From 16b28a8eea303ed4cf7f158d2f6d4de45a54f666 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Fri, 28 Apr 2006 23:09:17 +0000 Subject: [PATCH] gst/wavparse/gstwavparse.*: Add push (streaming) mode to wavparse (fixes #337625) Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (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_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_loop), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: Add push (streaming) mode to wavparse (fixes #337625) --- ChangeLog | 18 ++ gst/wavparse/gstwavparse.c | 594 +++++++++++++++++++++++++++---------- gst/wavparse/gstwavparse.h | 10 + 3 files changed, 462 insertions(+), 160 deletions(-) diff --git a/ChangeLog b/ChangeLog index fbda9cb5ed..5fa1e16af7 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,21 @@ +2006-04-29 Stefan Kost + + * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), + (gst_wavparse_class_init), (gst_wavparse_dispose), + (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_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_loop), + (gst_wavparse_chain), (gst_wavparse_srcpad_event), + (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), + (gst_wavparse_change_state), (plugin_init): + * gst/wavparse/gstwavparse.h: + Add push (streaming) mode to wavparse (fixes #337625) + 2006-04-28 Thomas Vander Stichele * configure.ac: diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 27b2964ee7..e9ab8bd0aa 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 + * 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 @@ -31,11 +32,18 @@ * Example launch line * * - * gst-launch filesrc sine.wav ! wavparse ! audioconvert ! alsasink + * gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink * * Read a wav file and output to the soundcard using the ALSA element. The * wav file is assumed to contain raw uncompressed samples. * + * + * + * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink + * + * Stream data from + * + * * * * Last reviewed on 2006-03-03 (0.10.3) @@ -51,18 +59,24 @@ #include "gst/riff/riff-media.h" #include +#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); @@ -77,6 +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, @@ -155,22 +175,13 @@ static void gst_wavparse_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - GstPadTemplate *templ; - static 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 @@ -185,13 +196,28 @@ 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; - - 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 +235,12 @@ gst_wavparse_reset (GstWavParse * wavparse) wavparse->dataleft = 0; wavparse->datasize = 0; wavparse->datastart = 0; + wavparse->got_fmt = FALSE; + wavparse->first = TRUE; + + if (wavparse->seek_event) + gst_event_unref (wavparse->seek_event); + wavparse->seek_event = NULL; /* we keep the segment info in time */ gst_segment_init (&wavparse->segment, GST_FORMAT_TIME); @@ -226,7 +258,12 @@ 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); + + /* src, will be created later */ + wavparse->srcpad = NULL; } static void @@ -254,6 +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 @@ -270,6 +309,8 @@ gst_wavparse_get_property (GObject * object, } } + + #if 0 static void gst_wavparse_parse_adtl (GstWavParse * wavparse, int len) @@ -453,9 +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) { @@ -518,49 +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) @@ -718,6 +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. * @@ -740,7 +779,6 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) gboolean update; GstSegment seeksegment; - if (event) { GST_DEBUG_OBJECT (wav, "doing seek with event"); @@ -770,10 +808,12 @@ 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 + } else { gst_pad_pause_task (wav->sinkpad); + } GST_PAD_STREAM_LOCK (wav->sinkpad); @@ -814,18 +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_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); + 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)); + 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)); @@ -853,8 +896,10 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event) } wav->segment_running = TRUE; - gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop, - wav->sinkpad); + if (!wav->streaming) { + gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop, + wav->sinkpad); + } GST_PAD_STREAM_UNLOCK (wav->sinkpad); @@ -868,6 +913,59 @@ 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 +985,127 @@ 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); + + 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); } - 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 +1114,11 @@ 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 +1131,19 @@ 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 +1161,7 @@ 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 +1221,32 @@ 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 +1267,8 @@ 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 +1318,7 @@ 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; @@ -1161,7 +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; @@ -1169,8 +1339,10 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) GstClockTime timestamp, next_timestamp; guint64 pos, nextpos; - GST_LOG_OBJECT (wav, "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, - wav->offset, wav->end_offset); +iterate_adapter: + GST_LOG_OBJECT (wav, + "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %" + G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft); /* Get the next n bytes and output them */ if (wav->dataleft == 0 || wav->dataleft < wav->blockalign) @@ -1187,18 +1359,32 @@ 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) { + guint avail = gst_adapter_available (wav->adapter); - obtained = GST_BUFFER_SIZE (buf); + 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); + GST_BUFFER_SIZE (buf) = desired; + } else { + if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset, + desired, &buf)) != GST_FLOW_OK) + goto pull_error; + } /* first chunk of data? create the source pad. We do this only here so * we can detect broken .wav files with dts disguised as raw PCM (sigh) */ - if (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; @@ -1220,7 +1406,7 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) /* don't forget to set the caps on the buffer */ gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad)); - GST_DEBUG_OBJECT (wav, + 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)); @@ -1230,10 +1416,18 @@ gst_wavparse_stream_data (GstWavParse * wav, gboolean first) if (obtained < wav->dataleft) { wav->dataleft -= 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, + "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset, + wav->end_offset); + goto iterate_adapter; + } + return res; /* ERROR */ @@ -1275,8 +1469,11 @@ 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; @@ -1284,15 +1481,15 @@ 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: @@ -1315,6 +1512,58 @@ pause: } } +static GstFlowReturn +gst_wavparse_chain (GstPad * pad, GstBuffer * buf) +{ + GstFlowReturn ret; + GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad)); + + GST_LOG_OBJECT (wav, "adapter_push %" 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; + /* 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 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 +1775,8 @@ 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 +1801,35 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event) static gboolean gst_wavparse_sink_activate (GstPad * sinkpad) { - if (gst_pad_check_pull_range (sinkpad)) - return gst_pad_activate_pull (sinkpad, TRUE); + GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad)); + gboolean res; + + if (gst_pad_check_pull_range (sinkpad)) { + GST_DEBUG ("going to pull mode"); + wav->streaming = FALSE; + wav->adapter = NULL; + res = gst_pad_activate_pull (sinkpad, TRUE); + } else { + GST_DEBUG ("going to push (streaming) mode"); + wav->streaming = TRUE; + wav->adapter = gst_adapter_new (); + res = gst_pad_activate_push (sinkpad, TRUE); + } + gst_object_unref (wav); + return res; +} - /* 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 +1845,10 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition) GstStateChangeReturn ret; GstWavParse *wav = GST_WAVPARSE (element); + GST_DEBUG_OBJECT (wav, "changing state %s - %s", + gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)), + gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition))); + switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; @@ -1603,8 +1872,11 @@ 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: @@ -1618,6 +1890,8 @@ 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 2d14061929..a0b27176f2 100644 --- a/gst/wavparse/gstwavparse.h +++ b/gst/wavparse/gstwavparse.h @@ -1,5 +1,6 @@ /* 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 @@ -25,6 +26,7 @@ #include #include "gst/riff/riff-ids.h" #include "gst/riff/riff-read.h" +#include G_BEGIN_DECLS @@ -93,9 +95,17 @@ 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; + + /* for late pad configuration */ + gboolean first; }; struct _GstWavParseClass {