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gstreamer/ext/flac/gstflacdec.c
Tim-Philipp Müller 230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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-1.0 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* |[
* gst-launch-1.0 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/tag/tag.h>
/* Taken from http://flac.sourceforge.net/format.html#frame_header */
static const GstAudioChannelPosition channel_positions[8][8] = {
{GST_AUDIO_CHANNEL_POSITION_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_LFE1,
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_LFE1,
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_LFE1,
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 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);
static void gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard);
static gboolean gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static gboolean gst_flac_dec_start (GstAudioDecoder * dec);
static gboolean gst_flac_dec_stop (GstAudioDecoder * dec);
static GstFlowReturn gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec,
GstBuffer * buf);
G_DEFINE_TYPE (GstFlacDec, gst_flac_dec, GST_TYPE_AUDIO_DECODER);
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define FORMATS "{ S8LE, S16LE, S24_32LE, S32LE } "
#else
#define FORMATS "{ S8BE, S16BE, S24_32BE, S32BE } "
#endif
#define GST_FLAC_DEC_SRC_CAPS \
"audio/x-raw, " \
"format = (string) " FORMATS ", " \
"layout = (string) interleaved, " \
"rate = (int) [ 1, 655350 ], " \
"channels = (int) [ 1, 8 ]"
#define GST_FLAC_DEC_SINK_CAPS \
"audio/x-flac, " \
"framed = (boolean) true, " \
"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 (GST_FLAC_DEC_SINK_CAPS));
static void
gst_flac_dec_class_init (GstFlacDecClass * klass)
{
GstAudioDecoderClass *audiodecoder_class;
GstElementClass *gstelement_class;
audiodecoder_class = (GstAudioDecoderClass *) klass;
gstelement_class = (GstElementClass *) klass;
GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder");
audiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_flac_dec_stop);
audiodecoder_class->start = GST_DEBUG_FUNCPTR (gst_flac_dec_start);
audiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_flac_dec_flush);
audiodecoder_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_dec_set_format);
audiodecoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_flac_dec_handle_frame);
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_static_metadata (gstelement_class, "FLAC audio decoder",
"Codec/Decoder/Audio", "Decodes FLAC lossless audio streams",
"Tim-Philipp Müller <tim@centricular.net>, "
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_flac_dec_init (GstFlacDec * flacdec)
{
/* nothing to do here */
}
static gboolean
gst_flac_dec_start (GstAudioDecoder * audio_dec)
{
FLAC__StreamDecoderInitStatus s;
GstFlacDec *dec;
dec = GST_FLAC_DEC (audio_dec);
dec->adapter = gst_adapter_new ();
dec->decoder = FLAC__stream_decoder_new ();
gst_audio_info_init (&dec->info);
dec->depth = 0;
/* no point calculating MD5 since it's never checked here */
FLAC__stream_decoder_set_md5_checking (dec->decoder, false);
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 FALSE;
}
dec->got_headers = FALSE;
return TRUE;
}
static gboolean
gst_flac_dec_stop (GstAudioDecoder * dec)
{
GstFlacDec *flacdec = GST_FLAC_DEC (dec);
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;
}
return TRUE;
}
static gboolean
gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
const GValue *headers;
GstFlacDec *flacdec;
GstStructure *s;
guint i, num;
flacdec = GST_FLAC_DEC (dec);
GST_LOG_OBJECT (dec, "sink caps: %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
headers = gst_structure_get_value (s, "streamheader");
if (headers == NULL || !GST_VALUE_HOLDS_ARRAY (headers)) {
GST_WARNING_OBJECT (dec, "no 'streamheader' field in input caps, try "
"adding a flacparse element upstream");
return FALSE;
}
if (gst_adapter_available (flacdec->adapter) > 0) {
GST_WARNING_OBJECT (dec, "unexpected data left in adapter");
gst_adapter_clear (flacdec->adapter);
}
num = gst_value_array_get_size (headers);
for (i = 0; i < num; ++i) {
const GValue *header_val;
GstBuffer *header_buf;
header_val = gst_value_array_get_value (headers, i);
if (header_val == NULL || !GST_VALUE_HOLDS_BUFFER (header_val))
return FALSE;
header_buf = g_value_dup_boxed (header_val);
GST_INFO_OBJECT (dec, "pushing header buffer of %" G_GSIZE_FORMAT " bytes "
"into adapter", gst_buffer_get_size (header_buf));
gst_adapter_push (flacdec->adapter, header_buf);
}
GST_DEBUG_OBJECT (dec, "Processing headers and metadata");
if (!FLAC__stream_decoder_process_until_end_of_metadata (flacdec->decoder)) {
GST_WARNING_OBJECT (dec, "process_until_end_of_metadata failed");
}
GST_INFO_OBJECT (dec, "headers and metadata are now processed");
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;
}
/* FIXME: for our purposes it's probably enough to just check for the sync
* marker - we just want to know if it's a header frame or not */
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;
val = data[4];
/* This is slightly faster than a loop */
if (!(val & 0x80)) {
val = 0;
} else if ((val & 0xc0) && !(val & 0x20)) {
val = 1;
} else if ((val & 0xe0) && !(val & 0x10)) {
val = 2;
} else if ((val & 0xf0) && !(val & 0x08)) {
val = 3;
} else if ((val & 0xf8) && !(val & 0x04)) {
val = 4;
} else if ((val & 0xfc) && !(val & 0x02)) {
val = 5;
} else if ((val & 0xfe) && !(val & 0x01)) {
val = 6;
} else {
GST_LOG_OBJECT (flacdec, "failed to read sample/frame");
return FALSE;
}
val++;
headerlen = 4 + val + (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 (!last_sample_num)
return TRUE;
/* 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;
}
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->info.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->info.rate));
}
return TRUE;
}
static gboolean
gst_flac_dec_handle_decoder_error (GstFlacDec * dec, gboolean msg)
{
gboolean ret;
dec->error_count++;
if (dec->error_count > 10) {
if (msg)
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), (NULL));
dec->last_flow = GST_FLOW_ERROR;
ret = TRUE;
} else {
GST_DEBUG_OBJECT (dec, "ignoring error for now at count %d",
dec->error_count);
ret = FALSE;
}
return ret;
}
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, width, gdepth;
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->depth = depth = metadata->data.stream_info.bits_per_sample;
if (depth < 9) {
gdepth = width = 8;
} else if (depth < 17) {
gdepth = width = 16;
} else if (depth < 25) {
gdepth = 24;
width = 32;
} else {
gdepth = width = 32;
}
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
metadata->data.stream_info.sample_rate,
metadata->data.stream_info.channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
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->info.rate, flacdec->info.channels);
GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth,
flacdec->info.finfo->width);
GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
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;
}
if (gst_flac_dec_handle_decoder_error (dec, FALSE))
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("%s (%d)", error, status));
}
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=%" G_GSIZE_FORMAT ", 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, gdepth;
guint sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstMapInfo map;
gboolean caps_changed;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", 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;
}
switch (depth) {
case 8:
gdepth = width = 8;
break;
case 12:
case 16:
gdepth = width = 16;
break;
case 20:
case 24:
gdepth = 24;
width = 32;
break;
case 32:
gdepth = width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->info.rate != 0) {
sample_rate = flacdec->info.rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
if (caps_changed
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
channels);
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
sample_rate, channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
flacdec->depth = depth;
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
&flacdec->info);
}
outbuf =
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
}
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &map);
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
if (flacdec->error_count)
flacdec->error_count--;
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the handle_frame 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_flush (GstAudioDecoder * audio_dec, gboolean hard)
{
GstFlacDec *dec = GST_FLAC_DEC (audio_dec);
if (!hard) {
guint available = gst_adapter_available (dec->adapter);
if (available > 0) {
GST_INFO_OBJECT (dec, "draining, %u bytes left in adapter", available);
FLAC__stream_decoder_process_until_end_of_stream (dec->decoder);
}
}
FLAC__stream_decoder_flush (dec->decoder);
gst_adapter_clear (dec->adapter);
}
static GstFlowReturn
gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf)
{
GstFlacDec *dec;
dec = GST_FLAC_DEC (audio_dec);
/* drain remaining data? */
if (G_UNLIKELY (buf == NULL)) {
gst_flac_dec_flush (audio_dec, FALSE);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (dec, "frame: ts %" GST_TIME_FORMAT ", flags 0x%04x, "
"%" G_GSIZE_FORMAT " bytes", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_BUFFER_FLAGS (buf), gst_buffer_get_size (buf));
/* drop any in-stream headers, we've processed those in set_format already */
if (G_UNLIKELY (!dec->got_headers)) {
gboolean got_audio_frame;
GstMapInfo map;
/* check if this is a flac audio frame (rather than a header or junk) */
gst_buffer_map (buf, &map, GST_MAP_READ);
got_audio_frame =
gst_flac_dec_scan_got_frame (dec, map.data, map.size, NULL);
gst_buffer_unmap (buf, &map);
if (!got_audio_frame) {
GST_INFO_OBJECT (dec, "dropping in-stream header, %" G_GSIZE_FORMAT " "
"bytes", map.size);
gst_audio_decoder_finish_frame (audio_dec, NULL, 1);
return GST_FLOW_OK;
}
GST_INFO_OBJECT (dec, "first audio frame, got all in-stream headers now");
dec->got_headers = TRUE;
}
gst_adapter_push (dec->adapter, gst_buffer_ref (buf));
buf = NULL;
dec->last_flow = GST_FLOW_OK;
/* framed - there should always be enough data to decode something */
GST_LOG_OBJECT (dec, "%" G_GSIZE_FORMAT " bytes available",
gst_adapter_available (dec->adapter));
if (!FLAC__stream_decoder_process_single (dec->decoder)) {
GST_INFO_OBJECT (dec, "process_single failed");
}
return dec->last_flow;
}
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