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gstreamer/ext/flac/gstflacdec.c
Tim-Philipp Müller 3d01b9f398 flacdec: get rid of granulepos handling 
Leave that to the parser or demuxer. There's still some
code for operating in DEFAULT (samples) format, but that
will be removed later.
2011-09-28 19:10:27 +01:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2006,2011> 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-0.11 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* |[
* gst-launch-0.11 souphttpsrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
* ]|
* </refsect2>
*/
#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 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_sink_activate (GstPad * sinkpad);
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_decoder (GstFlacDec * flacdec);
static FLAC__StreamDecoderReadStatus
gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder,
FLAC__byte buffer[], size_t * bytes, 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);
#define gst_flac_dec_parent_class parent_class
G_DEFINE_TYPE (GstFlacDec, gst_flac_dec, GST_TYPE_ELEMENT);
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define FORMATS "{ S8LE, S16LE, S32LE } "
#else
#define FORMATS "{ S8BE, S16BE, S32BE } "
#endif
/* 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, " \
"format = (string) " FORMATS ", " \
"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_class_init (GstFlacDecClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *gobject_class;
gstelement_class = (GstElementClass *) klass;
gobject_class = (GObjectClass *) klass;
GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder");
gobject_class->finalize = gst_flac_dec_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_flac_dec_change_state);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&flac_dec_src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&flac_dec_sink_factory));
gst_element_class_set_details_simple (gstelement_class, "FLAC audio decoder",
"Codec/Decoder/Audio",
"Decodes FLAC lossless audio streams", "Wim Taymans <wim@fluendo.com>");
}
static void
gst_flac_dec_init (GstFlacDec * flacdec)
{
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_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_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)
{
/* Clean up the decoder */
if (flacdec->decoder) {
FLAC__stream_decoder_delete (flacdec->decoder);
flacdec->decoder = NULL;
}
if (flacdec->adapter) {
gst_adapter_clear (flacdec->adapter);
g_object_unref (flacdec->adapter);
flacdec->adapter = NULL;
}
if (flacdec->tags) {
gst_tag_list_free (flacdec->tags);
flacdec->tags = NULL;
}
flacdec->segment.position = 0;
flacdec->init = TRUE;
}
static void
gst_flac_dec_setup_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->decoder = FLAC__stream_decoder_new ();
/* no point calculating since it's never checked here */
FLAC__stream_decoder_set_md5_checking (dec->decoder, false);
FLAC__stream_decoder_set_metadata_respond (dec->decoder,
FLAC__METADATA_TYPE_VORBIS_COMMENT);
FLAC__stream_decoder_set_metadata_respond (dec->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] & 0xFC) != 0xF8)
return FALSE;
if (data[1] & 1) {
GST_WARNING_OBJECT (flacdec, "Variable block size FLAC unsupported");
return FALSE;
}
bs = (data[2] & 0xF0) >> 4; /* blocksize marker */
sr = (data[2] & 0x0F); /* samplerate marker */
ca = (data[3] & 0xF0) >> 4; /* 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 (bs == 0 || sr == 0x0F || 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;
/* FIXME: This is can be 36 bit if variable block size is used,
* fortunately not encoder supports this yet and we check for that
* above.
*/
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]) {
GST_LOG_OBJECT (flacdec, "invalid checksum");
return FALSE;
}
if (flacdec->min_blocksize == flacdec->max_blocksize) {
*last_sample_num = (val + 1) * flacdec->min_blocksize;
} else {
*last_sample_num = 0; /* FIXME: + length of last block in samples */
}
/* FIXME: only valid for fixed block size streams */
GST_DEBUG_OBJECT (flacdec, "frame number: %" G_GINT64_FORMAT,
*last_sample_num);
if (flacdec->sample_rate > 0 && *last_sample_num != 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;
}
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;
guint depth;
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 = depth = metadata->data.stream_info.bits_per_sample;
flacdec->channels = metadata->data.stream_info.channels;
if (depth < 9)
flacdec->width = 8;
else if (depth < 17)
flacdec->width = 16;
else
flacdec->width = 32;
GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u",
flacdec->min_blocksize, flacdec->max_blocksize);
GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u",
flacdec->sample_rate, flacdec->channels);
GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth,
flacdec->width);
GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
/* in framed mode the demuxer/parser upstream has already pushed a
* newsegment event in TIME format which we've passed on */
if (samples > 0 && !flacdec->framed) {
gint64 duration;
GstSegment seg;
flacdec->segment.duration = samples;
/* convert duration to time */
duration = gst_util_uint64_scale_int (samples, GST_SECOND,
flacdec->sample_rate);
gst_segment_init (&seg, GST_FORMAT_TIME);
seg.rate = flacdec->segment.rate;
seg.applied_rate = flacdec->segment.applied_rate;
seg.start = 0;
seg.stop = duration;
seg.time = 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__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 sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstClockTime next;
gpointer data;
gsize size;
const gchar *format;
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.position >= 0 &&
flacdec->segment.position + samples > flacdec->segment.stop) {
samples = flacdec->segment.stop - flacdec->segment.position;
GST_DEBUG_OBJECT (flacdec,
"clipping last buffer to %d samples because of segment", samples);
}
if (depth == 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)
depth = 8;
else if (depth < 17)
depth = 16;
else
depth = 32;
}
switch (depth) {
case 8:
width = 8;
format = GST_AUDIO_NE (S8);
break;
case 12:
case 16:
width = 16;
format = GST_AUDIO_NE (S16);
break;
case 20:
case 24:
case 32:
width = 32;
format = GST_AUDIO_NE (S32);
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->sample_rate != 0) {
sample_rate = flacdec->sample_rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
if (!gst_pad_has_current_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",
"format", G_TYPE_STRING, format,
"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 = sample_rate;
gst_pad_set_caps (flacdec->srcpad, caps);
gst_caps_unref (caps);
}
if (flacdec->tags) {
gst_element_found_tags_for_pad (GST_ELEMENT (flacdec), flacdec->srcpad,
flacdec->tags);
flacdec->tags = NULL;
}
GST_LOG_OBJECT (flacdec, "alloc_buffer_and_set_caps");
outbuf = gst_buffer_new_allocate (NULL, samples * channels * (width / 8), 0);
GST_BUFFER_OFFSET (outbuf) = flacdec->segment.position;
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (flacdec->segment.position, GST_SECOND,
frame->header.sample_rate);
/* get next timestamp to calculate the duration */
next = gst_util_uint64_scale_int (flacdec->segment.position + samples,
GST_SECOND, frame->header.sample_rate);
GST_BUFFER_DURATION (outbuf) = next - GST_BUFFER_TIMESTAMP (outbuf);
data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) data;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[j][i];
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) data;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[j][i];
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) data;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[j][i];
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, data, size);
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)));
ret = gst_pad_push (flacdec->srcpad, outbuf);
GST_DEBUG_OBJECT (flacdec, "returned %s", gst_flow_get_name (ret));
flacdec->segment.position += samples;
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 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_CAPS:
res = TRUE;
gst_event_unref (event);
break;
case GST_EVENT_FLUSH_STOP:{
if (dec->init == FALSE) {
FLAC__stream_decoder_flush (dec->decoder);
gst_adapter_clear (dec->adapter);
}
res = gst_pad_push_event (dec->srcpad, event);
break;
}
case GST_EVENT_SEGMENT:{
GstSegment seg;
gint64 start, stop;
gst_event_copy_segment (event, &seg);
if (seg.format == 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, seg.start,
&dformat, &start) && start != -1 &&
gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME, seg.stop,
&dformat, &stop) && stop != -1) {
seg.start = start;
seg.stop = stop;
dec->segment = seg;
GST_DEBUG_OBJECT (dec, "segment %" GST_SEGMENT_FORMAT, &dec->segment);
} else {
GST_WARNING_OBJECT (dec, "couldn't convert time => samples");
}
} else if (seg.format == GST_FORMAT_BYTES || TRUE) {
/* FIXME: error out or post warning, we require parsed input */
gst_event_unref (event);
res = FALSE;
}
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->decoder);
}
FLAC__stream_decoder_flush (dec->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 gboolean
gst_flac_dec_chain_parse_headers (GstFlacDec * dec)
{
guint8 marker[4];
guint avail, off;
avail = gst_adapter_available (dec->adapter);
if (avail < 4)
return FALSE;
gst_adapter_copy (dec->adapter, marker, 0, 4);
if (strncmp ((const gchar *) marker, "fLaC", 4) != 0) {
GST_ERROR_OBJECT (dec, "Unexpected header, expected fLaC header");
return TRUE; /* abort header parsing */
}
GST_DEBUG_OBJECT (dec, "fLaC header : len 4 @ %7u", 0);
off = 4;
while (avail > (off + 1 + 3)) {
gboolean is_last;
guint8 mb_hdr[4];
guint len, block_type;
gst_adapter_copy (dec->adapter, mb_hdr, off, 4);
is_last = ((mb_hdr[0] & 0x80) == 0x80);
block_type = mb_hdr[0] & 0x7f;
len = GST_READ_UINT24_BE (mb_hdr + 1);
GST_DEBUG_OBJECT (dec, "Metadata block type %u: len %7u + 4 @ %7u%s",
block_type, len, off, (is_last) ? " (last)" : "");
off += 4 + len;
if (is_last)
break;
if (off >= avail) {
GST_LOG_OBJECT (dec, "Need more data: next offset %u > avail %u", off,
avail);
return FALSE;
}
}
/* want metadata blocks plus at least one frame */
return (off + FLAC__MAX_BLOCK_SIZE >= avail);
}
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 ", offset=%" G_GINT64_FORMAT
", end_offset=%" G_GINT64_FORMAT ", size=%" G_GSIZE_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_BUFFER_OFFSET (buf),
GST_BUFFER_OFFSET_END (buf), gst_buffer_get_size (buf));
if (dec->init) {
GST_DEBUG_OBJECT (dec, "initializing decoder");
s = FLAC__stream_decoder_init_stream (dec->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;
} else 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->decoder);
}
/* FIXME: should always be framed */
if (dec->framed) {
gint64 unused;
guint8 *data;
gsize size;
/* check if this is a flac audio frame (rather than a header or junk) */
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
got_audio_frame = gst_flac_dec_scan_got_frame (dec, data, size, &unused);
gst_buffer_unmap (buf, data, size);
if (G_LIKELY (got_audio_frame)) {
GstFormat dformat = GST_FORMAT_DEFAULT; /* FIXME: remove var */
gint64 position;
/* upstream (e.g. demuxer) presents us time,
* convert to default samples */
gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf), &dformat, &position);
dec->segment.position = position;
}
} else {
got_audio_frame = TRUE;
}
gst_adapter_push (dec->adapter, buf);
buf = NULL;
dec->last_flow = GST_FLOW_OK;
if (!dec->framed) {
if (G_UNLIKELY (!dec->got_headers)) {
if (!gst_flac_dec_chain_parse_headers (dec)) {
GST_LOG_OBJECT (dec, "don't have metadata blocks yet, need more data");
goto out;
}
GST_INFO_OBJECT (dec, "have all metadata blocks now");
dec->got_headers = TRUE;
}
/* 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->decoder)) {
GST_DEBUG_OBJECT (dec, "process_single failed");
break;
}
if (FLAC__stream_decoder_get_state (dec->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 (G_UNLIKELY (!dec->got_headers)) {
/* The first time we get audio data, we know we got all the headers.
* We then loop until all the metadata is processed, then do an extra
* "process_single" step for the audio frame. */
GST_DEBUG_OBJECT (dec,
"First audio frame, ensuring all metadata is processed");
if (!FLAC__stream_decoder_process_until_end_of_metadata (dec->decoder)) {
GST_DEBUG_OBJECT (dec, "process_until_end_of_metadata failed");
}
GST_DEBUG_OBJECT (dec,
"All metadata is now processed, reading to process audio data");
dec->got_headers = TRUE;
}
if (!FLAC__stream_decoder_process_single (dec->decoder)) {
GST_DEBUG_OBJECT (dec, "process_single failed");
}
} else {
GST_DEBUG_OBJECT (dec, "don't have all headers yet");
}
out:
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_round (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.position, &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.position;
}
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 or parsers before us first */
if ((fmt == GST_FORMAT_TIME || fmt == GST_FORMAT_DEFAULT) &&
peer != NULL && 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;
}
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_sink_activate (GstPad * sinkpad)
{
GST_DEBUG_OBJECT (sinkpad, "activating push");
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_decoder (dec);
dec->got_headers = FALSE;
}
return TRUE;
}
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->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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