gstreamer/gst/audioparsers/gstdcaparse.c

617 lines
19 KiB
C

/* GStreamer DCA parser
* Copyright (C) 2010 Tim-Philipp Müller <tim centricular net>
*
* 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-dcaparse
* @short_description: DCA (DTS Coherent Acoustics) parser
* @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse
*
* This is a DCA (DTS Coherent Acoustics) parser.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*/
/* TODO:
* - should accept framed and unframed input (needs decodebin fixes first)
* - seeking in raw .dts files doesn't seem to work, but duration estimate ok
*
* - if frames have 'odd' durations, the frame durations (plus timestamps)
* aren't adjusted up occasionally to make up for rounding error gaps.
* (e.g. if 512 samples per frame @ 48kHz = 10.666666667 ms/frame)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstdcaparse.h"
#include <gst/base/base.h>
#include <gst/pbutils/pbutils.h>
GST_DEBUG_CATEGORY_STATIC (dca_parse_debug);
#define GST_CAT_DEFAULT dca_parse_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-dts,"
" framed = (boolean) true,"
" channels = (int) [ 1, 8 ],"
" rate = (int) [ 8000, 192000 ],"
" depth = (int) { 14, 16 },"
" endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
" block-size = (int) [ 1, MAX], " " frame-size = (int) [ 1, MAX]"));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-dts; " "audio/x-private1-dts"));
static void gst_dca_parse_finalize (GObject * object);
static gboolean gst_dca_parse_start (GstBaseParse * parse);
static gboolean gst_dca_parse_stop (GstBaseParse * parse);
static GstFlowReturn gst_dca_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static GstFlowReturn gst_dca_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
static gboolean gst_dca_parse_set_sink_caps (GstBaseParse * parse,
GstCaps * caps);
#define gst_dca_parse_parent_class parent_class
G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE);
static void
gst_dca_parse_class_init (GstDcaParseClass * klass)
{
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (dca_parse_debug, "dcaparse", 0,
"DCA audio stream parser");
object_class->finalize = gst_dca_parse_finalize;
parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop);
parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_handle_frame);
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_dca_parse_pre_push_frame);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps);
parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_set_sink_caps);
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_static_metadata (element_class,
"DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio",
"DCA parser", "Tim-Philipp Müller <tim centricular net>");
}
static void
gst_dca_parse_reset (GstDcaParse * dcaparse)
{
dcaparse->channels = -1;
dcaparse->rate = -1;
dcaparse->depth = -1;
dcaparse->endianness = -1;
dcaparse->block_size = -1;
dcaparse->frame_size = -1;
dcaparse->last_sync = 0;
dcaparse->sent_codec_tag = FALSE;
}
static void
gst_dca_parse_init (GstDcaParse * dcaparse)
{
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse),
DCA_MIN_FRAMESIZE);
gst_dca_parse_reset (dcaparse);
dcaparse->baseparse_chainfunc =
GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE (dcaparse))->chainfunc;
GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (dcaparse));
GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (dcaparse));
}
static void
gst_dca_parse_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_dca_parse_start (GstBaseParse * parse)
{
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
GST_DEBUG_OBJECT (parse, "starting");
gst_dca_parse_reset (dcaparse);
return TRUE;
}
static gboolean
gst_dca_parse_stop (GstBaseParse * parse)
{
GST_DEBUG_OBJECT (parse, "stopping");
return TRUE;
}
static gboolean
gst_dca_parse_parse_header (GstDcaParse * dcaparse,
const GstByteReader * reader, guint * frame_size,
guint * sample_rate, guint * channels, guint * depth,
gint * endianness, guint * num_blocks, guint * samples_per_block,
gboolean * terminator)
{
static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025,
22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000
};
static const guint8 channels_table[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5,
6, 6, 6, 7, 8, 8
};
GstByteReader r = *reader;
guint16 hdr[8];
guint32 marker;
guint chans, lfe, i;
if (gst_byte_reader_get_remaining (&r) < (4 + sizeof (hdr)))
return FALSE;
marker = gst_byte_reader_peek_uint32_be_unchecked (&r);
/* raw big endian or 14-bit big endian */
if (marker == 0x7FFE8001 || marker == 0x1FFFE800) {
for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
hdr[i] = gst_byte_reader_get_uint16_be_unchecked (&r);
} else
/* raw little endian or 14-bit little endian */
if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) {
for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
hdr[i] = gst_byte_reader_get_uint16_le_unchecked (&r);
} else {
return FALSE;
}
GST_LOG_OBJECT (dcaparse, "dts sync marker 0x%08x at offset %u", marker,
gst_byte_reader_get_pos (reader));
/* 14-bit mode */
if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) {
if ((hdr[2] & 0xFFF0) != 0x07F0)
return FALSE;
/* discard top 2 bits (2 void), shift in 2 */
hdr[0] = (hdr[0] << 2) | ((hdr[1] >> 12) & 0x0003);
/* discard top 4 bits (2 void, 2 shifted into hdr[0]), shift in 4 etc. */
hdr[1] = (hdr[1] << 4) | ((hdr[2] >> 10) & 0x000F);
hdr[2] = (hdr[2] << 6) | ((hdr[3] >> 8) & 0x003F);
hdr[3] = (hdr[3] << 8) | ((hdr[4] >> 6) & 0x00FF);
hdr[4] = (hdr[4] << 10) | ((hdr[5] >> 4) & 0x03FF);
hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF);
hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF);
g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001);
}
GST_LOG_OBJECT (dcaparse, "frame header: %04x%04x%04x%04x",
hdr[2], hdr[3], hdr[4], hdr[5]);
*terminator = (hdr[2] & 0x80) ? FALSE : TRUE;
*samples_per_block = ((hdr[2] >> 10) & 0x1f) + 1;
*num_blocks = ((hdr[2] >> 2) & 0x7F) + 1;
*frame_size = (((hdr[2] & 0x03) << 12) | (hdr[3] >> 4)) + 1;
chans = ((hdr[3] & 0x0F) << 2) | (hdr[4] >> 14);
*sample_rate = sample_rates[(hdr[4] >> 10) & 0x0F];
lfe = (hdr[5] >> 9) & 0x03;
GST_TRACE_OBJECT (dcaparse, "frame size %u, num_blocks %u, rate %u, "
"samples per block %u", *frame_size, *num_blocks, *sample_rate,
*samples_per_block);
if (*num_blocks < 6 || *frame_size < 96 || *sample_rate == 0)
return FALSE;
if (marker == 0x1FFFE800 || marker == 0xFF1F00E8)
*frame_size = (*frame_size * 16) / 14; /* FIXME: round up? */
if (chans < G_N_ELEMENTS (channels_table))
*channels = channels_table[chans] + ((lfe) ? 1 : 0);
else
return FALSE;
if (depth)
*depth = (marker == 0x1FFFE800 || marker == 0xFF1F00E8) ? 14 : 16;
if (endianness)
*endianness = (marker == 0xFE7F0180 || marker == 0xFF1F00E8) ?
G_LITTLE_ENDIAN : G_BIG_ENDIAN;
GST_TRACE_OBJECT (dcaparse, "frame size %u, channels %u, rate %u, "
"num_blocks %u, samples_per_block %u", *frame_size, *channels,
*sample_rate, *num_blocks, *samples_per_block);
return TRUE;
}
static gint
gst_dca_parse_find_sync (GstDcaParse * dcaparse, GstByteReader * reader,
gsize bufsize, guint32 * sync)
{
guint32 best_sync = 0;
guint best_offset = G_MAXUINT;
gint off;
/* FIXME: verify syncs via _parse_header() here already */
/* Raw little endian */
off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xfe7f0180,
0, bufsize);
if (off >= 0 && off < best_offset) {
best_offset = off;
best_sync = 0xfe7f0180;
}
/* Raw big endian */
off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x7ffe8001,
0, bufsize);
if (off >= 0 && off < best_offset) {
best_offset = off;
best_sync = 0x7ffe8001;
}
/* FIXME: check next 2 bytes as well for 14-bit formats (but then don't
* forget to adjust the *skipsize= in _check_valid_frame() */
/* 14-bit little endian */
off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xff1f00e8,
0, bufsize);
if (off >= 0 && off < best_offset) {
best_offset = off;
best_sync = 0xff1f00e8;
}
/* 14-bit big endian */
off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x1fffe800,
0, bufsize);
if (off >= 0 && off < best_offset) {
best_offset = off;
best_sync = 0x1fffe800;
}
if (best_offset == G_MAXUINT)
return -1;
*sync = best_sync;
return best_offset;
}
static GstFlowReturn
gst_dca_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader r;
gboolean parser_in_sync;
gboolean terminator;
guint32 sync = 0;
guint size = 0, rate, chans, num_blocks, samples_per_block, depth;
gint block_size;
gint endianness;
gint off = -1;
GstMapInfo map;
GstFlowReturn ret = GST_FLOW_EOS;
gsize extra_size = 0;
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 16)) {
*skipsize = 1;
goto cleanup;
}
parser_in_sync = !GST_BASE_PARSE_LOST_SYNC (parse);
gst_byte_reader_init (&r, map.data, map.size);
if (G_LIKELY (parser_in_sync && dcaparse->last_sync != 0)) {
off = gst_byte_reader_masked_scan_uint32 (&r, 0xffffffff,
dcaparse->last_sync, 0, map.size);
}
if (G_UNLIKELY (off < 0)) {
off = gst_dca_parse_find_sync (dcaparse, &r, map.size, &sync);
}
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
*skipsize = map.size - 3;
GST_DEBUG_OBJECT (dcaparse, "no sync, skipping %d bytes", *skipsize);
goto cleanup;
}
GST_LOG_OBJECT (parse, "possible sync %08x at buffer offset %d", sync, off);
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto cleanup;
}
/* make sure the values in the frame header look sane */
if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, &depth,
&endianness, &num_blocks, &samples_per_block, &terminator)) {
*skipsize = 4;
goto cleanup;
}
GST_LOG_OBJECT (parse, "got frame, sync %08x, size %u, rate %d, channels %d",
sync, size, rate, chans);
dcaparse->last_sync = sync;
/* FIXME: Don't look for a second syncword, there are streams out there
* that consistently contain garbage between every frame so we never ever
* find a second consecutive syncword.
* See https://bugzilla.gnome.org/show_bug.cgi?id=738237
*/
#if 0
parser_draining = GST_BASE_PARSE_DRAINING (parse);
if (!parser_in_sync && !parser_draining) {
/* check for second frame to be sure */
GST_DEBUG_OBJECT (dcaparse, "resyncing; checking next frame syncword");
if (map.size >= (size + 16)) {
guint s2, r2, c2, n2, s3;
gboolean t;
GST_MEMDUMP ("buf", map.data, size + 16);
gst_byte_reader_init (&r, map.data, map.size);
gst_byte_reader_skip_unchecked (&r, size);
if (!gst_dca_parse_parse_header (dcaparse, &r, &s2, &r2, &c2, NULL, NULL,
&n2, &s3, &t)) {
GST_DEBUG_OBJECT (dcaparse, "didn't find second syncword");
*skipsize = 4;
goto cleanup;
}
/* ok, got sync now, let's assume constant frame size */
gst_base_parse_set_min_frame_size (parse, size);
} else {
/* wait for some more data */
GST_LOG_OBJECT (dcaparse,
"next sync out of reach (%" G_GSIZE_FORMAT " < %u)", map.size,
size + 16);
goto cleanup;
}
}
#endif
/* found frame */
ret = GST_FLOW_OK;
/* metadata handling */
block_size = num_blocks * samples_per_block;
if (G_UNLIKELY (dcaparse->rate != rate || dcaparse->channels != chans
|| dcaparse->depth != depth || dcaparse->endianness != endianness
|| (!terminator && dcaparse->block_size != block_size)
|| (size != dcaparse->frame_size))) {
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-dts",
"framed", G_TYPE_BOOLEAN, TRUE,
"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans,
"endianness", G_TYPE_INT, endianness, "depth", G_TYPE_INT, depth,
"block-size", G_TYPE_INT, block_size, "frame-size", G_TYPE_INT, size,
NULL);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
dcaparse->rate = rate;
dcaparse->channels = chans;
dcaparse->depth = depth;
dcaparse->endianness = endianness;
dcaparse->block_size = block_size;
dcaparse->frame_size = size;
gst_base_parse_set_frame_rate (parse, rate, block_size, 0, 0);
}
cleanup:
/* it is possible that DTS HD substream after DTS core */
if (parse->flags & GST_BASE_PARSE_FLAG_DRAINING || map.size >= size + 9) {
extra_size = 0;
if (map.size >= size + 9) {
const guint8 *next = map.data + size;
/* Check for DTS_SYNCWORD_SUBSTREAM */
if (next[0] == 0x64 && next[1] == 0x58 && next[2] == 0x20
&& next[3] == 0x25) {
/* 7.4.1 Extension Substream Header */
GstBitReader reader;
gst_bit_reader_init (&reader, next + 4, 5);
gst_bit_reader_skip (&reader, 8 + 2); /* skip UserDefinedBits and nExtSSIndex) */
if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) == 0) {
gst_bit_reader_skip (&reader, 8);
extra_size =
gst_bit_reader_get_bits_uint32_unchecked (&reader, 16) + 1;
} else {
gst_bit_reader_skip (&reader, 12);
extra_size =
gst_bit_reader_get_bits_uint32_unchecked (&reader, 20) + 1;
}
}
}
gst_buffer_unmap (buf, &map);
if (ret == GST_FLOW_OK && size + extra_size <= map.size) {
ret = gst_base_parse_finish_frame (parse, frame, size + extra_size);
} else {
ret = GST_FLOW_OK;
}
} else {
gst_buffer_unmap (buf, &map);
}
return ret;
}
/*
* MPEG-PS private1 streams add a 2 bytes "Audio Substream Headers" for each
* buffer (not each frame) with the offset of the next frame's start.
* These 2 bytes can be dropped safely as they do not include any timing
* information, only the offset to the start of the next frame.
* See gstac3parse.c for a more detailed description.
* */
static GstFlowReturn
gst_dca_parse_chain_priv (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstDcaParse *dcaparse = GST_DCA_PARSE (parent);
GstFlowReturn ret;
GstBuffer *newbuf;
gsize size;
size = gst_buffer_get_size (buffer);
if (size >= 2) {
newbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, 2, size - 2);
gst_buffer_unref (buffer);
ret = dcaparse->baseparse_chainfunc (pad, parent, newbuf);
} else {
gst_buffer_unref (buffer);
ret = GST_FLOW_OK;
}
return ret;
}
static void
remove_fields (GstCaps * caps)
{
guint i, n;
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (caps, i);
gst_structure_remove_field (s, "framed");
}
}
static GstCaps *
gst_dca_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps, *templ;
GstCaps *res;
templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
if (filter) {
GstCaps *fcopy = gst_caps_copy (filter);
/* Remove the fields we convert */
remove_fields (fcopy);
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
gst_caps_unref (fcopy);
} else
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
if (peercaps) {
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
remove_fields (peercaps);
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
gst_caps_unref (templ);
} else {
res = templ;
}
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (res);
res = intersection;
}
return res;
}
static gboolean
gst_dca_parse_set_sink_caps (GstBaseParse * parse, GstCaps * caps)
{
GstStructure *s;
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_has_name (s, "audio/x-private1-dts")) {
gst_pad_set_chain_function (parse->sinkpad, gst_dca_parse_chain_priv);
} else {
gst_pad_set_chain_function (parse->sinkpad, dcaparse->baseparse_chainfunc);
}
return TRUE;
}
static GstFlowReturn
gst_dca_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
GstDcaParse *dcaparse = GST_DCA_PARSE (parse);
if (!dcaparse->sent_codec_tag) {
GstTagList *taglist;
GstCaps *caps;
/* codec tag */
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (G_UNLIKELY (caps == NULL)) {
if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
GST_INFO_OBJECT (parse, "Src pad is flushing");
return GST_FLOW_FLUSHING;
} else {
GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
return GST_FLOW_NOT_NEGOTIATED;
}
}
taglist = gst_tag_list_new_empty ();
gst_pb_utils_add_codec_description_to_tag_list (taglist,
GST_TAG_AUDIO_CODEC, caps);
gst_caps_unref (caps);
gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
/* also signals the end of first-frame processing */
dcaparse->sent_codec_tag = TRUE;
}
return GST_FLOW_OK;
}