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gstreamer/gst/wavparse/gstwavparse.c
Rahul Bedarkar f05c0c036e wavparse: check for not NULL before clearing adapter 
In case wavparse receives a manually injected FLUSH_STOP event
while operating in pull mode we get criticals because we'd try
to clear a NULL adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=777123
2017-01-25 09:59:50 +00:00

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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 (wav->datasize > 0 && size + wav->offset > wav->datasize) {
GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
size = wav->datasize - 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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