From 562de2f6e0599ec73a1ac531744aba83663ef3f6 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 26 Apr 2006 20:11:18 +0000 Subject: [PATCH] gst/wavparse/gstwavparse.*: correct partial implementation of push mode (from my last commit) Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (plugin_init): * gst/wavparse/gstwavparse.h: correct partial implementation of push mode (from my last commit) --- ChangeLog | 14 +++ gst/wavparse/gstwavparse.c | 220 ++++++++++++++++++++++++++------------------- gst/wavparse/gstwavparse.h | 5 +- 3 files changed, 144 insertions(+), 95 deletions(-) diff --git a/ChangeLog b/ChangeLog index faedfda..16ca76b 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,17 @@ +2006-04-26 Stefan Kost + + * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), + (gst_wavparse_class_init), (gst_wavparse_reset), + (gst_wavparse_init), (gst_wavparse_create_sourcepad), + (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues), + (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), + (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), + (gst_wavparse_stream_data), (gst_wavparse_loop), + (gst_wavparse_chain), (plugin_init): + * gst/wavparse/gstwavparse.h: + correct partial implementation of push mode + (from my last commit) + 2006-04-26 Wim Taymans * ext/esd/esdsink.c: diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 87fdd5a..3e96c5e 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -1,7 +1,7 @@ /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */ /* GStreamer * Copyright (C) <1999> Erik Walthinsen - * Copyright (C) <2006> Nokia Corporation. + * 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 @@ -32,11 +32,18 @@ * Example launch line * * - * gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink + * gst-launch filesrc location=sine.wav ! queue ! wavparse ! audioconvert ! alsasink * * Read a wav file and output to the soundcard using the ALSA element. The * wav file is assumed to contain raw uncompressed samples. * + * + * + * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! wavparse ! audioconvert ! alsasink + * + * Stream data from + * + * * * * Last reviewed on 2006-03-03 (0.10.3) @@ -84,6 +91,12 @@ static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event); static void gst_wavparse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); +static const GstElementDetails gst_wavparse_details = +GST_ELEMENT_DETAILS ("WAV audio demuxer", + "Codec/Demuxer/Audio", + "Parse a .wav file into raw audio", + "Erik Walthinsen "); + static GstStaticPadTemplate sink_template_factory = GST_STATIC_PAD_TEMPLATE ("wavparse_sink", GST_PAD_SINK, @@ -162,22 +175,13 @@ static void gst_wavparse_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstPadTemplate *templ; - static const GstElementDetails gst_wavparse_details = - GST_ELEMENT_DETAILS ("WAV audio demuxer", - "Codec/Demuxer/Audio", - "Parse a .wav file into raw audio", - "Erik Walthinsen "); - - gst_element_class_set_details (element_class, &gst_wavparse_details); /* register src pads */ - templ = gst_static_pad_template_get (&sink_template_factory); - gst_element_class_add_pad_template (element_class, templ); - gst_object_unref (templ); - templ = gst_static_pad_template_get (&src_template_factory); - gst_element_class_add_pad_template (element_class, templ); - gst_object_unref (templ); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_template_factory)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_template_factory)); + gst_element_class_set_details (element_class, &gst_wavparse_details); } static void @@ -196,8 +200,6 @@ gst_wavparse_class_init (GstWavParseClass * klass) gstelement_class->change_state = gst_wavparse_change_state; gstelement_class->send_event = gst_wavparse_send_event; - - GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser"); } @@ -234,6 +236,7 @@ gst_wavparse_reset (GstWavParse * wavparse) wavparse->datasize = 0; wavparse->datastart = 0; wavparse->got_fmt = FALSE; + wavparse->first = TRUE; if (wavparse->seek_event) gst_event_unref (wavparse->seek_event); @@ -258,6 +261,9 @@ gst_wavparse_init (GstWavParse * wavparse) gst_pad_set_chain_function (wavparse->sinkpad, GST_DEBUG_FUNCPTR (gst_wavparse_chain)); gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad); + + /* src, will be created later */ + wavparse->srcpad = NULL; } static void @@ -285,6 +291,8 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse) GST_DEBUG_FUNCPTR (gst_wavparse_pad_query)); gst_pad_set_event_function (wavparse->srcpad, GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event)); + + GST_DEBUG_OBJECT (wavparse, "srcpad created"); } static void @@ -301,6 +309,8 @@ gst_wavparse_get_property (GObject * object, } } + + #if 0 static void gst_wavparse_parse_adtl (GstWavParse * wavparse, int len) @@ -484,9 +494,7 @@ gst_wavparse_parse_adtl (GstWavParse * wavparse, int len) g_object_notify (G_OBJECT (wavparse), "metadata"); } -#endif -#if 0 static void gst_wavparse_parse_cues (GstWavParse * wavparse, int len) { @@ -549,49 +557,7 @@ gst_wavparse_parse_cues (GstWavParse * wavparse, int len) g_object_notify (G_OBJECT (wavparse), "metadata"); } -#endif - -static gboolean -gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf) -{ - guint32 doctype; - - if (!gst_riff_parse_file_header (element, buf, &doctype)) - return FALSE; - - if (doctype != GST_RIFF_RIFF_WAVE) - goto not_wav; - - return TRUE; - - /* ERRORS */ -not_wav: - { - GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL), - ("File is not an WAVE file: %" GST_FOURCC_FORMAT, - GST_FOURCC_ARGS (doctype))); - return FALSE; - } -} - -static GstFlowReturn -gst_wavparse_stream_init (GstWavParse * wav) -{ - GstFlowReturn res; - GstBuffer *buf = NULL; - - if ((res = gst_pad_pull_range (wav->sinkpad, - wav->offset, 12, &buf)) != GST_FLOW_OK) - return res; - else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf)) - return GST_FLOW_ERROR; - - wav->offset += 12; - - return GST_FLOW_OK; -} -#if 0 /* Read 'fmt ' header */ static gboolean gst_wavparse_fmt (GstWavParse * wav) @@ -749,6 +715,48 @@ gst_wavparse_other (GstWavParse * wav) } #endif + + +static gboolean +gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf) +{ + guint32 doctype; + + if (!gst_riff_parse_file_header (element, buf, &doctype)) + return FALSE; + + if (doctype != GST_RIFF_RIFF_WAVE) + goto not_wav; + + return TRUE; + + /* ERRORS */ +not_wav: + { + GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL), + ("File is not an WAVE file: %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (doctype))); + return FALSE; + } +} + +static GstFlowReturn +gst_wavparse_stream_init (GstWavParse * wav) +{ + GstFlowReturn res; + GstBuffer *buf = NULL; + + if ((res = gst_pad_pull_range (wav->sinkpad, + wav->offset, 12, &buf)) != GST_FLOW_OK) + return res; + else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf)) + return GST_FLOW_ERROR; + + wav->offset += 12; + + return GST_FLOW_OK; +} + /* This function is used to perform seeks on the element in * pull mode. * @@ -800,7 +808,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) flush = flags & GST_SEEK_FLAG_FLUSH; - if (flush) { + if (flush && wav->srcpad) { GST_DEBUG_OBJECT (wav, "sending flush start"); gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ()); } else { @@ -846,19 +854,21 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop)); /* 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, - * close the segment first based on the last_stop. */ - GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT - " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop); - - gst_pad_push_event (wav->srcpad, - gst_event_new_new_segment (TRUE, - wav->segment.rate, wav->segment.format, - wav->segment.start, wav->segment.last_stop, wav->segment.time)); + if (wav->srcpad) { + if (flush) { + GST_DEBUG_OBJECT (wav, "sending flush stop"); + gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ()); + } else if (wav->segment_running) { + /* we are running the current segment and doing a non-flushing seek, + * close the segment first based on the last_stop. */ + GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT + " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop); + + gst_pad_push_event (wav->srcpad, + gst_event_new_new_segment (TRUE, + wav->segment.rate, wav->segment.format, + wav->segment.start, wav->segment.last_stop, wav->segment.time)); + } } memcpy (&wav->segment, &seeksegment, sizeof (GstSegment)); @@ -1080,7 +1090,6 @@ gst_wavparse_stream_headers (GstWavParse * wav) /* loop headers until we get data */ while (!gotdata) { - if (wav->streaming) { if (!gst_wavparse_peek_chunk_info (wav, &tag, &size)) return GST_FLOW_OK; @@ -1322,7 +1331,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) #define MAX_BUFFER_SIZE 4096 static GstFlowReturn -gst_wavparse_stream_data (GstWavParse * wav, gboolean first) +gst_wavparse_stream_data (GstWavParse * wav) { GstBuffer *buf = NULL; GstFlowReturn res = GST_FLOW_OK; @@ -1331,8 +1340,9 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) guint64 pos, nextpos; iterate_adapter: - GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, - wav->offset, wav->end_offset); + GST_LOG_OBJECT (wav, + "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %" + G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft); /* Get the next n bytes and output them */ if (wav->dataleft == 0 || wav->dataleft < wav->blockalign) @@ -1350,8 +1360,12 @@ iterate_adapter: "from the sinkpad", desired); if (wav->streaming) { - if (gst_adapter_available (wav->adapter) < desired) + guint avail = gst_adapter_available (wav->adapter); + + if (avail < desired) { + GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail); return GST_FLOW_OK; + } buf = gst_buffer_new (); GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired); @@ -1362,14 +1376,15 @@ iterate_adapter: goto pull_error; } - obtained = GST_BUFFER_SIZE (buf); - /* 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 (first) { + if (G_UNLIKELY (wav->first)) { + wav->first = FALSE; gst_wavparse_add_src_pad (wav, buf); } + obtained = GST_BUFFER_SIZE (buf); + /* our positions */ pos = wav->offset - wav->datastart; nextpos = pos + obtained; @@ -1400,16 +1415,17 @@ iterate_adapter: if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK) goto push_error; } else { - GST_DEBUG ("Srcpad not linked!"); gst_buffer_unref (buf); + goto not_linked; } if (obtained < wav->dataleft) { wav->dataleft -= obtained; - wav->offset += obtained; + //wav->offset += obtained; } else { wav->dataleft = 0; } + wav->offset += obtained; /* Iterate until need more data, so adapter size won't grow */ if (wav->streaming) { GST_LOG_OBJECT (wav, @@ -1451,6 +1467,11 @@ push_error: GST_DEBUG_OBJECT (wav, "Error pushing on srcpad"); return res; } +not_linked: + { + GST_DEBUG_OBJECT (wav, "Srcpad not linked!"); + return GST_FLOW_ERROR; + } } static void @@ -1459,8 +1480,11 @@ gst_wavparse_loop (GstPad * pad) GstFlowReturn ret; GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + GST_LOG_OBJECT (wav, "process data"); + switch (wav->state) { case GST_WAVPARSE_START: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK) goto pause; @@ -1468,15 +1492,15 @@ gst_wavparse_loop (GstPad * pad) /* fall-through */ case GST_WAVPARSE_HEADER: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER"); if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) goto pause; wav->state = GST_WAVPARSE_DATA; - if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK) - goto pause; - break; + /* fall-through */ + case GST_WAVPARSE_DATA: - if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK) + if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) goto pause; break; default: @@ -1505,24 +1529,30 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf) GstFlowReturn ret; GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + GST_LOG_OBJECT (wav, "adapter_push %" G_GINT64_FORMAT " bytes", + GST_BUFFER_SIZE (buf)); + gst_adapter_push (wav->adapter, buf); switch (wav->state) { case GST_WAVPARSE_START: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START"); if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK) goto pause; + + wav->state = GST_WAVPARSE_HEADER; /* fall-through */ case GST_WAVPARSE_HEADER: + GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER"); if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK) goto pause; wav->state = GST_WAVPARSE_DATA; - if ((ret = gst_wavparse_stream_data (wav, TRUE)) != GST_FLOW_OK) - goto pause; - break; + /* fall-through */ + case GST_WAVPARSE_DATA: - if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK) + if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK) goto pause; break; default: @@ -1868,6 +1898,8 @@ 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 9dd32e5..a0b2717 100644 --- a/gst/wavparse/gstwavparse.h +++ b/gst/wavparse/gstwavparse.h @@ -1,6 +1,6 @@ /* GStreamer * Copyright (C) <1999> Erik Walthinsen - * Copyright (C) <2006> Nokia Corporation. + * 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 @@ -103,6 +103,9 @@ struct _GstWavParse { /* configured segment, start/stop expressed in time */ GstSegment segment; gboolean segment_running; + + /* for late pad configuration */ + gboolean first; }; struct _GstWavParseClass { -- 2.7.4