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gstreamer/ext/flac/gstflacdec.c
Tim-Philipp Müller cee10cb439 flacdec: send newsegment event when operating push-based and unframed 
For some reason flac doesn't call our metadata callback when we operate
in push mode with unframed input, but that's where we set up the
newsegment event (since that's where we'd get the duration from the
stream info header), so we didn't send a newsegment event at all in this
case. Hack around this by storing a generic newsegment event for now
which will be used if we don't replace it with a better one that
includes the duration.
2009-08-01 17:07:42 +01:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
* Copyright (C) <2006> Jan Schmidt <thaytan at mad scientist 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-flacdec
* @see_also: #GstFlacEnc
*
* flacdec decodes FLAC streams.
* <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
* is a Free Lossless Audio Codec.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* |[
* gst-launch gnomevfssrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
* ]|
* </refsect2>
*/
/* TODO: add seeking when operating chain-based with unframed input */
/* FIXME: merge dec->seekable_decoder and dec->stream_decoder now that they're
* the same type */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstflacdec.h"
#include <gst/gst-i18n-plugin.h>
#include <gst/gsttagsetter.h>
#include <gst/base/gsttypefindhelper.h>
#include <gst/audio/multichannel.h>
#include <gst/tag/tag.h>
/* Taken from http://flac.sourceforge.net/format.html#frame_header */
static const GstAudioChannelPosition channel_positions[8][8] = {
{GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
/* FIXME: 7/8 channel layouts are not defined in the FLAC specs */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}
};
GST_DEBUG_CATEGORY_STATIC (flacdec_debug);
#define GST_CAT_DEFAULT flacdec_debug
static void gst_flac_dec_finalize (GObject * object);
static void gst_flac_dec_loop (GstPad * pad);
static GstStateChangeReturn gst_flac_dec_change_state (GstElement * element,
GstStateChange transition);
static const GstQueryType *gst_flac_dec_get_src_query_types (GstPad * pad);
static const GstQueryType *gst_flac_dec_get_sink_query_types (GstPad * pad);
static gboolean gst_flac_dec_sink_query (GstPad * pad, GstQuery * query);
static gboolean gst_flac_dec_src_query (GstPad * pad, GstQuery * query);
static gboolean gst_flac_dec_convert_src (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
static gboolean gst_flac_dec_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_flac_dec_sink_activate (GstPad * sinkpad);
static gboolean gst_flac_dec_sink_activate_pull (GstPad * sinkpad,
gboolean active);
static gboolean gst_flac_dec_sink_activate_push (GstPad * sinkpad,
gboolean active);
static gboolean gst_flac_dec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_flac_dec_chain (GstPad * pad, GstBuffer * buf);
static void gst_flac_dec_reset_decoders (GstFlacDec * flacdec);
static void gst_flac_dec_setup_seekable_decoder (GstFlacDec * flacdec);
static void gst_flac_dec_setup_stream_decoder (GstFlacDec * flacdec);
static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_seekable (const FLAC__StreamDecoder * decoder,
FLAC__byte buffer[], size_t * bytes, void *client_data);
static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
FLAC__byte buffer[], size_t * bytes, void *client_data);
static FLAC__StreamDecoderSeekStatus
gst_flac_dec_seek (const FLAC__StreamDecoder * decoder,
FLAC__uint64 position, void *client_data);
static FLAC__StreamDecoderTellStatus
gst_flac_dec_tell (const FLAC__StreamDecoder * decoder,
FLAC__uint64 * position, void *client_data);
static FLAC__StreamDecoderLengthStatus
gst_flac_dec_length (const FLAC__StreamDecoder * decoder,
FLAC__uint64 * length, void *client_data);
static FLAC__bool gst_flac_dec_eof (const FLAC__StreamDecoder * decoder,
void *client_data);
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
const FLAC__Frame * frame,
const FLAC__int32 * const buffer[], void *client_data);
static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder *
decoder, const FLAC__StreamMetadata * metadata, void *client_data);
static void gst_flac_dec_error_cb (const FLAC__StreamDecoder *
decoder, FLAC__StreamDecoderErrorStatus status, void *client_data);
GST_BOILERPLATE (GstFlacDec, gst_flac_dec, GstElement, GST_TYPE_ELEMENT);
/* FIXME 0.11: Use width=32 for all depths and let audioconvert
* handle the conversions instead of doing it ourself.
*/
#define GST_FLAC_DEC_SRC_CAPS \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) { 8, 16, 32 }, " \
"depth = (int) [ 4, 32 ], " \
"rate = (int) [ 1, 655350 ], " \
"channels = (int) [ 1, 8 ]"
static GstStaticPadTemplate flac_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_FLAC_DEC_SRC_CAPS));
static GstStaticPadTemplate flac_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-flac")
);
static void
gst_flac_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&flac_dec_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&flac_dec_sink_factory));
gst_element_class_set_details_simple (element_class, "FLAC audio decoder",
"Codec/Decoder/Audio",
"Decodes FLAC lossless audio streams", "Wim Taymans <wim@fluendo.com>");
GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder");
}
static void
gst_flac_dec_class_init (GstFlacDecClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *gobject_class;
gstelement_class = (GstElementClass *) klass;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = gst_flac_dec_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_flac_dec_change_state);
}
static void
gst_flac_dec_init (GstFlacDec * flacdec, GstFlacDecClass * klass)
{
flacdec->sinkpad =
gst_pad_new_from_static_template (&flac_dec_sink_factory, "sink");
gst_pad_set_activate_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_sink_activate));
gst_pad_set_activatepull_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_sink_activate_pull));
gst_pad_set_activatepush_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_sink_activate_push));
gst_pad_set_query_type_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_get_sink_query_types));
gst_pad_set_query_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_sink_query));
gst_pad_set_event_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_sink_event));
gst_pad_set_chain_function (flacdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_chain));
gst_element_add_pad (GST_ELEMENT (flacdec), flacdec->sinkpad);
flacdec->srcpad =
gst_pad_new_from_static_template (&flac_dec_src_factory, "src");
gst_pad_set_query_type_function (flacdec->srcpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_get_src_query_types));
gst_pad_set_query_function (flacdec->srcpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_src_query));
gst_pad_set_event_function (flacdec->srcpad,
GST_DEBUG_FUNCPTR (gst_flac_dec_src_event));
gst_pad_use_fixed_caps (flacdec->srcpad);
gst_element_add_pad (GST_ELEMENT (flacdec), flacdec->srcpad);
gst_flac_dec_reset_decoders (flacdec);
}
static void
gst_flac_dec_reset_decoders (GstFlacDec * flacdec)
{
if (flacdec->seekable_decoder) {
FLAC__stream_decoder_delete (flacdec->seekable_decoder);
flacdec->seekable_decoder = NULL;
}
/* Clean up the stream_decoder */
if (flacdec->stream_decoder) {
FLAC__stream_decoder_delete (flacdec->stream_decoder);
flacdec->stream_decoder = NULL;
}
if (flacdec->adapter) {
gst_adapter_clear (flacdec->adapter);
g_object_unref (flacdec->adapter);
flacdec->adapter = NULL;
}
if (flacdec->close_segment) {
gst_event_unref (flacdec->close_segment);
flacdec->close_segment = NULL;
}
if (flacdec->start_segment) {
gst_event_unref (flacdec->start_segment);
flacdec->start_segment = NULL;
}
if (flacdec->tags) {
gst_tag_list_free (flacdec->tags);
flacdec->tags = NULL;
}
if (flacdec->pending) {
gst_buffer_unref (flacdec->pending);
flacdec->pending = NULL;
}
flacdec->segment.last_stop = 0;
flacdec->offset = 0;
flacdec->init = TRUE;
}
static void
gst_flac_dec_setup_seekable_decoder (GstFlacDec * dec)
{
gst_flac_dec_reset_decoders (dec);
dec->tags = gst_tag_list_new ();
gst_tag_list_add (dec->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "FLAC", NULL);
dec->seekable_decoder = FLAC__stream_decoder_new ();
FLAC__stream_decoder_set_metadata_respond (dec->seekable_decoder,
FLAC__METADATA_TYPE_VORBIS_COMMENT);
FLAC__stream_decoder_set_metadata_respond (dec->seekable_decoder,
FLAC__METADATA_TYPE_PICTURE);
FLAC__stream_decoder_set_md5_checking (dec->seekable_decoder, false); /* no point calculating since it's never checked here */
}
static void
gst_flac_dec_setup_stream_decoder (GstFlacDec * dec)
{
gst_flac_dec_reset_decoders (dec);
dec->tags = gst_tag_list_new ();
gst_tag_list_add (dec->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "FLAC", NULL);
dec->adapter = gst_adapter_new ();
dec->stream_decoder = FLAC__stream_decoder_new ();
FLAC__stream_decoder_set_md5_checking (dec->stream_decoder, false); /* no point calculating since it's never checked here */
FLAC__stream_decoder_set_metadata_respond (dec->stream_decoder,
FLAC__METADATA_TYPE_VORBIS_COMMENT);
FLAC__stream_decoder_set_metadata_respond (dec->stream_decoder,
FLAC__METADATA_TYPE_PICTURE);
}
static void
gst_flac_dec_finalize (GObject * object)
{
GstFlacDec *flacdec;
flacdec = GST_FLAC_DEC (object);
gst_flac_dec_reset_decoders (flacdec);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_flac_dec_update_metadata (GstFlacDec * flacdec,
const FLAC__StreamMetadata * metadata)
{
GstTagList *list;
guint num, i;
if (flacdec->tags)
list = flacdec->tags;
else
flacdec->tags = list = gst_tag_list_new ();
num = metadata->data.vorbis_comment.num_comments;
GST_DEBUG_OBJECT (flacdec, "%u tag(s) found", num);
for (i = 0; i < num; ++i) {
gchar *vc, *name, *value;
vc = g_strndup ((gchar *) metadata->data.vorbis_comment.comments[i].entry,
metadata->data.vorbis_comment.comments[i].length);
if (gst_tag_parse_extended_comment (vc, &name, NULL, &value, TRUE)) {
GST_DEBUG_OBJECT (flacdec, "%s : %s", name, value);
if (value && strlen (value))
gst_vorbis_tag_add (list, name, value);
g_free (name);
g_free (value);
}
g_free (vc);
}
return TRUE;
}
/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
static const guint8 crc8_table[256] = {
0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
};
static guint8
gst_flac_calculate_crc8 (guint8 * data, guint length)
{
guint8 crc = 0;
while (length--) {
crc = crc8_table[crc ^ *data];
++data;
}
return crc;
}
static gboolean
gst_flac_dec_scan_got_frame (GstFlacDec * flacdec, guint8 * data, guint size,
gint64 * last_sample_num)
{
guint headerlen;
guint sr_from_end = 0; /* can be 0, 8 or 16 */
guint bs_from_end = 0; /* can be 0, 8 or 16 */
guint32 val = 0;
guint8 bs, sr, ca, ss, pb;
if (size < 10)
return FALSE;
/* sync */
if (data[0] != 0xFF || data[1] != 0xF8)
return FALSE;
bs = (data[2] & 0xF0) >> 8; /* blocksize marker */
sr = (data[2] & 0x0F); /* samplerate marker */
ca = (data[3] & 0xF0) >> 8; /* channel assignment */
ss = (data[3] & 0x0F) >> 1; /* sample size marker */
pb = (data[3] & 0x01); /* padding bit */
GST_LOG_OBJECT (flacdec,
"got sync, bs=%x,sr=%x,ca=%x,ss=%x,pb=%x", bs, sr, ca, ss, pb);
if (sr == 0x0F || sr == 0x01 || sr == 0x02 || sr == 0x03 ||
ca >= 0x0B || ss == 0x03 || ss == 0x07) {
return FALSE;
}
/* read block size from end of header? */
if (bs == 6)
bs_from_end = 8;
else if (bs == 7)
bs_from_end = 16;
/* read sample rate from end of header? */
if (sr == 0x0C)
sr_from_end = 8;
else if (sr == 0x0D || sr == 0x0E)
sr_from_end = 16;
val = (guint32) g_utf8_get_char_validated ((gchar *) data + 4, -1);
if (val == (guint32) - 1 || val == (guint32) - 2) {
GST_LOG_OBJECT (flacdec, "failed to read sample/frame");
return FALSE;
}
headerlen = 4 + g_unichar_to_utf8 ((gunichar) val, NULL) +
(bs_from_end / 8) + (sr_from_end / 8);
if (gst_flac_calculate_crc8 (data, headerlen) != data[headerlen])
return FALSE;
if (flacdec->min_blocksize == flacdec->max_blocksize) {
*last_sample_num = (val + 1) * flacdec->min_blocksize;
} else {
*last_sample_num = val; /* FIXME: + length of last block in samples */
}
if (flacdec->sample_rate > 0) {
GST_DEBUG_OBJECT (flacdec, "last sample %" G_GINT64_FORMAT " = %"
GST_TIME_FORMAT, *last_sample_num,
GST_TIME_ARGS (*last_sample_num * GST_SECOND / flacdec->sample_rate));
}
return TRUE;
}
#define SCANBLOCK_SIZE (64*1024)
static void
gst_flac_dec_scan_for_last_block (GstFlacDec * flacdec, gint64 * samples)
{
GstFormat format = GST_FORMAT_BYTES;
gint64 file_size, offset;
GST_INFO_OBJECT (flacdec, "total number of samples unknown, scanning file");
if (!gst_pad_query_peer_duration (flacdec->sinkpad, &format, &file_size)) {
GST_WARNING_OBJECT (flacdec, "failed to query upstream size!");
return;
}
GST_DEBUG_OBJECT (flacdec, "upstream size: %" G_GINT64_FORMAT, file_size);
offset = file_size - 1;
while (offset >= MAX (SCANBLOCK_SIZE / 2, file_size / 2)) {
GstFlowReturn flow;
GstBuffer *buf = NULL;
guint8 *data;
guint size;
/* divide by 2 = not very sophisticated way to deal with overlapping */
offset -= SCANBLOCK_SIZE / 2;
GST_LOG_OBJECT (flacdec, "looking for frame at %" G_GINT64_FORMAT
"-%" G_GINT64_FORMAT, offset, offset + SCANBLOCK_SIZE);
flow = gst_pad_pull_range (flacdec->sinkpad, offset, SCANBLOCK_SIZE, &buf);
if (flow != GST_FLOW_OK) {
GST_DEBUG_OBJECT (flacdec, "flow = %s", gst_flow_get_name (flow));
return;
}
size = GST_BUFFER_SIZE (buf);
data = GST_BUFFER_DATA (buf);
while (size > 16) {
if (gst_flac_dec_scan_got_frame (flacdec, data, size, samples)) {
GST_DEBUG_OBJECT (flacdec, "frame sync at offset %" G_GINT64_FORMAT,
offset + GST_BUFFER_SIZE (buf) - size);
gst_buffer_unref (buf);
return;
}
++data;
--size;
}
gst_buffer_unref (buf);
}
}
static void
gst_flac_extract_picture_buffer (GstFlacDec * dec,
const FLAC__StreamMetadata * metadata)
{
FLAC__StreamMetadata_Picture picture;
GstTagList *tags;
g_return_if_fail (metadata->type == FLAC__METADATA_TYPE_PICTURE);
GST_LOG_OBJECT (dec, "Got PICTURE block");
picture = metadata->data.picture;
GST_DEBUG_OBJECT (dec, "declared MIME type is: '%s'",
GST_STR_NULL (picture.mime_type));
GST_DEBUG_OBJECT (dec, "image data is %u bytes", picture.data_length);
tags = gst_tag_list_new ();
gst_tag_list_add_id3_image (tags, (guint8 *) picture.data,
picture.data_length, picture.type);
if (!gst_tag_list_is_empty (tags)) {
gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, tags);
} else {
GST_DEBUG_OBJECT (dec, "problem parsing PICTURE block, skipping");
gst_tag_list_free (tags);
}
}
static void
gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder,
const FLAC__StreamMetadata * metadata, void *client_data)
{
GstFlacDec *flacdec = GST_FLAC_DEC (client_data);
GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type);
switch (metadata->type) {
case FLAC__METADATA_TYPE_STREAMINFO:{
gint64 samples;
samples = metadata->data.stream_info.total_samples;
flacdec->min_blocksize = metadata->data.stream_info.min_blocksize;
flacdec->max_blocksize = metadata->data.stream_info.max_blocksize;
flacdec->sample_rate = metadata->data.stream_info.sample_rate;
flacdec->depth = metadata->data.stream_info.bits_per_sample;
GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u",
flacdec->min_blocksize, flacdec->max_blocksize);
/* Only scan for last block in pull-mode, since it uses pull_range() */
if (samples == 0 && flacdec->seekable_decoder) {
gst_flac_dec_scan_for_last_block (flacdec, &samples);
}
GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
if (samples > 0) {
gint64 duration;
gst_segment_set_duration (&flacdec->segment, GST_FORMAT_DEFAULT,
samples);
/* convert duration to time */
duration = gst_util_uint64_scale_int (samples, GST_SECOND,
flacdec->sample_rate);
/* fixme, at this time we could seek to the queued seek event if we have
* any */
if (flacdec->start_segment)
gst_event_unref (flacdec->start_segment);
flacdec->start_segment =
gst_event_new_new_segment_full (FALSE,
flacdec->segment.rate, flacdec->segment.applied_rate,
GST_FORMAT_TIME, 0, duration, 0);
}
break;
}
case FLAC__METADATA_TYPE_PICTURE:{
gst_flac_extract_picture_buffer (flacdec, metadata);
break;
}
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
gst_flac_dec_update_metadata (flacdec, metadata);
break;
default:
break;
}
}
static void
gst_flac_dec_error_cb (const FLAC__StreamDecoder * d,
FLAC__StreamDecoderErrorStatus status, void *client_data)
{
const gchar *error;
GstFlacDec *dec;
dec = GST_FLAC_DEC (client_data);
switch (status) {
case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC:
/* Ignore this error and keep processing */
return;
case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER:
error = "bad header";
break;
case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH:
error = "CRC mismatch";
break;
default:
error = "unknown error";
break;
}
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("%s (%d)", error, status));
dec->last_flow = GST_FLOW_ERROR;
}
static FLAC__StreamDecoderSeekStatus
gst_flac_dec_seek (const FLAC__StreamDecoder * decoder,
FLAC__uint64 position, void *client_data)
{
GstFlacDec *flacdec;
flacdec = GST_FLAC_DEC (client_data);
GST_DEBUG_OBJECT (flacdec, "seek %" G_GINT64_FORMAT, position);
flacdec->offset = position;
return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
}
static FLAC__StreamDecoderTellStatus
gst_flac_dec_tell (const FLAC__StreamDecoder * decoder,
FLAC__uint64 * position, void *client_data)
{
GstFlacDec *flacdec;
flacdec = GST_FLAC_DEC (client_data);
*position = flacdec->offset;
GST_DEBUG_OBJECT (flacdec, "tell %" G_GINT64_FORMAT, *position);
return FLAC__STREAM_DECODER_TELL_STATUS_OK;
}
static FLAC__StreamDecoderLengthStatus
gst_flac_dec_length (const FLAC__StreamDecoder * decoder,
FLAC__uint64 * length, void *client_data)
{
GstFlacDec *flacdec;
GstFormat fmt = GST_FORMAT_BYTES;
gint64 len;
GstPad *peer;
flacdec = GST_FLAC_DEC (client_data);
if (!(peer = gst_pad_get_peer (flacdec->sinkpad)))
return FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR;
gst_pad_query_duration (peer, &fmt, &len);
gst_object_unref (peer);
if (fmt != GST_FORMAT_BYTES || len == -1)
return FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR;
*length = len;
GST_DEBUG_OBJECT (flacdec, "encoded byte length %" G_GINT64_FORMAT, *length);
return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
}
static FLAC__bool
gst_flac_dec_eof (const FLAC__StreamDecoder * decoder, void *client_data)
{
GstFlacDec *flacdec;
GstFormat fmt;
GstPad *peer;
gboolean ret = FALSE;
gint64 len;
flacdec = GST_FLAC_DEC (client_data);
if (!(peer = gst_pad_get_peer (flacdec->sinkpad))) {
GST_WARNING_OBJECT (flacdec, "no peer pad, returning EOF");
return TRUE;
}
fmt = GST_FORMAT_BYTES;
if (gst_pad_query_duration (peer, &fmt, &len) && fmt == GST_FORMAT_BYTES &&
len != -1 && flacdec->offset >= len) {
GST_DEBUG_OBJECT (flacdec,
"offset=%" G_GINT64_FORMAT ", len=%" G_GINT64_FORMAT
", returning EOF", flacdec->offset, len);
ret = TRUE;
}
gst_object_unref (peer);
return ret;
}
static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_seekable (const FLAC__StreamDecoder * decoder,
FLAC__byte buffer[], size_t * bytes, void *client_data)
{
GstFlowReturn flow;
GstFlacDec *flacdec;
GstBuffer *buf;
flacdec = GST_FLAC_DEC (client_data);
flow = gst_pad_pull_range (flacdec->sinkpad, flacdec->offset, *bytes, &buf);
GST_PAD_STREAM_LOCK (flacdec->sinkpad);
flacdec->pull_flow = flow;
GST_PAD_STREAM_UNLOCK (flacdec->sinkpad);
if (G_UNLIKELY (flow != GST_FLOW_OK)) {
GST_INFO_OBJECT (flacdec, "pull_range flow: %s", gst_flow_get_name (flow));
if (flow == GST_FLOW_UNEXPECTED)
return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
else
return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
}
GST_DEBUG_OBJECT (flacdec, "Read %d bytes at %" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (buf), flacdec->offset);
memcpy (buffer, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
*bytes = GST_BUFFER_SIZE (buf);
gst_buffer_unref (buf);
flacdec->offset += *bytes;
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}
static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
FLAC__byte buffer[], size_t * bytes, void *client_data)
{
GstFlacDec *dec = GST_FLAC_DEC (client_data);
guint len;
len = MIN (gst_adapter_available (dec->adapter), *bytes);
if (len == 0) {
GST_LOG_OBJECT (dec, "0 bytes available at the moment");
return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
}
GST_LOG_OBJECT (dec, "feeding %u bytes to decoder (available=%u, bytes=%u)",
len, gst_adapter_available (dec->adapter), (guint) * bytes);
gst_adapter_copy (dec->adapter, buffer, 0, len);
*bytes = len;
gst_adapter_flush (dec->adapter, len);
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
const FLAC__int32 * const buffer[])
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
guint depth = frame->header.bits_per_sample;
guint width;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstClockTime next;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
/* if a DEFAULT segment is configured, don't send samples past the end
* of the segment */
if (flacdec->segment.format == GST_FORMAT_DEFAULT &&
flacdec->segment.stop != -1 &&
flacdec->segment.last_stop >= 0 &&
flacdec->segment.last_stop + samples > flacdec->segment.stop) {
samples = flacdec->segment.stop - flacdec->segment.last_stop;
GST_DEBUG_OBJECT (flacdec,
"clipping last buffer to %d samples because of segment", samples);
}
switch (depth) {
case 8:
width = 8;
break;
case 12:
case 16:
width = 16;
break;
case 20:
case 24:
case 32:
width = 32;
break;
case 0:
if (flacdec->depth < 4 || flacdec->depth > 32) {
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
flacdec->depth);
ret = GST_FLOW_ERROR;
goto done;
}
depth = flacdec->depth;
if (depth < 9)
width = 8;
else if (depth < 17)
width = 16;
else
width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (!GST_PAD_CAPS (flacdec->srcpad)) {
GstCaps *caps;
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels",
frame->header.sample_rate, channels);
caps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, width,
"depth", G_TYPE_INT, depth,
"rate", G_TYPE_INT, frame->header.sample_rate,
"channels", G_TYPE_INT, channels, NULL);
if (channels > 2) {
GstStructure *s = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (s, channel_positions[channels - 1]);
}
flacdec->depth = depth;
flacdec->width = width;
flacdec->channels = channels;
flacdec->sample_rate = frame->header.sample_rate;
gst_pad_set_caps (flacdec->srcpad, caps);
gst_caps_unref (caps);
}
if (flacdec->close_segment) {
GST_DEBUG_OBJECT (flacdec, "pushing close segment");
gst_pad_push_event (flacdec->srcpad, flacdec->close_segment);
flacdec->close_segment = NULL;
}
if (flacdec->start_segment) {
GST_DEBUG_OBJECT (flacdec, "pushing start segment");
gst_pad_push_event (flacdec->srcpad, flacdec->start_segment);
flacdec->start_segment = NULL;
}
if (flacdec->tags) {
gst_element_found_tags_for_pad (GST_ELEMENT (flacdec), flacdec->srcpad,
flacdec->tags);
flacdec->tags = NULL;
}
if (flacdec->pending) {
GST_DEBUG_OBJECT (flacdec,
"pushing pending samples at offset %" G_GINT64_FORMAT " (%"
GST_TIME_FORMAT " + %" GST_TIME_FORMAT ")",
GST_BUFFER_OFFSET (flacdec->pending),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (flacdec->pending)),
GST_TIME_ARGS (GST_BUFFER_DURATION (flacdec->pending)));
/* Pending buffer was always allocated from the seeking thread,
* which means it wasn't gst_buffer_alloc'd. Do so now to let
* downstream negotiation work on older basetransform */
ret = gst_pad_alloc_buffer_and_set_caps (flacdec->srcpad,
GST_BUFFER_OFFSET (flacdec->pending),
GST_BUFFER_SIZE (flacdec->pending),
GST_BUFFER_CAPS (flacdec->pending), &outbuf);
if (ret == GST_FLOW_OK) {
gst_pad_push (flacdec->srcpad, flacdec->pending);
gst_buffer_unref (outbuf);
}
outbuf = flacdec->pending = NULL;
flacdec->segment.last_stop += flacdec->pending_samples;
flacdec->pending_samples = 0;
}
if (flacdec->seeking) {
GST_DEBUG_OBJECT (flacdec, "a pad_alloc would block here, do normal alloc");
outbuf = gst_buffer_new_and_alloc (samples * channels * (width / 8));
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (flacdec->srcpad));
GST_BUFFER_OFFSET (outbuf) = flacdec->segment.last_stop;
} else {
GST_LOG_OBJECT (flacdec, "alloc_buffer_and_set_caps");
ret = gst_pad_alloc_buffer_and_set_caps (flacdec->srcpad,
flacdec->segment.last_stop, samples * channels * (width / 8),
GST_PAD_CAPS (flacdec->srcpad), &outbuf);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (flacdec, "gst_pad_alloc_buffer() returned %s",
gst_flow_get_name (ret));
goto done;
}
}
if (flacdec->cur_granulepos != GST_BUFFER_OFFSET_NONE) {
/* this should be fine since it should be one flac frame per ogg packet */
flacdec->segment.last_stop = flacdec->cur_granulepos - samples;
GST_LOG_OBJECT (flacdec, "granulepos = %" G_GINT64_FORMAT ", samples = %u",
flacdec->cur_granulepos, samples);
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (flacdec->segment.last_stop, GST_SECOND,
frame->header.sample_rate);
/* get next timestamp to calculate the duration */
next = gst_util_uint64_scale_int (flacdec->segment.last_stop + samples,
GST_SECOND, frame->header.sample_rate);
GST_BUFFER_DURATION (outbuf) = next - GST_BUFFER_TIMESTAMP (outbuf);
if (width == 8) {
gint8 *outbuffer = (gint8 *) GST_BUFFER_DATA (outbuf);
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[j][i];
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) GST_BUFFER_DATA (outbuf);
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[j][i];
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) GST_BUFFER_DATA (outbuf);
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[j][i];
}
}
} else {
g_assert_not_reached ();
}
if (!flacdec->seeking) {
GST_DEBUG_OBJECT (flacdec, "pushing %d samples at offset %" G_GINT64_FORMAT
" (%" GST_TIME_FORMAT " + %" GST_TIME_FORMAT ")",
samples, GST_BUFFER_OFFSET (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
if (flacdec->discont) {
GST_DEBUG_OBJECT (flacdec, "marking discont");
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
flacdec->discont = FALSE;
}
ret = gst_pad_push (flacdec->srcpad, outbuf);
GST_DEBUG_OBJECT (flacdec, "returned %s", gst_flow_get_name (ret));
flacdec->segment.last_stop += samples;
} else {
GST_DEBUG_OBJECT (flacdec,
"not pushing %d samples at offset %" G_GINT64_FORMAT
" (in seek)", samples, GST_BUFFER_OFFSET (outbuf));
gst_buffer_replace (&flacdec->pending, outbuf);
gst_buffer_unref (outbuf);
flacdec->pending_samples = samples;
ret = GST_FLOW_OK;
}
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (flacdec, "gst_pad_push() returned %s",
gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the loop function, as we don't
* want to mess up the internal decoder state by returning ABORT when the
* error is in fact non-fatal (like a pad in flushing mode) and we want
* to continue later. So just pretend everything's dandy and act later. */
flacdec->last_flow = ret;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder,
const FLAC__Frame * frame,
const FLAC__int32 * const buffer[], void *client_data)
{
return gst_flac_dec_write (GST_FLAC_DEC (client_data), frame, buffer);
}
static void
gst_flac_dec_loop (GstPad * sinkpad)
{
GstFlacDec *flacdec;
FLAC__StreamDecoderState s;
FLAC__StreamDecoderInitStatus is;
flacdec = GST_FLAC_DEC (GST_OBJECT_PARENT (sinkpad));
GST_LOG_OBJECT (flacdec, "entering loop");
if (flacdec->init) {
GST_DEBUG_OBJECT (flacdec, "initializing new decoder");
is = FLAC__stream_decoder_init_stream (flacdec->seekable_decoder,
gst_flac_dec_read_seekable, gst_flac_dec_seek, gst_flac_dec_tell,
gst_flac_dec_length, gst_flac_dec_eof, gst_flac_dec_write_stream,
gst_flac_dec_metadata_cb, gst_flac_dec_error_cb, flacdec);
if (is != FLAC__STREAM_DECODER_INIT_STATUS_OK)
goto analyze_state;
/* FLAC__seekable_stream_decoder_process_metadata (flacdec->seekable_decoder); */
flacdec->init = FALSE;
}
flacdec->cur_granulepos = GST_BUFFER_OFFSET_NONE;
flacdec->last_flow = GST_FLOW_OK;
GST_LOG_OBJECT (flacdec, "processing single");
FLAC__stream_decoder_process_single (flacdec->seekable_decoder);
analyze_state:
GST_LOG_OBJECT (flacdec, "done processing, checking encoder state");
s = FLAC__stream_decoder_get_state (flacdec->seekable_decoder);
switch (s) {
case FLAC__STREAM_DECODER_SEARCH_FOR_METADATA:
case FLAC__STREAM_DECODER_READ_METADATA:
case FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC:
case FLAC__STREAM_DECODER_READ_FRAME:
{
GST_DEBUG_OBJECT (flacdec, "everything ok");
if (GST_FLOW_IS_FATAL (flacdec->last_flow) ||
flacdec->last_flow == GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (flacdec, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s",
gst_flow_get_name (flacdec->last_flow)));
goto eos_and_pause;
} else if (flacdec->last_flow != GST_FLOW_OK) {
goto pause;
}
/* check if we're at the end of a configured segment */
if (flacdec->segment.stop != -1 &&
flacdec->segment.last_stop > 0 &&
flacdec->segment.last_stop >= flacdec->segment.stop) {
GST_DEBUG_OBJECT (flacdec, "reached end of the configured segment");
if ((flacdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0) {
goto eos_and_pause;
} else {
goto segment_done_and_pause;
}
g_assert_not_reached ();
}
return;
}
case FLAC__STREAM_DECODER_END_OF_STREAM:{
GST_DEBUG_OBJECT (flacdec, "EOS");
FLAC__stream_decoder_reset (flacdec->seekable_decoder);
if ((flacdec->segment.flags & GST_SEEK_FLAG_SEGMENT) != 0) {
if (flacdec->segment.duration > 0) {
flacdec->segment.stop = flacdec->segment.duration;
} else {
flacdec->segment.stop = flacdec->segment.last_stop;
}
goto segment_done_and_pause;
}
goto eos_and_pause;
}
/* gst_flac_dec_read_seekable() returned ABORTED */
case FLAC__STREAM_DECODER_ABORTED:
{
GST_INFO_OBJECT (flacdec, "read aborted: last pull_range flow = %s",
gst_flow_get_name (flacdec->pull_flow));
if (!GST_FLOW_IS_FATAL (flacdec->pull_flow)) {
/* it seems we need to flush the decoder here to reset the decoder
* state after the abort for FLAC__stream_decoder_seek_absolute()
* to work properly */
GST_DEBUG_OBJECT (flacdec, "flushing decoder to reset decoder state");
FLAC__stream_decoder_flush (flacdec->seekable_decoder);
goto pause;
}
/* fall through */
}
case FLAC__STREAM_DECODER_OGG_ERROR:
case FLAC__STREAM_DECODER_SEEK_ERROR:
case FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR:
case FLAC__STREAM_DECODER_UNINITIALIZED:
default:{
/* fixme: this error sucks -- should try to figure out when/if an more
specific error was already sent via the callback */
GST_ELEMENT_ERROR (flacdec, STREAM, DECODE, (NULL),
("%s", FLAC__StreamDecoderStateString[s]));
goto eos_and_pause;
}
}
return;
segment_done_and_pause:
{
gint64 stop_time;
stop_time = gst_util_uint64_scale_int (flacdec->segment.stop,
GST_SECOND, flacdec->sample_rate);
GST_DEBUG_OBJECT (flacdec, "posting SEGMENT_DONE message, stop time %"
GST_TIME_FORMAT, GST_TIME_ARGS (stop_time));
gst_element_post_message (GST_ELEMENT (flacdec),
gst_message_new_segment_done (GST_OBJECT (flacdec),
GST_FORMAT_TIME, stop_time));
goto pause;
}
eos_and_pause:
{
GST_DEBUG_OBJECT (flacdec, "sending EOS event");
flacdec->running = FALSE;
gst_pad_push_event (flacdec->srcpad, gst_event_new_eos ());
/* fall through to pause */
}
pause:
{
GST_DEBUG_OBJECT (flacdec, "pausing");
gst_pad_pause_task (sinkpad);
return;
}
}
static gboolean
gst_flac_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstFlacDec *dec;
gboolean res;
dec = GST_FLAC_DEC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:{
if (dec->init == FALSE) {
FLAC__stream_decoder_flush (dec->stream_decoder);
gst_adapter_clear (dec->adapter);
}
res = gst_pad_push_event (dec->srcpad, event);
break;
}
case GST_EVENT_NEWSEGMENT:{
GstFormat fmt;
gboolean update;
gdouble rate, applied_rate;
gint64 cur, stop, time;
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
&fmt, &cur, &stop, &time);
if (fmt == GST_FORMAT_TIME) {
GstFormat dformat = GST_FORMAT_DEFAULT;
GST_DEBUG_OBJECT (dec, "newsegment event in TIME format => framed");
dec->framed = TRUE;
res = gst_pad_push_event (dec->srcpad, event);
/* this won't work for the first newsegment event though ... */
if (gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME, cur,
&dformat, &cur) && cur != -1 &&
gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME, stop,
&dformat, &stop) && stop != -1) {
gst_segment_set_newsegment_full (&dec->segment, update, rate,
applied_rate, dformat, cur, stop, time);
GST_DEBUG_OBJECT (dec, "segment %" GST_SEGMENT_FORMAT, &dec->segment);
} else {
GST_WARNING_OBJECT (dec, "couldn't convert time => samples");
}
} else if (fmt == GST_FORMAT_BYTES || TRUE) {
GST_DEBUG_OBJECT (dec, "newsegment event in %s format => not framed",
gst_format_get_name (fmt));
dec->framed = FALSE;
/* prepare generic newsegment event, for some reason our metadata
* callback where we usually set this up is not being called in
* push mode */
dec->start_segment = gst_event_new_new_segment (FALSE, 1.0,
GST_FORMAT_TIME, 0, -1, 0);
gst_event_unref (event);
res = TRUE;
}
break;
}
case GST_EVENT_EOS:{
GST_LOG_OBJECT (dec, "EOS, with %u bytes available in adapter",
gst_adapter_available (dec->adapter));
if (dec->init == FALSE) {
if (gst_adapter_available (dec->adapter) > 0) {
FLAC__stream_decoder_process_until_end_of_stream
(dec->stream_decoder);
}
FLAC__stream_decoder_flush (dec->stream_decoder);
}
gst_adapter_clear (dec->adapter);
res = gst_pad_push_event (dec->srcpad, event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (dec);
return res;
}
static GstFlowReturn
gst_flac_dec_chain (GstPad * pad, GstBuffer * buf)
{
FLAC__StreamDecoderInitStatus s;
GstFlacDec *dec;
gboolean got_audio_frame;
dec = GST_FLAC_DEC (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (dec, "buffer with ts=%" GST_TIME_FORMAT ", end_offset=%"
G_GINT64_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_BUFFER_OFFSET_END (buf));
if (dec->init) {
GST_DEBUG_OBJECT (dec, "initializing decoder");
s = FLAC__stream_decoder_init_stream (dec->stream_decoder,
gst_flac_dec_read_stream, NULL, NULL, NULL, NULL,
gst_flac_dec_write_stream, gst_flac_dec_metadata_cb,
gst_flac_dec_error_cb, dec);
if (s != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
GST_ELEMENT_ERROR (GST_ELEMENT (dec), LIBRARY, INIT, (NULL), (NULL));
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "initialized (framed=%d)", dec->framed);
dec->init = FALSE;
}
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
/* Clear the adapter and the decoder */
gst_adapter_clear (dec->adapter);
FLAC__stream_decoder_flush (dec->stream_decoder);
}
if (dec->framed) {
gint64 unused;
/* check if this is a flac audio frame (rather than a header or junk) */
got_audio_frame = gst_flac_dec_scan_got_frame (dec, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf), &unused);
/* oggdemux will set granulepos in OFFSET_END instead of timestamp */
if (got_audio_frame && !GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
dec->cur_granulepos = GST_BUFFER_OFFSET_END (buf);
}
} else {
dec->cur_granulepos = GST_BUFFER_OFFSET_NONE;
got_audio_frame = TRUE;
}
gst_adapter_push (dec->adapter, buf);
buf = NULL;
dec->last_flow = GST_FLOW_OK;
if (!dec->framed) {
/* wait until we have at least 64kB because libflac's StreamDecoder
* interface is a bit dumb it seems (if we don't have as much data as
* it wants it will call our read callback repeatedly and the only
* way to stop that is to error out or EOS, which will affect the
* decoder state). And the decoder seems to always ask for MAX_BLOCK_SIZE
* bytes rather than the max. block size from the header). Requiring
* MAX_BLOCK_SIZE bytes here should make sure it always gets enough data
* to decode at least one block */
while (gst_adapter_available (dec->adapter) >= FLAC__MAX_BLOCK_SIZE &&
dec->last_flow == GST_FLOW_OK) {
GST_LOG_OBJECT (dec, "%u bytes available",
gst_adapter_available (dec->adapter));
if (!FLAC__stream_decoder_process_single (dec->stream_decoder)) {
GST_DEBUG_OBJECT (dec, "process_single failed");
break;
}
if (FLAC__stream_decoder_get_state (dec->stream_decoder) ==
FLAC__STREAM_DECODER_ABORTED) {
GST_WARNING_OBJECT (dec, "Read callback caused internal abort");
dec->last_flow = GST_FLOW_ERROR;
break;
}
}
} else if (dec->framed && got_audio_frame) {
/* framed - there should always be enough data to decode something */
GST_LOG_OBJECT (dec, "%u bytes available",
gst_adapter_available (dec->adapter));
if (!FLAC__stream_decoder_process_single (dec->stream_decoder)) {
GST_DEBUG_OBJECT (dec, "process_single failed");
}
} else {
GST_DEBUG_OBJECT (dec, "don't have all headers yet");
}
return dec->last_flow;
}
static gboolean
gst_flac_dec_convert_sink (GstFlacDec * dec, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = TRUE;
if (dec->width == 0 || dec->channels == 0 || dec->sample_rate == 0) {
/* no frame decoded yet */
GST_DEBUG_OBJECT (dec, "cannot convert: not set up yet");
return FALSE;
}
switch (src_format) {
case GST_FORMAT_BYTES:{
res = FALSE;
break;
}
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
res = FALSE;
break;
case GST_FORMAT_TIME:
/* granulepos = sample */
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
dec->sample_rate);
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
res = FALSE;
break;
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale_int (src_value,
dec->sample_rate, GST_SECOND);
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
return res;
}
static const GstQueryType *
gst_flac_dec_get_sink_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_CONVERT,
0,
};
return types;
}
static gboolean
gst_flac_dec_sink_query (GstPad * pad, GstQuery * query)
{
GstFlacDec *dec;
gboolean res = FALSE;
dec = GST_FLAC_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "%s query", GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res = gst_flac_dec_convert_sink (dec, src_fmt, src_val, &dest_fmt,
&dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
GST_LOG_OBJECT (dec, "conversion %s", (res) ? "ok" : "FAILED");
break;
}
default:{
res = gst_pad_query_default (pad, query);
break;
}
}
gst_object_unref (dec);
return res;
}
static gboolean
gst_flac_dec_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
GstFlacDec *flacdec = GST_FLAC_DEC (GST_PAD_PARENT (pad));
gboolean res = TRUE;
guint bytes_per_sample;
guint scale = 1;
if (flacdec->width == 0 || flacdec->channels == 0 ||
flacdec->sample_rate == 0) {
/* no frame decoded yet */
GST_DEBUG_OBJECT (flacdec, "cannot convert: not set up yet");
return FALSE;
}
bytes_per_sample = flacdec->channels * (flacdec->width / 8);
switch (src_format) {
case GST_FORMAT_BYTES:{
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value =
gst_util_uint64_scale_int (src_value, 1, bytes_per_sample);
break;
case GST_FORMAT_TIME:
{
gint byterate = bytes_per_sample * flacdec->sample_rate;
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
byterate);
break;
}
default:
res = FALSE;
}
break;
}
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
flacdec->sample_rate);
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = bytes_per_sample;
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale_int (src_value,
scale * flacdec->sample_rate, GST_SECOND);
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
return res;
}
static const GstQueryType *
gst_flac_dec_get_src_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
GST_QUERY_SEEKING,
0,
};
return types;
}
static gboolean
gst_flac_dec_src_query (GstPad * pad, GstQuery * query)
{
GstFlacDec *flacdec;
gboolean res = TRUE;
GstPad *peer;
flacdec = GST_FLAC_DEC (gst_pad_get_parent (pad));
peer = gst_pad_get_peer (flacdec->sinkpad);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:{
GstFormat fmt;
gint64 pos;
gst_query_parse_position (query, &fmt, NULL);
/* there might be a demuxer in front of us who can handle this */
if (fmt == GST_FORMAT_TIME && (res = gst_pad_query (peer, query)))
break;
if (fmt != GST_FORMAT_DEFAULT) {
if (!gst_flac_dec_convert_src (flacdec->srcpad, GST_FORMAT_DEFAULT,
flacdec->segment.last_stop, &fmt, &pos)) {
GST_DEBUG_OBJECT (flacdec, "failed to convert position into %s "
"format", gst_format_get_name (fmt));
res = FALSE;
goto done;
}
} else {
pos = flacdec->segment.last_stop;
}
gst_query_set_position (query, fmt, pos);
GST_DEBUG_OBJECT (flacdec, "returning position %" G_GUINT64_FORMAT
" (format: %s)", pos, gst_format_get_name (fmt));
res = TRUE;
break;
}
case GST_QUERY_DURATION:{
GstFormat fmt;
gint64 len;
gst_query_parse_duration (query, &fmt, NULL);
/* try any demuxers before us first */
if (fmt == GST_FORMAT_TIME && peer && gst_pad_query (peer, query)) {
gst_query_parse_duration (query, NULL, &len);
GST_DEBUG_OBJECT (flacdec, "peer returned duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (len));
res = TRUE;
goto done;
}
if (flacdec->segment.duration == 0 || flacdec->segment.duration == -1) {
GST_DEBUG_OBJECT (flacdec, "duration not known yet");
res = FALSE;
goto done;
}
/* convert total number of samples to request format */
if (fmt != GST_FORMAT_DEFAULT) {
if (!gst_flac_dec_convert_src (flacdec->srcpad, GST_FORMAT_DEFAULT,
flacdec->segment.duration, &fmt, &len)) {
GST_DEBUG_OBJECT (flacdec, "failed to convert duration into %s "
"format", gst_format_get_name (fmt));
res = FALSE;
goto done;
}
} else {
len = flacdec->segment.duration;
}
gst_query_set_duration (query, fmt, len);
GST_DEBUG_OBJECT (flacdec, "returning duration %" G_GUINT64_FORMAT
" (format: %s)", len, gst_format_get_name (fmt));
res = TRUE;
break;
}
case GST_QUERY_CONVERT:{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res = gst_flac_dec_convert_src (pad, src_fmt, src_val, &dest_fmt,
&dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
break;
}
case GST_QUERY_SEEKING:{
GstFormat fmt;
gboolean seekable = FALSE;
res = TRUE;
/* If upstream can handle the query we're done */
seekable = gst_pad_peer_query (flacdec->sinkpad, query);
if (seekable)
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
if (seekable)
goto done;
gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
if ((fmt != GST_FORMAT_TIME && fmt != GST_FORMAT_DEFAULT) ||
!flacdec->seekable_decoder) {
gst_query_set_seeking (query, fmt, FALSE, -1, -1);
} else {
gst_query_set_seeking (query, GST_FORMAT_TIME, TRUE, 0, -1);
}
break;
}
default:{
res = gst_pad_query_default (pad, query);
break;
}
}
done:
if (peer)
gst_object_unref (peer);
gst_object_unref (flacdec);
return res;
}
static gboolean
gst_flac_dec_handle_seek_event (GstFlacDec * flacdec, GstEvent * event)
{
FLAC__bool seek_ok;
GstSeekFlags seek_flags;
GstSeekType start_type;
GstSeekType stop_type;
GstSegment segment;
GstFormat seek_format;
gboolean only_update = FALSE;
gboolean flush;
gdouble rate;
gint64 start, last_stop;
gint64 stop;
if (flacdec->seekable_decoder == NULL) {
GST_DEBUG_OBJECT (flacdec, "seeking in streaming mode not implemented yet");
return FALSE;
}
gst_event_parse_seek (event, &rate, &seek_format, &seek_flags, &start_type,
&start, &stop_type, &stop);
if (seek_format != GST_FORMAT_DEFAULT && seek_format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (flacdec,
"seeking is only supported in TIME or DEFAULT format");
return FALSE;
}
if (rate < 0.0) {
GST_DEBUG_OBJECT (flacdec,
"only forward playback supported, rate %f not allowed", rate);
return FALSE;
}
if (seek_format != GST_FORMAT_DEFAULT) {
GstFormat target_format = GST_FORMAT_DEFAULT;
if (start_type != GST_SEEK_TYPE_NONE &&
!gst_flac_dec_convert_src (flacdec->srcpad, seek_format, start,
&target_format, &start)) {
GST_DEBUG_OBJECT (flacdec, "failed to convert start to DEFAULT format");
return FALSE;
}
if (stop_type != GST_SEEK_TYPE_NONE &&
!gst_flac_dec_convert_src (flacdec->srcpad, seek_format, stop,
&target_format, &stop)) {
GST_DEBUG_OBJECT (flacdec, "failed to convert stop to DEFAULT format");
return FALSE;
}
}
flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);
if (flush) {
/* flushing seek, clear the pipeline of stuff, we need a newsegment after
* this. */
GST_DEBUG_OBJECT (flacdec, "flushing");
gst_pad_push_event (flacdec->sinkpad, gst_event_new_flush_start ());
gst_pad_push_event (flacdec->srcpad, gst_event_new_flush_start ());
} else {
/* non flushing seek, pause the task */
GST_DEBUG_OBJECT (flacdec, "stopping task");
gst_pad_stop_task (flacdec->sinkpad);
}
/* acquire the stream lock, this either happens when the streaming thread
* stopped because of the flush or when the task is paused after the loop
* function finished an iteration, which can never happen when it's blocked
* downstream in PAUSED, for example */
GST_PAD_STREAM_LOCK (flacdec->sinkpad);
/* save a segment copy until we know the seek worked. The idea is that
* when the seek fails, we want to restore with what we were doing. */
segment = flacdec->segment;
/* update the segment with the seek values, last_stop will contain the new
* position we should seek to */
gst_segment_set_seek (&flacdec->segment, rate, GST_FORMAT_DEFAULT,
seek_flags, start_type, start, stop_type, stop, &only_update);
GST_DEBUG_OBJECT (flacdec,
"configured segment: [%" G_GINT64_FORMAT "-%" G_GINT64_FORMAT
"] = [%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "]",
flacdec->segment.start, flacdec->segment.stop,
GST_TIME_ARGS (flacdec->segment.start * GST_SECOND /
flacdec->sample_rate),
GST_TIME_ARGS (flacdec->segment.stop * GST_SECOND /
flacdec->sample_rate));
GST_DEBUG_OBJECT (flacdec, "performing seek to sample %" G_GINT64_FORMAT,
flacdec->segment.last_stop);
/* flush sinkpad again because we need to pull and push buffers while doing
* the seek */
if (flush) {
GST_DEBUG_OBJECT (flacdec, "flushing stop");
gst_pad_push_event (flacdec->sinkpad, gst_event_new_flush_stop ());
gst_pad_push_event (flacdec->srcpad, gst_event_new_flush_stop ());
}
/* mark ourselves as seeking because the above lines will trigger some
* callbacks that need to behave differently when seeking */
flacdec->seeking = TRUE;
GST_LOG_OBJECT (flacdec, "calling seek_absolute");
seek_ok = FLAC__stream_decoder_seek_absolute (flacdec->seekable_decoder,
flacdec->segment.last_stop);
GST_LOG_OBJECT (flacdec, "done with seek_absolute, seek_ok=%d", seek_ok);
flacdec->seeking = FALSE;
GST_DEBUG_OBJECT (flacdec, "performed seek to sample %" G_GINT64_FORMAT,
flacdec->segment.last_stop);
if (!seek_ok) {
GST_WARNING_OBJECT (flacdec, "seek failed");
/* seek failed, restore the segment and start streaming again with
* the previous segment values */
flacdec->segment = segment;
} else if (!flush && flacdec->running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the last_stop. */
GST_DEBUG_OBJECT (flacdec, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, segment.start, segment.last_stop);
/* convert the old segment values to time to close the old segment */
start = gst_util_uint64_scale_int (segment.start, GST_SECOND,
flacdec->sample_rate);
last_stop =
gst_util_uint64_scale_int (segment.last_stop, GST_SECOND,
flacdec->sample_rate);
/* queue the segment for sending in the stream thread, start and time are
* always the same. */
if (flacdec->close_segment)
gst_event_unref (flacdec->close_segment);
flacdec->close_segment =
gst_event_new_new_segment_full (TRUE,
segment.rate, segment.applied_rate, GST_FORMAT_TIME,
start, last_stop, start);
}
if (seek_ok) {
/* seek succeeded, flacdec->segment contains the new positions */
GST_DEBUG_OBJECT (flacdec, "seek successful");
}
/* convert the (new) segment values to time, we will need them to generate the
* new segment events. */
start = gst_util_uint64_scale_int (flacdec->segment.start, GST_SECOND,
flacdec->sample_rate);
last_stop = gst_util_uint64_scale_int (flacdec->segment.last_stop, GST_SECOND,
flacdec->sample_rate);
/* for deriving a stop position for the playback segment from the seek
* segment, we must take the duration when the stop is not set */
if (flacdec->segment.stop != -1)
stop = gst_util_uint64_scale_int (flacdec->segment.stop, GST_SECOND,
flacdec->sample_rate);
else
stop = gst_util_uint64_scale_int (flacdec->segment.duration, GST_SECOND,
flacdec->sample_rate);
/* notify start of new segment when we were asked to do so. */
if (flacdec->segment.flags & GST_SEEK_FLAG_SEGMENT) {
/* last_stop contains the position we start from */
gst_element_post_message (GST_ELEMENT (flacdec),
gst_message_new_segment_start (GST_OBJECT (flacdec),
GST_FORMAT_TIME, last_stop));
}
/* if the seek was ok or (when it failed) we are flushing, we need to send out
* a new segment. If we did not flush and the seek failed, we simply do
* nothing here and continue where we were. */
if (seek_ok || flush) {
GST_DEBUG_OBJECT (flacdec, "Creating newsegment from %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT, GST_TIME_ARGS (last_stop),
GST_TIME_ARGS (stop));
/* now replace the old segment so that we send it in the stream thread the
* next time it is scheduled. */
if (flacdec->start_segment)
gst_event_unref (flacdec->start_segment);
flacdec->start_segment =
gst_event_new_new_segment_full (FALSE,
flacdec->segment.rate, flacdec->segment.applied_rate, GST_FORMAT_TIME,
last_stop, stop, last_stop);
}
/* we'll generate a discont on the next buffer */
flacdec->discont = TRUE;
/* the task is running again now */
flacdec->running = TRUE;
gst_pad_start_task (flacdec->sinkpad,
(GstTaskFunction) gst_flac_dec_loop, flacdec->sinkpad);
GST_PAD_STREAM_UNLOCK (flacdec->sinkpad);
return seek_ok;
}
static gboolean
gst_flac_dec_src_event (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstFlacDec *flacdec;
flacdec = GST_FLAC_DEC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
GST_DEBUG_OBJECT (flacdec, "received seek event %p", event);
/* first, see if we're before a demuxer that
* might handle the seek for us */
gst_event_ref (event);
res = gst_pad_event_default (pad, event);
/* if not, try to handle it ourselves */
if (!res) {
GST_DEBUG_OBJECT (flacdec, "default failed, handling ourselves");
res = gst_flac_dec_handle_seek_event (flacdec, event);
}
gst_event_unref (event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (flacdec);
return res;
}
static gboolean
gst_flac_dec_sink_activate (GstPad * sinkpad)
{
if (gst_pad_check_pull_range (sinkpad))
return gst_pad_activate_pull (sinkpad, TRUE);
return gst_pad_activate_push (sinkpad, TRUE);
}
static gboolean
gst_flac_dec_sink_activate_push (GstPad * sinkpad, gboolean active)
{
GstFlacDec *dec = GST_FLAC_DEC (GST_OBJECT_PARENT (sinkpad));
if (active)
gst_flac_dec_setup_stream_decoder (dec);
return TRUE;
}
static gboolean
gst_flac_dec_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
gboolean res;
if (active) {
GstFlacDec *flacdec;
flacdec = GST_FLAC_DEC (GST_PAD_PARENT (sinkpad));
flacdec->offset = 0;
gst_flac_dec_setup_seekable_decoder (flacdec);
flacdec->running = TRUE;
res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_flac_dec_loop,
sinkpad);
} else {
res = gst_pad_stop_task (sinkpad);
}
return res;
}
static GstStateChangeReturn
gst_flac_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstFlacDec *flacdec = GST_FLAC_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
flacdec->seeking = FALSE;
flacdec->channels = 0;
flacdec->depth = 0;
flacdec->width = 0;
flacdec->sample_rate = 0;
gst_segment_init (&flacdec->segment, GST_FORMAT_DEFAULT);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_segment_init (&flacdec->segment, GST_FORMAT_UNDEFINED);
gst_flac_dec_reset_decoders (flacdec);
break;
default:
break;
}
return ret;
}
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