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gstreamer/gst/wavparse/gstwavparse.c
Sebastian Dröge a82e38d607 wavparse: Actually clip to upstream size instead of size of the data chunk 
There might be other chunks after the data chunk, so clipping the chunk
size with the data size can lead to a negative number and all following
calculations go wrong and cause crashes or worse.

This was introduced in 3ac119bbe2.

https://bugzilla.gnome.org/show_bug.cgi?id=783760
2017-06-14 00:11:17 +03:00

2925 lines
86 KiB
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/* -*- 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, Stefan Kost <stefan.kost@nokia.com>.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-wavparse
*
* Parse a .wav file into raw or compressed audio.
*
* Wavparse supports both push and pull mode operations, making it possible to
* stream from a network source.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
* ]| Read a wav file and output to the soundcard using the ALSA element. The
* wav file is assumed to contain raw uncompressed samples.
* |[
* gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* ]| Stream data from a network url.
* </refsect2>
*/
/*
* TODO:
* http://replaygain.hydrogenaudio.org/file_format_wav.html
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include "gstwavparse.h"
#include "gst/riff/riff-media.h"
#include <gst/base/gsttypefindhelper.h>
#include <gst/pbutils/descriptions.h>
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
#define GST_CAT_DEFAULT (wavparse_debug)
/* Data size chunk of RF64,
* see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
#define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
static void gst_wavparse_dispose (GObject * object);
static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
GstObject * parent);
static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
GstObject * parent, GstPadMode mode, gboolean active);
static gboolean gst_wavparse_send_event (GstElement * element,
GstEvent * event);
static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static void gst_wavparse_loop (GstPad * pad);
static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static void gst_wavparse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavparse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
#define DEFAULT_IGNORE_LENGTH FALSE
enum
{
PROP_0,
PROP_IGNORE_LENGTH,
};
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wav")
);
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
#define gst_wavparse_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
DEBUG_INIT);
typedef struct
{
/* Offset Size Description Value
* 0x00 4 ID unique identification value
* 0x04 4 Position play order position
* 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
* 0x0c 4 Chunk Start Byte Offset of Data Chunk *
* 0x10 4 Block Start Byte Offset to sample of First Channel
* 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
*/
guint32 id;
guint32 position;
guint32 data_chunk_id;
guint32 chunk_start;
guint32 block_start;
guint32 sample_offset;
} GstWavParseCue;
typedef struct
{
/* Offset Size Description Value
* 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
* 0x0c Text
*/
guint32 cue_point_id;
gchar *text;
} GstWavParseLabl, GstWavParseNote;
static void
gst_wavparse_class_init (GstWavParseClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *object_class;
GstPadTemplate *src_template;
gstelement_class = (GstElementClass *) klass;
object_class = (GObjectClass *) klass;
parent_class = g_type_class_peek_parent (klass);
object_class->dispose = gst_wavparse_dispose;
object_class->set_property = gst_wavparse_set_property;
object_class->get_property = gst_wavparse_get_property;
/**
* GstWavParse:ignore-length:
*
* This selects whether the length found in a data chunk
* should be ignored. This may be useful for streamed audio
* where the length is unknown until the end of streaming,
* and various software/hardware just puts some random value
* in there and hopes it doesn't break too much.
*/
g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
g_param_spec_boolean ("ignore-length",
"Ignore length",
"Ignore length from the Wave header",
DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
/* register pads */
gst_element_class_add_static_pad_template (gstelement_class,
&sink_template_factory);
src_template = gst_pad_template_new ("src", GST_PAD_SRC,
GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
gst_element_class_add_pad_template (gstelement_class, src_template);
gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
}
static void
gst_wavparse_reset (GstWavParse * wav)
{
wav->state = GST_WAVPARSE_START;
/* These will all be set correctly in the fmt chunk */
wav->depth = 0;
wav->rate = 0;
wav->width = 0;
wav->channels = 0;
wav->blockalign = 0;
wav->bps = 0;
wav->fact = 0;
wav->offset = 0;
wav->end_offset = 0;
wav->dataleft = 0;
wav->datasize = 0;
wav->datastart = 0;
wav->duration = 0;
wav->got_fmt = FALSE;
wav->first = TRUE;
if (wav->seek_event)
gst_event_unref (wav->seek_event);
wav->seek_event = NULL;
if (wav->adapter) {
gst_adapter_clear (wav->adapter);
g_object_unref (wav->adapter);
wav->adapter = NULL;
}
if (wav->tags)
gst_tag_list_unref (wav->tags);
wav->tags = NULL;
if (wav->toc)
gst_toc_unref (wav->toc);
wav->toc = NULL;
if (wav->cues)
g_list_free_full (wav->cues, g_free);
wav->cues = NULL;
if (wav->labls)
g_list_free_full (wav->labls, g_free);
wav->labls = NULL;
if (wav->caps)
gst_caps_unref (wav->caps);
wav->caps = NULL;
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment = NULL;
}
static void
gst_wavparse_dispose (GObject * object)
{
GstWavParse *wav = GST_WAVPARSE (object);
GST_DEBUG_OBJECT (wav, "WAV: Dispose");
gst_wavparse_reset (wav);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wavparse_init (GstWavParse * wavparse)
{
gst_wavparse_reset (wavparse);
/* sink */
wavparse->sinkpad =
gst_pad_new_from_static_template (&sink_template_factory, "sink");
gst_pad_set_activate_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
gst_pad_set_activatemode_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_pad_set_event_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
/* src */
wavparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
gst_pad_use_fixed_caps (wavparse->srcpad);
gst_pad_set_query_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
gst_pad_set_event_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
}
static gboolean
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
{
guint32 doctype;
if (!gst_riff_parse_file_header (element, buf, &doctype))
return FALSE;
if (doctype != GST_RIFF_RIFF_WAVE)
goto not_wav;
return TRUE;
/* ERRORS */
not_wav:
{
GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
return FALSE;
}
}
static GstFlowReturn
gst_wavparse_stream_init (GstWavParse * wav)
{
GstFlowReturn res;
GstBuffer *buf = NULL;
if ((res = gst_pad_pull_range (wav->sinkpad,
wav->offset, 12, &buf)) != GST_FLOW_OK)
return res;
else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
return GST_FLOW_ERROR;
wav->offset += 12;
return GST_FLOW_OK;
}
static gboolean
gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
{
/* -1 always maps to -1 */
if (ts == -1) {
*bytepos = -1;
return TRUE;
}
/* 0 always maps to 0 */
if (ts == 0) {
*bytepos = 0;
return TRUE;
}
if (wav->bps > 0) {
*bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
return TRUE;
} else if (wav->fact) {
guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
*bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
return TRUE;
}
return FALSE;
}
/* This function is used to perform seeks on the element.
*
* It also works when event is NULL, in which case it will just
* start from the last configured segment. This technique is
* used when activating the element and to perform the seek in
* READY.
*/
static gboolean
gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
{
gboolean res;
gdouble rate;
GstFormat format, bformat;
GstSeekFlags flags;
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
gint64 cur, stop, upstream_size;
gboolean flush;
gboolean update;
GstSegment seeksegment = { 0, };
gint64 last_stop;
guint32 seqnum = 0;
if (event) {
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
seqnum = gst_event_get_seqnum (event);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
if (format != wav->segment.format) {
GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
gst_format_get_name (format),
gst_format_get_name (wav->segment.format));
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, cur,
wav->segment.format, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, stop,
wav->segment.format, &stop);
if (!res)
goto no_format;
format = wav->segment.format;
}
} else {
GST_DEBUG_OBJECT (wav, "doing seek without event");
flags = 0;
rate = 1.0;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* in push mode, we must delegate to upstream */
if (wav->streaming) {
gboolean res = FALSE;
/* if streaming not yet started; only prepare initial newsegment */
if (!event || wav->state != GST_WAVPARSE_DATA) {
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment = gst_event_new_segment (&wav->segment);
res = TRUE;
} else {
/* convert seek positions to byte positions in data sections */
if (format == GST_FORMAT_TIME) {
/* should not fail */
if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
goto no_position;
if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
goto no_position;
}
/* mind sample boundary and header */
if (cur >= 0) {
cur -= (cur % wav->bytes_per_sample);
cur += wav->datastart;
}
if (stop >= 0) {
stop -= (stop % wav->bytes_per_sample);
stop += wav->datastart;
}
GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
"start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
stop);
/* BYTE seek event */
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
stop_type, stop);
gst_event_set_seqnum (event, seqnum);
res = gst_pad_push_event (wav->sinkpad, event);
}
return res;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
GstEvent *fevent;
GST_DEBUG_OBJECT (wav, "sending flush start");
fevent = gst_event_new_flush_start ();
gst_event_set_seqnum (fevent, seqnum);
gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
gst_pad_push_event (wav->srcpad, fevent);
} else {
gst_pad_pause_task (wav->sinkpad);
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* save current position */
last_stop = wav->segment.position;
GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
gst_segment_do_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
(gint64 *) & wav->offset))
wav->offset = seeksegment.position;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset -= (wav->offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset += wav->datastart;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
} else {
GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
wav->offset);
}
if (stop_type != GST_SEEK_TYPE_NONE) {
if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
wav->end_offset = stop;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset += wav->datastart;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
} else {
GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
wav->end_offset);
}
/* make sure filesize is not exceeded due to rounding errors or so,
* same precaution as in _stream_headers */
bformat = GST_FORMAT_BYTES;
if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
wav->end_offset = MIN (wav->end_offset, upstream_size);
if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
wav->end_offset = wav->datastart + wav->datasize;
/* this is the range of bytes we will use for playback */
wav->offset = MIN (wav->offset, wav->end_offset);
wav->dataleft = wav->end_offset - wav->offset;
GST_DEBUG_OBJECT (wav,
"seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
if (flush) {
GstEvent *fevent;
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
GST_DEBUG_OBJECT (wav, "sending flush stop");
fevent = gst_event_new_flush_stop (TRUE);
gst_event_set_seqnum (fevent, seqnum);
gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
gst_pad_push_event (wav->srcpad, fevent);
}
/* now we did the seek and can activate the new segment values */
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_start (GST_OBJECT_CAST (wav),
wav->segment.format, wav->segment.position));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.position, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment = gst_event_new_segment (&wav->segment);
gst_event_set_seqnum (wav->start_segment, seqnum);
/* mark discont if we are going to stream from another position. */
if (last_stop != wav->segment.position) {
GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
wav->discont = TRUE;
}
/* and start the streaming task again */
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
wav->sinkpad, NULL);
}
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
return TRUE;
/* ERRORS */
negative_rate:
{
GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
return FALSE;
}
no_format:
{
GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
return FALSE;
}
no_position:
{
GST_DEBUG_OBJECT (wav,
"Could not determine byte position for desired time");
return FALSE;
}
}
/*
* gst_wavparse_peek_chunk_info:
* @wav Wavparse object
* @tag holder for tag
* @size holder for tag size
*
* Peek next chunk info (tag and size)
*
* Returns: %TRUE when the chunk info (header) is available
*/
static gboolean
gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
{
const guint8 *data = NULL;
if (gst_adapter_available (wav->adapter) < 8)
return FALSE;
data = gst_adapter_map (wav->adapter, 8);
*tag = GST_READ_UINT32_LE (data);
*size = GST_READ_UINT32_LE (data + 4);
gst_adapter_unmap (wav->adapter);
GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
GST_FOURCC_ARGS (*tag));
return TRUE;
}
/*
* gst_wavparse_peek_chunk:
* @wav Wavparse object
* @tag holder for tag
* @size holder for tag size
*
* Peek enough data for one full chunk
*
* Returns: %TRUE when the full chunk is available
*/
static gboolean
gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
{
guint32 peek_size = 0;
guint available;
if (!gst_wavparse_peek_chunk_info (wav, tag, size))
return FALSE;
/* size 0 -> empty data buffer would surprise most callers,
* large size -> do not bother trying to squeeze that into adapter,
* so we throw poor man's exception, which can be caught if caller really
* wants to handle 0 size chunk */
if (!(*size) || (*size) >= (1 << 30)) {
GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
*size, GST_FOURCC_ARGS (*tag));
/* chain should give up */
wav->abort_buffering = TRUE;
return FALSE;
}
peek_size = (*size + 1) & ~1;
available = gst_adapter_available (wav->adapter);
if (available >= (8 + peek_size)) {
return TRUE;
} else {
GST_LOG ("but only %u bytes available now", available);
return FALSE;
}
}
/*
* gst_wavparse_calculate_duration:
* @wav: wavparse object
*
* Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
* fallback.
*
* Returns: %TRUE if duration is available.
*/
static gboolean
gst_wavparse_calculate_duration (GstWavParse * wav)
{
if (wav->duration > 0)
return TRUE;
if (wav->bps > 0) {
GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
wav->duration =
gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
(guint64) wav->bps);
GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
} else if (wav->fact) {
wav->duration =
gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
}
return FALSE;
}
static gboolean
gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
guint32 size)
{
guint flush;
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
return FALSE;
}
GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (tag));
flush = 8 + ((size + 1) & ~1);
wav->offset += flush;
if (wav->streaming) {
gst_adapter_flush (wav->adapter, flush);
} else {
gst_buffer_unref (buf);
}
return TRUE;
}
/*
* gst_wavparse_cue_chunk:
* @wav GstWavParse object
* @data holder for data
* @size holder for data size
*
* Parse cue chunk from @data to wav->cues.
*
* Returns: %TRUE when cue chunk is available
*/
static gboolean
gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
{
guint32 i, ncues;
GList *cues = NULL;
GstWavParseCue *cue;
if (wav->cues) {
GST_WARNING_OBJECT (wav, "found another cue's");
return TRUE;
}
ncues = GST_READ_UINT32_LE (data);
if (size < 4 + ncues * 24) {
GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
return FALSE;
}
/* parse data */
data += 4;
for (i = 0; i < ncues; i++) {
cue = g_new0 (GstWavParseCue, 1);
cue->id = GST_READ_UINT32_LE (data);
cue->position = GST_READ_UINT32_LE (data + 4);
cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
cue->chunk_start = GST_READ_UINT32_LE (data + 12);
cue->block_start = GST_READ_UINT32_LE (data + 16);
cue->sample_offset = GST_READ_UINT32_LE (data + 20);
cues = g_list_append (cues, cue);
data += 24;
}
wav->cues = cues;
return TRUE;
}
/*
* gst_wavparse_labl_chunk:
* @wav GstWavParse object
* @data holder for data
* @size holder for data size
*
* Parse labl from @data to wav->labls.
*
* Returns: %TRUE when labl chunk is available
*/
static gboolean
gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
{
GstWavParseLabl *labl;
if (size < 5)
return FALSE;
labl = g_new0 (GstWavParseLabl, 1);
/* parse data */
data += 8;
labl->cue_point_id = GST_READ_UINT32_LE (data);
labl->text = g_memdup (data + 4, size - 4);
wav->labls = g_list_append (wav->labls, labl);
return TRUE;
}
/*
* gst_wavparse_note_chunk:
* @wav GstWavParse object
* @data holder for data
* @size holder for data size
*
* Parse note from @data to wav->notes.
*
* Returns: %TRUE when note chunk is available
*/
static gboolean
gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
{
GstWavParseNote *note;
if (size < 5)
return FALSE;
note = g_new0 (GstWavParseNote, 1);
/* parse data */
data += 8;
note->cue_point_id = GST_READ_UINT32_LE (data);
note->text = g_memdup (data + 4, size - 4);
wav->notes = g_list_append (wav->notes, note);
return TRUE;
}
/*
* gst_wavparse_smpl_chunk:
* @wav GstWavParse object
* @data holder for data
* @size holder for data size
*
* Parse smpl chunk from @data.
*
* Returns: %TRUE when cue chunk is available
*/
static gboolean
gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
{
guint32 note_number;
/*
manufacturer_id = GST_READ_UINT32_LE (data);
product_id = GST_READ_UINT32_LE (data + 4);
sample_period = GST_READ_UINT32_LE (data + 8);
*/
note_number = GST_READ_UINT32_LE (data + 12);
/*
pitch_fraction = GST_READ_UINT32_LE (data + 16);
SMPTE_format = GST_READ_UINT32_LE (data + 20);
SMPTE_offset = GST_READ_UINT32_LE (data + 24);
num_sample_loops = GST_READ_UINT32_LE (data + 28);
List of Sample Loops, 24 bytes each
*/
if (!wav->tags)
wav->tags = gst_tag_list_new_empty ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
return TRUE;
}
/*
* gst_wavparse_adtl_chunk:
* @wav GstWavParse object
* @data holder for data
* @size holder for data size
*
* Parse adtl from @data.
*
* Returns: %TRUE when adtl chunk is available
*/
static gboolean
gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
{
guint32 ltag, lsize, offset = 0;
while (size >= 8) {
ltag = GST_READ_UINT32_LE (data + offset);
lsize = GST_READ_UINT32_LE (data + offset + 4);
if (lsize + 8 > size) {
GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
return FALSE;
}
switch (ltag) {
case GST_RIFF_TAG_labl:
gst_wavparse_labl_chunk (wav, data + offset, size);
break;
case GST_RIFF_TAG_note:
gst_wavparse_note_chunk (wav, data + offset, size);
break;
default:
GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (ltag));
GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
break;
}
offset += 8 + GST_ROUND_UP_2 (lsize);
size -= 8 + GST_ROUND_UP_2 (lsize);
}
return TRUE;
}
static GstTagList *
gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
{
GstTagList *tags = NULL;
GstTocEntry *entry = NULL;
entry = gst_toc_find_entry (toc, id);
if (entry != NULL) {
tags = gst_toc_entry_get_tags (entry);
if (tags == NULL) {
tags = gst_tag_list_new_empty ();
gst_toc_entry_set_tags (entry, tags);
}
}
return tags;
}
/*
* gst_wavparse_create_toc:
* @wav GstWavParse object
*
* Create TOC from wav->cues and wav->labls.
*/
static gboolean
gst_wavparse_create_toc (GstWavParse * wav)
{
gint64 start, stop;
gchar *id;
GList *list;
GstWavParseCue *cue;
GstWavParseLabl *labl;
GstWavParseNote *note;
GstTagList *tags;
GstToc *toc;
GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
GST_OBJECT_LOCK (wav);
if (wav->toc) {
GST_OBJECT_UNLOCK (wav);
GST_WARNING_OBJECT (wav, "found another TOC");
return FALSE;
}
if (!wav->cues) {
GST_OBJECT_UNLOCK (wav);
return TRUE;
}
/* FIXME: send CURRENT scope toc too */
toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
/* add cue edition */
entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
gst_toc_append_entry (toc, entry);
/* add tracks in cue edition */
list = wav->cues;
while (list) {
cue = list->data;
prev_subentry = cur_subentry;
/* previous track stop time = current track start time */
if (prev_subentry != NULL) {
gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
}
id = g_strdup_printf ("%08x", cue->id);
cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
g_free (id);
start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
stop = wav->duration;
gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
gst_toc_entry_append_sub_entry (entry, cur_subentry);
list = g_list_next (list);
}
/* add tags in tracks */
list = wav->labls;
while (list) {
labl = list->data;
id = g_strdup_printf ("%08x", labl->cue_point_id);
tags = gst_wavparse_get_tags_toc_entry (toc, id);
g_free (id);
if (tags != NULL) {
gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
NULL);
}
list = g_list_next (list);
}
list = wav->notes;
while (list) {
note = list->data;
id = g_strdup_printf ("%08x", note->cue_point_id);
tags = gst_wavparse_get_tags_toc_entry (toc, id);
g_free (id);
if (tags != NULL) {
gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
note->text, NULL);
}
list = g_list_next (list);
}
/* send data as TOC */
wav->toc = toc;
/* send TOC event */
if (wav->toc) {
GST_OBJECT_UNLOCK (wav);
gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
}
return TRUE;
}
#define MAX_BUFFER_SIZE 4096
static gboolean
parse_ds64 (GstWavParse * wav, GstBuffer * buf)
{
GstMapInfo map;
guint32 dataSizeLow, dataSizeHigh;
guint32 sampleCountLow, sampleCountHigh;
gst_buffer_map (buf, &map, GST_MAP_READ);
dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
gst_buffer_unmap (buf, &map);
if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
}
if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
}
GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
" fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
return TRUE;
}
static GstFlowReturn
gst_wavparse_stream_headers (GstWavParse * wav)
{
GstFlowReturn res = GST_FLOW_OK;
GstBuffer *buf = NULL;
gst_riff_strf_auds *header = NULL;
guint32 tag, size;
gboolean gotdata = FALSE;
GstCaps *caps = NULL;
gchar *codec_name = NULL;
gint64 upstream_size = 0;
GstStructure *s;
/* search for "_fmt" chunk, which must be before "data" */
while (!wav->got_fmt) {
GstBuffer *extra;
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
return res;
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
if (size) {
buf = gst_adapter_take_buffer (wav->adapter, size);
if (size & 1)
gst_adapter_flush (wav->adapter, 1);
wav->offset += GST_ROUND_UP_2 (size);
} else {
buf = gst_buffer_new ();
}
} else {
if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
&wav->offset, &tag, &buf)) != GST_FLOW_OK)
return res;
}
if (tag == GST_RS64_TAG_DS64) {
if (!parse_ds64 (wav, buf))
goto fail;
else
continue;
}
if (tag != GST_RIFF_TAG_fmt) {
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
GST_FOURCC_ARGS (tag));
gst_buffer_unref (buf);
buf = NULL;
continue;
}
if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
&extra)))
goto parse_header_error;
buf = NULL; /* parse_strf_auds() took ownership of buffer */
/* do sanity checks of header fields */
if (header->channels == 0)
goto no_channels;
if (header->rate == 0)
goto no_rate;
GST_DEBUG_OBJECT (wav, "creating the caps");
/* Note: gst_riff_create_audio_caps might need to fix values in
* the header header depending on the format, so call it first */
/* FIXME: Need to handle the channel reorder map */
caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
NULL, &codec_name, NULL);
if (extra)
gst_buffer_unref (extra);
if (!caps)
goto unknown_format;
/* If we got raw audio from upstream, we remove the codec_data field,
* which may have been added if the wav header included an extended
* chunk. We want to keep it for non raw audio.
*/
s = gst_caps_get_structure (caps, 0);
if (s && gst_structure_has_name (s, "audio/x-raw")) {
gst_structure_remove_field (s, "codec_data");
}
/* do more sanity checks of header fields
* (these can be sanitized by gst_riff_create_audio_caps()
*/
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
wav->blockalign = header->blockalign;
wav->depth = header->bits_per_sample;
wav->av_bps = header->av_bps;
wav->vbr = FALSE;
g_free (header);
header = NULL;
/* do format specific handling */
switch (wav->format) {
case GST_RIFF_WAVE_FORMAT_MPEGL12:
case GST_RIFF_WAVE_FORMAT_MPEGL3:
{
/* Note: workaround for mp2/mp3 embedded in wav, that relies on the
* bitrate inside the mpeg stream */
GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
wav->bps = 0;
break;
}
case GST_RIFF_WAVE_FORMAT_PCM:
if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
goto invalid_blockalign;
/* fall through */
default:
if (wav->av_bps > wav->blockalign * wav->rate)
goto invalid_bps;
/* use the configured bps */
wav->bps = wav->av_bps;
break;
}
wav->width = (wav->blockalign * 8) / wav->channels;
wav->bytes_per_sample = wav->channels * wav->width / 8;
if (wav->bytes_per_sample <= 0)
goto no_bytes_per_sample;
GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
/* bps can be 0 when we don't have a valid bitrate (mostly for compressed
* formats). This will make the element output a BYTE format segment and
* will not timestamp the outgoing buffers.
*/
GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
/* create pad later so we can sniff the first few bytes
* of the real data and correct our caps if necessary */
gst_caps_replace (&wav->caps, caps);
gst_caps_replace (&caps, NULL);
wav->got_fmt = TRUE;
if (wav->tags == NULL)
wav->tags = gst_tag_list_new_empty ();
{
GstCaps *templ_caps = gst_pad_get_pad_template_caps (wav->sinkpad);
gst_pb_utils_add_codec_description_to_tag_list (wav->tags,
GST_TAG_CONTAINER_FORMAT, templ_caps);
gst_caps_unref (templ_caps);
}
/* If bps is nonzero, then we do have a valid bitrate that can be
* announced in a tag list. */
if (wav->bps) {
guint bitrate = wav->bps * 8;
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, bitrate, NULL);
}
if (codec_name) {
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, codec_name, NULL);
g_free (codec_name);
codec_name = NULL;
}
}
gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
/* loop headers until we get data */
while (!gotdata) {
if (wav->streaming) {
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
goto exit;
} else {
GstMapInfo map;
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_map (buf, &map, GST_MAP_READ);
tag = GST_READ_UINT32_LE (map.data);
size = GST_READ_UINT32_LE (map.data + 4);
gst_buffer_unmap (buf, &map);
}
GST_INFO_OBJECT (wav,
"Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
/* Maximum valid size is INT_MAX */
if (size & 0x80000000) {
GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
size = 0x7fffffff;
}
/* Clip to upstream size if known */
if (upstream_size > 0 && size + wav->offset > upstream_size) {
GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
g_assert (upstream_size >= wav->offset);
size = upstream_size - wav->offset;
}
/* wav is a st00pid format, we don't know for sure where data starts.
* So we have to go bit by bit until we find the 'data' header
*/
switch (tag) {
case GST_RIFF_TAG_data:{
guint64 size64;
GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
size64 = size;
if (wav->ignore_length) {
GST_DEBUG_OBJECT (wav, "Ignoring length");
size64 = 0;
}
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8);
gotdata = TRUE;
} else {
gst_buffer_unref (buf);
}
wav->offset += 8;
wav->datastart = wav->offset;
/* use size from ds64 chunk if available */
if (size64 == -1 && wav->datasize > 0) {
GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
size64 = wav->datasize;
}
/* If size is zero, then the data chunk probably actually extends to
the end of the file */
if (size64 == 0 && upstream_size) {
size64 = upstream_size - wav->datastart;
}
/* Or the file might be truncated */
else if (upstream_size) {
size64 = MIN (size64, (upstream_size - wav->datastart));
}
wav->datasize = size64;
wav->dataleft = size64;
wav->end_offset = size64 + wav->datastart;
if (!wav->streaming) {
/* We will continue parsing tags 'till end */
wav->offset += size64;
}
GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
break;
}
case GST_RIFF_TAG_fact:{
if (wav->fact == 0 &&
wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
const guint data_size = 4;
GST_INFO_OBJECT (wav, "Have fact chunk");
if (size < data_size) {
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
/* need more data */
goto exit;
}
GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
data_size, size);
break;
}
/* number of samples (for compressed formats) */
if (wav->streaming) {
const guint8 *data = NULL;
if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
data = gst_adapter_map (wav->adapter, data_size);
wav->fact = GST_READ_UINT32_LE (data);
gst_adapter_unmap (wav->adapter);
gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
} else {
gst_buffer_unref (buf);
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_extract (buf, 0, &wav->fact, 4);
wav->fact = GUINT32_FROM_LE (wav->fact);
gst_buffer_unref (buf);
}
GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
wav->offset += 8 + GST_ROUND_UP_2 (size);
break;
} else {
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
/* need more data */
goto exit;
}
}
break;
}
case GST_RIFF_TAG_acid:{
const gst_riff_acid *acid = NULL;
const guint data_size = sizeof (gst_riff_acid);
gfloat tempo;
GST_INFO_OBJECT (wav, "Have acid chunk");
if (size < data_size) {
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
/* need more data */
goto exit;
}
GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
data_size, size);
break;
}
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
data_size);
tempo = acid->tempo;
gst_adapter_unmap (wav->adapter);
} else {
GstMapInfo map;
gst_buffer_unref (buf);
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
size, &buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_map (buf, &map, GST_MAP_READ);
acid = (const gst_riff_acid *) map.data;
tempo = acid->tempo;
gst_buffer_unmap (buf, &map);
}
/* send data as tags */
if (!wav->tags)
wav->tags = gst_tag_list_new_empty ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
size = GST_ROUND_UP_2 (size);
if (wav->streaming) {
gst_adapter_flush (wav->adapter, size);
} else {
gst_buffer_unref (buf);
}
wav->offset += 8 + size;
break;
}
/* FIXME: all list tags after data are ignored in streaming mode */
case GST_RIFF_TAG_LIST:{
guint32 ltag;
if (wav->streaming) {
const guint8 *data = NULL;
if (gst_adapter_available (wav->adapter) < 12) {
goto exit;
}
data = gst_adapter_map (wav->adapter, 12);
ltag = GST_READ_UINT32_LE (data + 8);
gst_adapter_unmap (wav->adapter);
} else {
gst_buffer_unref (buf);
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
&buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_extract (buf, 8, &ltag, 4);
ltag = GUINT32_FROM_LE (ltag);
}
switch (ltag) {
case GST_RIFF_LIST_INFO:{
const gint data_size = size - 4;
GstTagList *new;
GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
if (wav->streaming) {
if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
goto exit;
}
gst_adapter_flush (wav->adapter, 12);
wav->offset += 12;
if (data_size > 0) {
buf = gst_adapter_take_buffer (wav->adapter, data_size);
if (data_size & 1)
gst_adapter_flush (wav->adapter, 1);
}
} else {
wav->offset += 12;
gst_buffer_unref (buf);
buf = NULL;
if (data_size > 0) {
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
}
}
if (data_size > 0) {
/* parse tags */
gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
if (new) {
GstTagList *old = wav->tags;
wav->tags =
gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
if (old)
gst_tag_list_unref (old);
gst_tag_list_unref (new);
}
gst_buffer_unref (buf);
wav->offset += GST_ROUND_UP_2 (data_size);
}
break;
}
case GST_RIFF_LIST_adtl:{
const gint data_size = size - 4;
GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
if (wav->streaming) {
const guint8 *data = NULL;
gst_adapter_flush (wav->adapter, 12);
wav->offset += 12;
data = gst_adapter_map (wav->adapter, data_size);
gst_wavparse_adtl_chunk (wav, data, data_size);
gst_adapter_unmap (wav->adapter);
} else {
GstMapInfo map;
gst_buffer_unref (buf);
buf = NULL;
wav->offset += 12;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
data_size);
gst_buffer_unmap (buf, &map);
}
wav->offset += GST_ROUND_UP_2 (data_size);
break;
}
default:
GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (ltag));
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
/* need more data */
goto exit;
break;
}
break;
}
case GST_RIFF_TAG_cue:{
const guint data_size = size;
GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
if (wav->streaming) {
const guint8 *data = NULL;
if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
data = gst_adapter_map (wav->adapter, data_size);
if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
goto header_read_error;
}
gst_adapter_unmap (wav->adapter);
} else {
GstMapInfo map;
wav->offset += 8;
gst_buffer_unref (buf);
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
data_size)) {
goto header_read_error;
}
gst_buffer_unmap (buf, &map);
}
size = GST_ROUND_UP_2 (size);
if (wav->streaming) {
gst_adapter_flush (wav->adapter, size);
} else {
gst_buffer_unref (buf);
}
size = GST_ROUND_UP_2 (size);
wav->offset += size;
break;
}
case GST_RIFF_TAG_smpl:{
const gint data_size = size;
GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
if (wav->streaming) {
const guint8 *data = NULL;
if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
data = gst_adapter_map (wav->adapter, data_size);
if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
goto header_read_error;
}
gst_adapter_unmap (wav->adapter);
} else {
GstMapInfo map;
wav->offset += 8;
gst_buffer_unref (buf);
buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
data_size)) {
goto header_read_error;
}
gst_buffer_unmap (buf, &map);
}
size = GST_ROUND_UP_2 (size);
if (wav->streaming) {
gst_adapter_flush (wav->adapter, size);
} else {
gst_buffer_unref (buf);
}
size = GST_ROUND_UP_2 (size);
wav->offset += size;
break;
}
default:
GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (tag));
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
/* need more data */
goto exit;
break;
}
if (upstream_size && (wav->offset >= upstream_size)) {
/* Now we are gone through the whole file */
gotdata = TRUE;
}
}
GST_DEBUG_OBJECT (wav, "Finished parsing headers");
if (wav->bps <= 0 && wav->fact) {
#if 0
/* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
wav->bps =
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
(guint64) wav->fact);
GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
#endif
wav->vbr = TRUE;
}
if (gst_wavparse_calculate_duration (wav)) {
gst_segment_init (&wav->segment, GST_FORMAT_TIME);
if (!wav->ignore_length)
wav->segment.duration = wav->duration;
if (!wav->toc)
gst_wavparse_create_toc (wav);
} else {
/* no bitrate, let downstream peer do the math, we'll feed it bytes. */
gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
if (!wav->ignore_length)
wav->segment.duration = wav->datasize;
}
/* now we have all the info to perform a pending seek if any, if no
* event, this will still do the right thing and it will also send
* the right newsegment event downstream. */
gst_wavparse_perform_seek (wav, wav->seek_event);
/* remove pending event */
gst_event_replace (&wav->seek_event, NULL);
/* we just started, we are discont */
wav->discont = TRUE;
wav->state = GST_WAVPARSE_DATA;
/* determine reasonable max buffer size,
* that is, buffers not too small either size or time wise
* so we do not end up with too many of them */
/* var abuse */
if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
wav->max_buf_size = upstream_size;
else
wav->max_buf_size = 0;
wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
if (wav->blockalign > 0)
wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
return GST_FLOW_OK;
/* ERROR */
exit:
{
g_free (codec_name);
g_free (header);
if (caps)
gst_caps_unref (caps);
return res;
}
fail:
{
res = GST_FLOW_ERROR;
goto exit;
}
parse_header_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
("Couldn't parse audio header"));
goto fail;
}
no_channels:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to contain no channels - invalid data"));
goto fail;
}
no_rate:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream with sample_rate == 0 - invalid data"));
goto fail;
}
invalid_blockalign:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims blockalign = %u, which is more than %u - invalid data",
wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
goto fail;
}
invalid_bps:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims av_bsp = %u, which is more than %u - invalid data",
wav->av_bps, wav->blockalign * wav->rate));
goto fail;
}
no_bytes_per_sample:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Could not caluclate bytes per sample - invalid data"));
goto fail;
}
unknown_format:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
("No caps found for format 0x%x, %u channels, %u Hz",
wav->format, wav->channels, wav->rate));
goto fail;
}
header_read_error:
{
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
goto fail;
}
}
/*
* Read WAV file tag when streaming
*/
static GstFlowReturn
gst_wavparse_parse_stream_init (GstWavParse * wav)
{
if (gst_adapter_available (wav->adapter) >= 12) {
GstBuffer *tmp;
/* _take flushes the data */
tmp = gst_adapter_take_buffer (wav->adapter, 12);
GST_DEBUG ("Parsing wav header");
if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
return GST_FLOW_ERROR;
wav->offset += 12;
/* Go to next state */
wav->state = GST_WAVPARSE_HEADER;
}
return GST_FLOW_OK;
}
/* handle an event sent directly to the element.
*
* This event can be sent either in the READY state or the
* >READY state. The only event of interest really is the seek
* event.
*
* In the READY state we can only store the event and try to
* respect it when going to PAUSED. We assume we are in the
* READY state when our parsing state != GST_WAVPARSE_DATA.
*
* When we are steaming, we can simply perform the seek right
* away.
*/
static gboolean
gst_wavparse_send_event (GstElement * element, GstEvent * event)
{
GstWavParse *wav = GST_WAVPARSE (element);
gboolean res = FALSE;
GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
if (wav->state == GST_WAVPARSE_DATA) {
/* we can handle the seek directly when streaming data */
res = gst_wavparse_perform_seek (wav, event);
} else {
GST_DEBUG_OBJECT (wav, "queuing seek for later");
gst_event_replace (&wav->seek_event, event);
/* we always return true */
res = TRUE;
}
break;
default:
break;
}
gst_event_unref (event);
return res;
}
static gboolean
gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
{
GstStructure *s;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_name (s, "audio/x-dts"))
return FALSE;
/* typefind behavior for DTS:
* MAXIMUM: multiple frame syncs detected, certainly DTS
* LIKELY: single frame sync at offset 0. Maybe DTS?
* POSSIBLE: single frame sync, not at offset 0. Highly unlikely
* to be DTS. */
if (prob > GST_TYPE_FIND_LIKELY)
return TRUE;
if (prob <= GST_TYPE_FIND_POSSIBLE)
return FALSE;
/* for maybe, check for at least a valid-looking rate and channels */
if (!gst_structure_has_field (s, "channels"))
return FALSE;
/* and for extra assurance we could also check the rate from the DTS frame
* against the one in the wav header, but for now let's not do that */
return gst_structure_has_field (s, "rate");
}
static GstTagList *
gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
{
GstTagList *tags = NULL;
GstEvent *ev;
gint i;
i = 0;
while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
gst_event_parse_tag (ev, &tags);
if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
tags = gst_tag_list_copy (tags);
gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
gst_event_unref (ev);
break;
}
tags = NULL;
gst_event_unref (ev);
}
return tags;
}
static void
gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
{
GstStructure *s;
GstTagList *tags, *utags;
GST_DEBUG_OBJECT (wav, "adding src pad");
g_assert (wav->caps != NULL);
s = gst_caps_get_structure (wav->caps, 0);
if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
GstTypeFindProbability prob;
GstCaps *tf_caps;
tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
if (tf_caps != NULL) {
GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
gst_caps_unref (wav->caps);
wav->caps = tf_caps;
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "dts", NULL);
} else {
GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
"marked as raw PCM audio, but ignoring for now", tf_caps);
gst_caps_unref (tf_caps);
}
}
}
gst_pad_set_caps (wav->srcpad, wav->caps);
if (wav->start_segment) {
GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
}
/* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
* that there'll be only one scope/type of tag list from upstream, if any */
utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
if (utags == NULL)
utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
/* if there's a tag upstream it's probably been added to override the
* tags from inside the wav header, so keep upstream tags if in doubt */
tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
if (wav->tags != NULL) {
gst_tag_list_unref (wav->tags);
wav->tags = NULL;
}
if (utags != NULL)
gst_tag_list_unref (utags);
/* send tags downstream, if any */
if (tags != NULL)
gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
}
static GstFlowReturn
gst_wavparse_stream_data (GstWavParse * wav)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (wav,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
/* Get the next n bytes and output them */
if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
goto found_eos;
/* scale the amount of data by the segment rate so we get equal
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (wav->dataleft),
wav->max_buf_size * ABS (wav->segment.rate));
if (desired >= wav->blockalign && wav->blockalign > 0)
desired -= (desired % wav->blockalign);
GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
"from the sinkpad", desired);
if (wav->streaming) {
guint avail = gst_adapter_available (wav->adapter);
guint extra;
/* flush some bytes if evil upstream sends segment that starts
* before data or does is not send sample aligned segment */
if (G_LIKELY (wav->offset >= wav->datastart)) {
extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
} else {
extra = wav->datastart - wav->offset;
}
if (G_UNLIKELY (extra)) {
extra = wav->bytes_per_sample - extra;
if (extra <= avail) {
GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
gst_adapter_flush (wav->adapter, extra);
wav->offset += extra;
wav->dataleft -= extra;
goto iterate_adapter;
} else {
GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
gst_adapter_clear (wav->adapter);
wav->offset += avail;
wav->dataleft -= avail;
return GST_FLOW_OK;
}
}
if (avail < desired) {
GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
return GST_FLOW_OK;
}
buf = gst_adapter_take_buffer (wav->adapter, desired);
} else {
if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
/* we may get a short buffer at the end of the file */
if (gst_buffer_get_size (buf) < desired) {
gsize size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
if (size >= wav->blockalign) {
if (wav->blockalign > 0) {
buf = gst_buffer_make_writable (buf);
gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
}
} else {
gst_buffer_unref (buf);
goto found_eos;
}
}
}
obtained = gst_buffer_get_size (buf);
/* our positions in bytes */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
buf = gst_buffer_make_writable (buf);
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
/* first chunk of data? create the source pad. We do this only here so
* we can detect broken .wav files with dts disguised as raw PCM (sigh) */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
/* this will also push the segment events */
gst_wavparse_add_src_pad (wav, buf);
} else {
/* If we have a pending start segment, send it now. */
if (G_UNLIKELY (wav->start_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
}
}
if (wav->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp =
gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
next_timestamp =
gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
if (G_LIKELY (next_timestamp >= wav->segment.start))
wav->segment.position = next_timestamp;
} else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
duration = next_timestamp - timestamp;
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
* other positions and durations have unknown timestamp. */
if (pos == 0)
timestamp = 0;
else
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
if (G_LIKELY (nextpos >= wav->segment.start))
wav->segment.position = nextpos;
}
if ((pos > 0) && wav->vbr) {
/* don't set timestamps for VBR files if it's not the first buffer */
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
}
if (wav->discont) {
GST_DEBUG_OBJECT (wav, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
wav->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_LOG_OBJECT (wav,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < wav->dataleft) {
wav->offset += obtained;
wav->dataleft -= obtained;
} else {
wav->offset += wav->dataleft;
wav->dataleft = 0;
}
/* Iterate until need more data, so adapter size won't grow */
if (wav->streaming) {
GST_LOG_OBJECT (wav,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
wav->end_offset);
goto iterate_adapter;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (wav, "found EOS");
return GST_FLOW_EOS;
}
pull_error:
{
/* check if we got EOS */
if (res == GST_FLOW_EOS)
goto found_eos;
GST_WARNING_OBJECT (wav,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
return res;
}
push_error:
{
GST_INFO_OBJECT (wav,
"Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
gst_pad_is_linked (wav->srcpad));
return res;
}
}
static void
gst_wavparse_loop (GstPad * pad)
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
GstEvent *event;
gchar *stream_id;
GST_LOG_OBJECT (wav, "process data");
switch (wav->state) {
case GST_WAVPARSE_START:
GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
goto pause;
stream_id =
gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
gst_pad_push_event (wav->srcpad, event);
g_free (stream_id);
wav->state = GST_WAVPARSE_HEADER;
/* fall-through */
case GST_WAVPARSE_HEADER:
GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto pause;
wav->state = GST_WAVPARSE_DATA;
GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
/* fall-through */
case GST_WAVPARSE_DATA:
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto pause;
break;
default:
g_assert_not_reached ();
}
return;
/* ERRORS */
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
gst_pad_pause_task (pad);
if (ret == GST_FLOW_EOS) {
/* handle end-of-stream/segment */
/* so align our position with the end of it, if there is one
* this ensures a subsequent will arrive at correct base/acc time */
if (wav->segment.format == GST_FORMAT_TIME) {
if (wav->segment.rate > 0.0 &&
GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
wav->segment.position = wav->segment.stop;
else if (wav->segment.rate < 0.0)
wav->segment.position = wav->segment.start;
}
if (wav->state == GST_WAVPARSE_START || !wav->caps) {
GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
("No valid input found before end of stream"));
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
} else {
/* add pad before we perform EOS */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, NULL);
}
/* perform EOS logic */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
if ((stop = wav->segment.stop) == -1)
stop = wav->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_done (GST_OBJECT_CAST (wav),
wav->segment.format, stop));
gst_pad_push_event (wav->srcpad,
gst_event_new_segment_done (wav->segment.format, stop));
} else {
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_FLOW_ERROR (wav, ret);
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
return;
}
}
static GstFlowReturn
gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (parent);
GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buf));
gst_adapter_push (wav->adapter, buf);
switch (wav->state) {
case GST_WAVPARSE_START:
GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
goto done;
if (wav->state != GST_WAVPARSE_HEADER)
break;
/* otherwise fall-through */
case GST_WAVPARSE_HEADER:
GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
goto done;
if (!wav->got_fmt || wav->datastart == 0)
break;
wav->state = GST_WAVPARSE_DATA;
GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
/* fall-through */
case GST_WAVPARSE_DATA:
if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
wav->discont = TRUE;
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
goto done;
break;
default:
g_return_val_if_reached (GST_FLOW_ERROR);
}
done:
if (G_UNLIKELY (wav->abort_buffering)) {
wav->abort_buffering = FALSE;
ret = GST_FLOW_ERROR;
/* sort of demux/parse error */
GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
}
return ret;
}
static GstFlowReturn
gst_wavparse_flush_data (GstWavParse * wav)
{
GstFlowReturn ret = GST_FLOW_OK;
guint av;
if ((av = gst_adapter_available (wav->adapter)) > 0) {
ret = gst_wavparse_stream_data (wav);
}
return ret;
}
static gboolean
gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstWavParse *wav = GST_WAVPARSE (parent);
gboolean ret = TRUE;
GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
/* discard, we'll come up with proper src caps */
gst_event_unref (event);
break;
}
case GST_EVENT_SEGMENT:
{
gint64 start, stop, offset = 0, end_offset = -1;
GstSegment segment;
/* some debug output */
gst_event_copy_segment (event, &segment);
GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
&segment);
if (wav->state != GST_WAVPARSE_DATA) {
GST_DEBUG_OBJECT (wav, "still starting, eating event");
goto exit;
}
/* now we are either committed to TIME or BYTE format,
* and we only expect a BYTE segment, e.g. following a seek */
if (segment.format == GST_FORMAT_BYTES) {
/* handle (un)signed issues */
start = segment.start;
stop = segment.stop;
if (start > 0) {
offset = start;
start -= wav->datastart;
start = MAX (start, 0);
}
if (stop > 0) {
end_offset = stop;
stop -= wav->datastart;
stop = MAX (stop, 0);
}
if (wav->segment.format == GST_FORMAT_TIME) {
guint64 bps = wav->bps;
/* operating in format TIME, so we can convert */
if (!bps && wav->fact)
bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
if (bps) {
if (start >= 0)
start =
gst_util_uint64_scale_ceil (start, GST_SECOND,
(guint64) wav->bps);
if (stop >= 0)
stop =
gst_util_uint64_scale_ceil (stop, GST_SECOND,
(guint64) wav->bps);
}
}
} else {
GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
goto exit;
}
segment.start = start;
segment.stop = stop;
/* accept upstream's notion of segment and distribute along */
segment.format = wav->segment.format;
segment.time = segment.position = segment.start;
segment.duration = wav->segment.duration;
segment.base = gst_segment_to_running_time (&wav->segment,
GST_FORMAT_TIME, wav->segment.position);
gst_segment_copy_into (&segment, &wav->segment);
/* also store the newsegment event for the streaming thread */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
wav->start_segment = gst_event_new_segment (&segment);
/* stream leftover data in current segment */
gst_wavparse_flush_data (wav);
/* and set up streaming thread for next one */
wav->offset = offset;
wav->end_offset = end_offset;
if (wav->datasize > 0 && (wav->end_offset == -1
|| wav->end_offset > wav->datastart + wav->datasize))
wav->end_offset = wav->datastart + wav->datasize;
if (wav->end_offset != -1) {
wav->dataleft = wav->end_offset - wav->offset;
} else {
/* infinity; upstream will EOS when done */
wav->dataleft = G_MAXUINT64;
}
exit:
gst_event_unref (event);
break;
}
case GST_EVENT_EOS:
if (wav->state == GST_WAVPARSE_START || !wav->caps) {
GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
("No valid input found before end of stream"));
} else {
/* add pad if needed so EOS is seen downstream */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, NULL);
} else {
/* stream leftover data in current segment */
gst_wavparse_flush_data (wav);
}
}
/* fall-through */
case GST_EVENT_FLUSH_STOP:
{
GstClockTime dur;
if (wav->adapter)
gst_adapter_clear (wav->adapter);
wav->discont = TRUE;
dur = wav->segment.duration;
gst_segment_init (&wav->segment, wav->segment.format);
wav->segment.duration = dur;
/* fall-through */
}
default:
ret = gst_pad_event_default (wav->sinkpad, parent, event);
break;
}
return ret;
}
#if 0
/* convert and query stuff */
static const GstFormat *
gst_wavparse_get_formats (GstPad * pad)
{
static const GstFormat formats[] = {
GST_FORMAT_TIME,
GST_FORMAT_BYTES,
GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
0
};
return formats;
}
#endif
static gboolean
gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
GstWavParse *wavparse;
gboolean res = TRUE;
wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
if (*dest_format == src_format) {
*dest_value = src_value;
return TRUE;
}
if ((wavparse->bps == 0) && !wavparse->fact)
goto no_bps_fact;
GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
gst_format_get_name (src_format), gst_format_get_name (*dest_format));
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / wavparse->bytes_per_sample;
/* make sure we end up on a sample boundary */
*dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
/* src_value + datastart = offset */
GST_INFO_OBJECT (wavparse,
"src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
wavparse->offset);
if (wavparse->bps > 0)
*dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
(guint64) wavparse->bps);
else if (wavparse->fact) {
guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
wavparse->rate, wavparse->fact);
*dest_value =
gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
} else {
res = FALSE;
}
break;
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
(guint64) wavparse->rate);
break;
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
if (wavparse->bps > 0)
*dest_value = gst_util_uint64_scale (src_value,
(guint64) wavparse->bps, GST_SECOND);
else {
guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
wavparse->rate, wavparse->fact);
*dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
}
/* make sure we end up on a sample boundary */
*dest_value -= *dest_value % wavparse->blockalign;
break;
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale (src_value,
(guint64) wavparse->rate, GST_SECOND);
break;
default:
res = FALSE;
goto done;
}
break;
default:
res = FALSE;
goto done;
}
done:
return res;
/* ERRORS */
no_bps_fact:
{
GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
res = FALSE;
goto done;
}
}
/* handle queries for location and length in requested format */
static gboolean
gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
gboolean res = TRUE;
GstWavParse *wav = GST_WAVPARSE (parent);
/* only if we know */
if (wav->state != GST_WAVPARSE_DATA) {
return FALSE;
}
GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
gint64 curb;
gint64 cur;
GstFormat format;
/* this is not very precise, as we have pushed severla buffer upstream for prerolling */
curb = wav->offset - wav->datastart;
gst_query_parse_position (query, &format, NULL);
GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
switch (format) {
case GST_FORMAT_BYTES:
format = GST_FORMAT_BYTES;
cur = curb;
break;
default:
res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
&format, &cur);
break;
}
if (res)
gst_query_set_position (query, format, cur);
break;
}
case GST_QUERY_DURATION:
{
gint64 duration = 0;
GstFormat format;
if (wav->ignore_length) {
res = FALSE;
break;
}
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_BYTES:{
format = GST_FORMAT_BYTES;
duration = wav->datasize;
break;
}
case GST_FORMAT_TIME:
if ((res = gst_wavparse_calculate_duration (wav))) {
duration = wav->duration;
}
break;
default:
res = FALSE;
break;
}
if (res)
gst_query_set_duration (query, format, duration);
break;
}
case GST_QUERY_CONVERT:
{
gint64 srcvalue, dstvalue;
GstFormat srcformat, dstformat;
gst_query_parse_convert (query, &srcformat, &srcvalue,
&dstformat, &dstvalue);
res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
&dstformat, &dstvalue);
if (res)
gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
break;
}
case GST_QUERY_SEEKING:{
GstFormat fmt;
gboolean seekable = FALSE;
gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
if (fmt == wav->segment.format) {
if (wav->streaming) {
GstQuery *q;
q = gst_query_new_seeking (GST_FORMAT_BYTES);
if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
}
gst_query_unref (q);
} else {
GST_LOG_OBJECT (wav, "looping => seekable");
seekable = TRUE;
res = TRUE;
}
} else if (fmt == GST_FORMAT_TIME) {
res = TRUE;
}
if (res) {
gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static gboolean
gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstWavParse *wavparse = GST_WAVPARSE (parent);
gboolean res = FALSE;
GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
/* can only handle events when we are in the data state */
if (wavparse->state == GST_WAVPARSE_DATA) {
res = gst_wavparse_perform_seek (wavparse, event);
}
gst_event_unref (event);
break;
case GST_EVENT_TOC_SELECT:
{
char *uid = NULL;
GstTocEntry *entry = NULL;
GstEvent *seek_event;
gint64 start_pos;
if (!wavparse->toc) {
GST_DEBUG_OBJECT (wavparse, "no TOC to select");
return FALSE;
} else {
gst_event_parse_toc_select (event, &uid);
if (uid != NULL) {
GST_OBJECT_LOCK (wavparse);
entry = gst_toc_find_entry (wavparse->toc, uid);
if (entry == NULL) {
GST_OBJECT_UNLOCK (wavparse);
GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
uid);
res = FALSE;
} else {
gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
GST_OBJECT_UNLOCK (wavparse);
seek_event = gst_event_new_seek (1.0,
GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
res = gst_wavparse_perform_seek (wavparse, seek_event);
gst_event_unref (seek_event);
}
g_free (uid);
} else {
GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
res = FALSE;
}
}
gst_event_unref (event);
break;
}
default:
res = gst_pad_push_event (wavparse->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
{
GstWavParse *wav = GST_WAVPARSE (parent);
GstQuery *query;
gboolean pull_mode;
if (wav->adapter) {
gst_adapter_clear (wav->adapter);
g_object_unref (wav->adapter);
wav->adapter = NULL;
}
query = gst_query_new_scheduling ();
if (!gst_pad_peer_query (sinkpad, query)) {
gst_query_unref (query);
goto activate_push;
}
pull_mode = gst_query_has_scheduling_mode_with_flags (query,
GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
gst_query_unref (query);
if (!pull_mode)
goto activate_push;
GST_DEBUG_OBJECT (sinkpad, "activating pull");
wav->streaming = FALSE;
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
activate_push:
{
GST_DEBUG_OBJECT (sinkpad, "activating push");
wav->streaming = TRUE;
wav->adapter = gst_adapter_new ();
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
}
}
static gboolean
gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean res;
switch (mode) {
case GST_PAD_MODE_PUSH:
res = TRUE;
break;
case GST_PAD_MODE_PULL:
if (active) {
/* if we have a scheduler we can start the task */
res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
sinkpad, NULL);
} else {
res = gst_pad_stop_task (sinkpad);
}
break;
default:
res = FALSE;
break;
}
return res;
}
static GstStateChangeReturn
gst_wavparse_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstWavParse *wav = GST_WAVPARSE (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_wavparse_reset (wav);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavparse_reset (wav);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_wavparse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWavParse *self;
g_return_if_fail (GST_IS_WAVPARSE (object));
self = GST_WAVPARSE (object);
switch (prop_id) {
case PROP_IGNORE_LENGTH:
self->ignore_length = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
}
}
static void
gst_wavparse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWavParse *self;
g_return_if_fail (GST_IS_WAVPARSE (object));
self = GST_WAVPARSE (object);
switch (prop_id) {
case PROP_IGNORE_LENGTH:
g_value_set_boolean (value, self->ignore_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
gst_riff_init ();
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
GST_TYPE_WAVPARSE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
wavparse,
"Parse a .wav file into raw audio",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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