean gst_wavparse_sink_activate (GstPad * sinkpad); static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active); static gboolean gst_wavparse_send_event (GstElement * element, GstEvent * event); +static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf); static GstStateChangeReturn gst_wavparse_change_state (GstElement * element, GstStateChange transition); @@ -156,7 +163,7 @@ gst_wavparse_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstPadTemplate *templ; - static GstElementDetails gst_wavparse_details = + static const GstElementDetails gst_wavparse_details = GST_ELEMENT_DETAILS ("WAV audio demuxer", "Codec/Demuxer/Audio", "Parse a .wav file into raw audio", @@ -185,6 +192,7 @@ 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; gstelement_class->send_event = gst_wavparse_send_event; @@ -192,6 +200,22 @@ gst_wavparse_class_init (GstWavParseClass * klass) GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser"); } + +static void +gst_wavparse_dispose (GObject * object) +{ + GST_DEBUG ("WAV: Dispose\n"); + GstWavParse *wav = GST_WAVPARSE (object); + + if (wav->adapter) { + g_object_unref (wav->adapter); + wav->adapter = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + + static void gst_wavparse_reset (GstWavParse * wavparse) { @@ -209,6 +233,11 @@ gst_wavparse_reset (GstWavParse * wavparse) wavparse->dataleft = 0; wavparse->datasize = 0; wavparse->datastart = 0; + wavparse->got_fmt = FALSE; + + 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); @@ -226,6 +255,8 @@ gst_wavparse_init (GstWavParse * wavparse) GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate)); gst_pad_set_activatepull_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull)); + gst_pad_set_chain_function (wavparse->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavparse_chain)); gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad); } @@ -740,7 +771,6 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) gboolean update; GstSegment seeksegment; - if (event) { GST_DEBUG_OBJECT (wav, "doing seek with event"); @@ -770,10 +800,12 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) flush = flags & GST_SEEK_FLAG_FLUSH; - if (flush) + if (flush) { + GST_DEBUG_OBJECT (wav, "sending flush start"); gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ()); - else + } else { gst_pad_pause_task (wav->sinkpad); + } GST_PAD_STREAM_LOCK (wav->sinkpad); @@ -815,6 +847,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) /* prepare for streaming again */ if (flush) { + GST_DEBUG_OBJECT (wav, "sending flush stop"); gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ()); } else if (wav->segment_running) { /* we are running the current segment and doing a non-flushing seek, @@ -868,6 +901,61 @@ no_format: } } + +/* + * gst_wavparse_peek_chunk_info: + * @wav Wavparse object + * @tag holder for tag + * @size holder for tag size + * + * Peek next chunk info (tag and size) + * + * Returns: %TRUE when one chunk info has been got from the adapter + */ +static gboolean +gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size) +{ + const guint8 *data = NULL; + + if (gst_adapter_available (wav->adapter) < 8) { + return FALSE; + } + + GST_DEBUG ("Next chunk size is %d bytes", *size); + data = gst_adapter_peek (wav->adapter, 8); + *tag = GST_READ_UINT32_LE (data); + *size = GST_READ_UINT32_LE (data + 4); + + return TRUE; +} + + +/* + * gst_wavparse_peek_chunk: + * @wav Wavparse object + * @tag holder for tag + * @size holder for tag size + * + * Peek enough data for one full chunk + * + * Returns: %TRUE when one chunk has been got + */ +static gboolean +gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size) +{ + guint32 peek_size = 0; + + gst_wavparse_peek_chunk_info (wav, tag, size); + GST_DEBUG ("Need to peek chunk of %d bytes", *size); + peek_size = (*size + 1) & ~1; + + if (gst_adapter_available (wav->adapter) >= (8 + peek_size)) { + return TRUE; + } else { + return FALSE; + } +} + static gboolean gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len) { @@ -887,97 +975,128 @@ static GstFlowReturn gst_wavparse_stream_headers (GstWavParse * wav) { GstFlowReturn res; - GstBuffer *buf, *extra; + GstBuffer *buf; gst_riff_strf_auds *header = NULL; - guint32 tag; + guint32 tag, size; gboolean gotdata = FALSE; GstCaps *caps; gint64 duration; gchar *codec_name = NULL; GstEvent **event_p; - /* The header start with a 'fmt ' tag */ - if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad, - &wav->offset, &tag, &buf)) != GST_FLOW_OK) - return res; + if (!wav->got_fmt) { + GstBuffer *extra; - else if (tag != GST_RIFF_TAG_fmt) - goto invalid_wav; + /* The header start with a 'fmt ' tag */ - if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra))) - goto parse_header_error; + if (wav->streaming) { + if (!gst_wavparse_peek_chunk (wav, &tag, &size)) + return GST_FLOW_OK; - /* Note: gst_riff_create_audio_caps might nedd to fix values in - * the header header depending on the format, so call it first */ - caps = - gst_riff_create_audio_caps (header->format, NULL, header, extra, - NULL, &codec_name); + 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; - if (extra) - gst_buffer_unref (extra); + } else { + if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad, + &wav->offset, &tag, &buf)) != GST_FLOW_OK) + return res; + } - wav->format = header->format; - wav->rate = header->rate; - wav->channels = header->channels; + if (tag != GST_RIFF_TAG_fmt) + goto invalid_wav; - if (wav->channels == 0) - goto no_channels; + if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra))) + goto parse_header_error; - wav->blockalign = header->blockalign; - wav->width = (header->blockalign * 8) / header->channels; - wav->depth = header->size; - wav->bps = header->av_bps; + if (extra) + gst_buffer_unref (extra); - if (wav->bps <= 0) - goto no_bitrate; + if (wav->streaming) { + gst_adapter_flush (wav->adapter, size); + wav->offset += size; + GST_BUFFER_DATA (buf) = NULL; + gst_buffer_unref (buf); + } - wav->bytes_per_sample = wav->channels * wav->width / 8; - if (wav->bytes_per_sample <= 0) - goto no_bytes_per_sample; + /* Note: gst_riff_create_audio_caps might nedd to fix values in + * the header header depending on the format, so call it first */ + caps = + gst_riff_create_audio_caps (header->format, NULL, header, NULL, + NULL, &codec_name); - g_free (header); + wav->format = header->format; + wav->rate = header->rate; + wav->channels = header->channels; - if (!caps) - goto unknown_format; + if (wav->channels == 0) + goto no_channels; - 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); + wav->blockalign = header->blockalign; + wav->width = (header->blockalign * 8) / header->channels; + wav->depth = header->size; + wav->bps = header->av_bps; - /* create pad later so we can sniff the first few bytes - * of the real data and correct our caps if necessary */ - gst_caps_replace (&wav->caps, caps); - gst_caps_replace (&caps, NULL); + if (wav->bps <= 0) + goto no_bitrate; - if (codec_name) { - wav->tags = gst_tag_list_new (); + wav->bytes_per_sample = wav->channels * wav->width / 8; + if (wav->bytes_per_sample <= 0) + goto no_bytes_per_sample; - gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, - GST_TAG_AUDIO_CODEC, codec_name, NULL); + g_free (header); - g_free (codec_name); - codec_name = NULL; - } + if (!caps) + goto unknown_format; + + GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign); + GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width); + GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth); + GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps); + + /* create pad later so we can sniff the first few bytes + * of the real data and correct our caps if necessary */ + gst_caps_replace (&wav->caps, caps); + gst_caps_replace (&caps, NULL); - GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate, wav->channels); + wav->got_fmt = TRUE; + + if (codec_name) { + wav->tags = gst_tag_list_new (); + + gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE, + GST_TAG_AUDIO_CODEC, codec_name, NULL); + + g_free (codec_name); + codec_name = NULL; + } + + GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate, + wav->channels); + } /* loop headers until we get data */ while (!gotdata) { - guint size; - guint32 tag; - if ((res = - gst_pad_pull_range (wav->sinkpad, wav->offset, 8, - &buf)) != GST_FLOW_OK) - goto header_read_error; + if (wav->streaming) { + if (!gst_wavparse_peek_chunk_info (wav, &tag, &size)) + return GST_FLOW_OK; + } else { + if ((res = + gst_pad_pull_range (wav->sinkpad, wav->offset, 8, + &buf)) != GST_FLOW_OK) + goto header_read_error; + tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf)); + size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4); + } /* wav is a st00pid format, we don't know for sure where data starts. So we have to go bit by bit until we find the 'data' header */ - tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf)); - size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4); switch (tag) { /* TODO : Implement the various cases */ @@ -986,6 +1105,11 @@ gst_wavparse_stream_headers (GstWavParse * wav) GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size); gotdata = TRUE; + if (wav->streaming) { + gst_adapter_flush (wav->adapter, 8); + } else { + gst_buffer_unref (buf); + } wav->offset += 8; wav->datastart = wav->offset; /* file might be truncated */ @@ -998,12 +1122,19 @@ gst_wavparse_stream_headers (GstWavParse * wav) break; } default: + if (wav->streaming) { + if (!gst_wavparse_peek_chunk (wav, &tag, &size)) + return GST_FLOW_OK; + } GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)); wav->offset += 8 + ((size + 1) & ~1); - break; + if (wav->streaming) { + gst_adapter_flush (wav->adapter, 8 + ((size + 1) & ~1)); + } else { + gst_buffer_unref (buf); + } } - gst_buffer_unref (buf); } GST_DEBUG_OBJECT (wav, "Finished parsing headers"); @@ -1021,6 +1152,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) event_p = &wav->seek_event; gst_event_replace (event_p, NULL); + wav->state = GST_WAVPARSE_DATA; return GST_FLOW_OK; /* ERROR */ @@ -1080,6 +1212,32 @@ header_read_error: } } + +/* + * Read WAV file tag when streaming + */ +static GstFlowReturn +gst_wavparse_parse_stream_init (GstWavParse * wav) +{ + if (gst_adapter_available (wav->adapter) >= 12) { + GstBuffer *tmp = gst_buffer_new (); + + /* _take flushes the data */ + GST_BUFFER_DATA (tmp) = gst_adapter_take (wav->adapter, 12); + GST_BUFFER_SIZE (tmp) = 12; + + GST_DEBUG ("Parsing wav header"); + if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) { + return GST_FLOW_ERROR; + } + + wav->offset += 12; + /* Go to next state */ + wav->state = GST_WAVPARSE_HEADER; + } + return GST_FLOW_OK; +} + /* handle an event sent directly to the element. * * This event can be sent either in the READY state or the @@ -1100,6 +1258,8 @@ gst_wavparse_send_event (GstElement * element, GstEvent * event) gboolean res = FALSE; GstEvent **event_p; + GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event)); + switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: if (wav->state == GST_WAVPARSE_DATA) { @@ -1149,6 +1309,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad); gst_element_no_more_pads (GST_ELEMENT (wav)); + GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad"); gst_pad_push_event (wav->srcpad, wav->newsegment); wav->newsegment = NULL; @@ -1169,6 +1330,7 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) GstClockTime timestamp, next_timestamp; guint64 pos, nextpos; +iterate_adapter: GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset, wav->end_offset); @@ -1187,9 +1349,18 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data " "from the sinkpad", desired); - if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, - desired, &buf)) != GST_FLOW_OK) - goto pull_error; + if (wav->streaming) { + if (gst_adapter_available (wav->adapter) < desired) + return GST_FLOW_OK; + + buf = gst_buffer_new (); + GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired); + GST_BUFFER_SIZE (buf) = desired; + } else { + if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, + desired, &buf)) != GST_FLOW_OK) + goto pull_error; + } obtained = GST_BUFFER_SIZE (buf); @@ -1225,8 +1396,13 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) ", size:%u", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf)); - if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK) - goto push_error; + if (gst_pad_is_linked (wav->srcpad)) { + if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK) + goto push_error; + } else { + GST_DEBUG ("Srcpad not linked!"); + gst_buffer_unref (buf); + } if (obtained < wav->dataleft) { wav->dataleft -= obtained; @@ -1234,6 +1410,14 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) } else { 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 */ @@ -1315,6 +1499,52 @@ pause: } } +static GstFlowReturn +gst_wavparse_chain (GstPad * pad, GstBuffer * buf) +{ + GstFlowReturn ret; + GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + + gst_adapter_push (wav->adapter, buf); + + switch (wav->state) { + case GST_WAVPARSE_START: + if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK) + goto pause; + /* fall-through */ + + case GST_WAVPARSE_HEADER: + if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) + goto pause; + + wav->state = GST_WAVPARSE_DATA; + if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK) + goto pause; + break; + case GST_WAVPARSE_DATA: + if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK) + goto pause; + break; + default: + g_assert_not_reached (); + } + + return ret; + +pause: + GST_LOG_OBJECT (wav, "pausing task %d", ret); + gst_pad_pause_task (wav->sinkpad); + if (GST_FLOW_IS_FATAL (ret)) { + /* for fatal errors we post an error message */ + GST_ELEMENT_ERROR (wav, STREAM, FAILED, + (_("Internal data stream error.")), + ("streaming stopped, reason %s", gst_flow_get_name (ret))); + if (wav->srcpad != NULL) + gst_pad_push_event (wav->srcpad, gst_event_new_eos ()); + } + return ret; +} + #if 0 /* convert and query stuff */ static const GstFormat * @@ -1526,7 +1756,8 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad)); gboolean res = TRUE; - GST_DEBUG_OBJECT (wavparse, "event %d", GST_EVENT_TYPE (event)); + GST_DEBUG_OBJECT (wavparse, "event %d, %s", GST_EVENT_TYPE (event), + GST_EVENT_TYPE_NAME (event)); /* can only handle events when we are in the data state */ if (wavparse->state != GST_WAVPARSE_DATA) @@ -1551,20 +1782,34 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) static gboolean gst_wavparse_sink_activate (GstPad * sinkpad) { - if (gst_pad_check_pull_range (sinkpad)) + GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad)); + + if (gst_pad_check_pull_range (sinkpad)) { + GST_DEBUG ("going to pull mode"); + wav->streaming = FALSE; + wav->adapter = NULL; + gst_object_unref (wav); return gst_pad_activate_pull (sinkpad, TRUE); + } else { + GST_DEBUG ("going to push (streaming) mode"); + wav->streaming = TRUE; + wav->adapter = gst_adapter_new (); + gst_object_unref (wav); + return gst_pad_activate_push (sinkpad, TRUE); + } +} - /* FIXME, we can only operate in pull mode for now */ - GST_DEBUG_OBJECT (sinkpad, "pull_range not supported on sinkpad"); - return FALSE; -}; static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active) { GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad)); + GST_DEBUG_OBJECT (wav, "activating pull"); + if (active) { + /* if we have a scheduler we can start the task */ + wav->segment_running = TRUE; gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop, sinkpad); } else { gst_pad_stop_task (sinkpad); @@ -1580,6 +1825,8 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition) GstStateChangeReturn ret; GstWavParse *wav = GST_WAVPARSE (element); + GST_DEBUG_OBJECT (wav, "chaning state"); + switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; @@ -1603,8 +1850,11 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition) 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: diff --git a/gst/wavparse/gstwavparse.h b/gst/wavparse/gstwavparse.h index 2d140619..9dd32e5b 100644 --- a/gst/wavparse/gstwavparse.h +++ b/gst/wavparse/gstwavparse.h @@ -1,5 +1,6 @@ /* GStreamer * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) <2006> Nokia Corporation. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -25,6 +26,7 @@ #include <gst/gst.h> #include "gst/riff/riff-ids.h" #include "gst/riff/riff-read.h" +#include <gst/base/gstadapter.h> G_BEGIN_DECLS @@ -93,6 +95,11 @@ struct _GstWavParse { /* pending seek */ GstEvent *seek_event; + /* For streaming */ + GstAdapter *adapter; + gboolean got_fmt; + gboolean streaming; + /* configured segment, start/stop expressed in time */ GstSegment segment; gboolean segment_running; |