mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-20 23:36:38 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/dts/gstdtsdec.c
Mark Nauwelaerts 268636c880 dtsdec: prevent compiler warning if orc not present
2010-06-14 15:48:34 +02:00

967 lines
27 KiB
C
Raw Blame History

/* GStreamer DTS decoder plugin based on libdtsdec
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2009 Jan Schmidt <thaytan@noraisin.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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-dtsdec
*
* Digital Theatre System (DTS) audio decoder
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Play a DTS audio track from a dvd.
* |[
* gst-launch filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Decode a standalone file and play it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "_stdint.h"
#include <stdlib.h>
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#ifndef DTS_OLD
#include <dca.h>
#else
#include <dts.h>
typedef struct dts_state_s dca_state_t;
#define DCA_MONO DTS_MONO
#define DCA_CHANNEL DTS_CHANNEL
#define DCA_STEREO DTS_STEREO
#define DCA_STEREO_SUMDIFF DTS_STEREO_SUMDIFF
#define DCA_STEREO_TOTAL DTS_STEREO_TOTAL
#define DCA_3F DTS_3F
#define DCA_2F1R DTS_2F1R
#define DCA_3F1R DTS_3F1R
#define DCA_2F2R DTS_2F2R
#define DCA_3F2R DTS_3F2R
#define DCA_4F2R DTS_4F2R
#define DCA_DOLBY DTS_DOLBY
#define DCA_CHANNEL_MAX DTS_CHANNEL_MAX
#define DCA_CHANNEL_BITS DTS_CHANNEL_BITS
#define DCA_CHANNEL_MASK DTS_CHANNEL_MASK
#define DCA_LFE DTS_LFE
#define DCA_ADJUST_LEVEL DTS_ADJUST_LEVEL
#define dca_init dts_init
#define dca_syncinfo dts_syncinfo
#define dca_frame dts_frame
#define dca_dynrng dts_dynrng
#define dca_blocks_num dts_blocks_num
#define dca_block dts_block
#define dca_samples dts_samples
#define dca_free dts_free
#endif
#include "gstdtsdec.h"
#if HAVE_ORC
#include <orc/orc.h>
#endif
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define SAMPLE_WIDTH 16
#elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE)
#define SAMPLE_WIDTH 64
#else
#define SAMPLE_WIDTH 32
#endif
GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
#define GST_CAT_DEFAULT (dtsdec_debug)
enum
{
ARG_0,
ARG_DRC
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-dts; audio/x-private1-dts")
);
#if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED)
#define DTS_CAPS "audio/x-raw-int, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"signed = (boolean) true, " \
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " \
"depth = (int) 16"
#else
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH)
#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (DTS_CAPS ", "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT);
static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf);
static GstFlowReturn gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf);
static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element,
GstStateChange transition);
static void gst_dtsdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtsdec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_dtsdec_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 (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "DTS audio decoder",
"Codec/Decoder/Audio",
"Decodes DTS audio streams",
"Jan Schmidt <thaytan@noraisin.net>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder");
}
static void
gst_dtsdec_class_init (GstDtsDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_dtsdec_set_property;
gobject_class->get_property = gst_dtsdec_get_property;
gstelement_class->change_state = gst_dtsdec_change_state;
/**
* GstDtsDec::drc
*
* Set to true to apply the recommended DTS dynamic range compression
* to the audio stream. Dynamic range compression makes loud sounds
* softer and soft sounds louder, so you can more easily listen
* to the stream without disturbing other people.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
klass->dts_cpuflags = 0;
#if HAVE_ORC
cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
if (cpuflags & ORC_TARGET_MMX_MMX)
klass->dts_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & ORC_TARGET_MMX_3DNOW)
klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & ORC_TARGET_MMX_MMXEXT)
klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT;
#else
cpuflags = 0;
klass->dts_cpuflags = 0;
#endif
GST_LOG ("CPU flags: dts=%08x, liboil=%08x", klass->dts_cpuflags, cpuflags);
}
static void
gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class)
{
/* create the sink and src pads */
dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps));
gst_pad_set_chain_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
gst_pad_set_event_function (dtsdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event));
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad);
dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad);
dtsdec->request_channels = DCA_CHANNEL;
dtsdec->dynamic_range_compression = FALSE;
gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
}
static gint
gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos)
{
gint chans = 0;
GstAudioChannelPosition *tpos = NULL;
if (pos) {
/* Allocate the maximum, for ease */
tpos = *pos = g_new (GstAudioChannelPosition, 7);
if (!tpos)
return 0;
}
switch (flags & DCA_CHANNEL_MASK) {
case DCA_MONO:
chans = 1;
if (tpos)
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
break;
/* case DCA_CHANNEL: */
case DCA_STEREO:
case DCA_STEREO_SUMDIFF:
case DCA_STEREO_TOTAL:
case DCA_DOLBY:
chans = 2;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DCA_3F:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
break;
case DCA_2F1R:
chans = 3;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DCA_3F1R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
break;
case DCA_2F2R:
chans = 4;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DCA_3F2R:
chans = 5;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
case DCA_4F2R:
chans = 6;
if (tpos) {
tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
tpos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
tpos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
break;
default:
g_warning ("dtsdec: invalid flags 0x%x", flags);
return 0;
}
if (flags & DCA_LFE) {
if (tpos) {
tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE;
}
chans += 1;
}
return chans;
}
static void
clear_queued (GstDtsDec * dec)
{
g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->queued);
dec->queued = NULL;
}
static GstFlowReturn
flush_queued (GstDtsDec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
while (dec->queued) {
GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* iterate ouput queue an push downstream */
ret = gst_pad_push (dec->srcpad, buf);
dec->queued = g_list_delete_link (dec->queued, dec->queued);
}
return ret;
}
static GstFlowReturn
gst_dtsdec_drain (GstDtsDec * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
if (dec->segment.rate < 0.0) {
/* if we have some queued frames for reverse playback, flush
* them now */
ret = flush_queued (dec);
}
return ret;
}
static GstFlowReturn
gst_dtsdec_push (GstDtsDec * dtsdec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
GstFlowReturn result;
flags &= (DCA_CHANNEL_MASK | DCA_LFE);
chans = gst_dtsdec_channels (flags, NULL);
if (!chans) {
GST_ELEMENT_ERROR (GST_ELEMENT (dtsdec), STREAM, DECODE, (NULL),
("Invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
result =
gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
if (result != GST_FLOW_OK)
return result;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / dtsdec->sample_rate;
result = GST_FLOW_OK;
if ((buf = gst_audio_buffer_clip (buf, &dtsdec->segment,
dtsdec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
/* set discont when needed */
if (dtsdec->discont) {
GST_LOG_OBJECT (dtsdec, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
dtsdec->discont = FALSE;
}
if (dtsdec->segment.rate > 0.0) {
GST_DEBUG_OBJECT (dtsdec,
"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
result = gst_pad_push (srcpad, buf);
} else {
/* reverse playback, queue frame till later when we get a discont. */
GST_DEBUG_OBJECT (dtsdec, "queued frame");
dtsdec->queued = g_list_prepend (dtsdec->queued, buf);
}
}
return result;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
GstAudioChannelPosition *pos;
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
gint channels = gst_dtsdec_channels (dts->using_channels, &pos);
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (dts->srcpad, caps))
goto done;
result = TRUE;
done:
if (caps) {
gst_caps_unref (caps);
}
return result;
}
static gboolean
gst_dtsdec_sink_event (GstPad * pad, GstEvent * event)
{
GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad));
gboolean ret = FALSE;
GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat format;
gboolean update;
gint64 start, end, pos;
gdouble rate;
gst_event_parse_new_segment (event, &update, &rate, &format, &start, &end,
&pos);
/* drain queued buffers before activating the segment so that we can clip
* against the old segment first */
gst_dtsdec_drain (dtsdec);
if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
GST_WARNING ("No time in newsegment event %p (format is %s)",
event, gst_format_get_name (format));
gst_event_unref (event);
dtsdec->sent_segment = FALSE;
/* set some dummy values, FIXME: do proper conversion */
dtsdec->time = start = pos = 0;
format = GST_FORMAT_TIME;
end = -1;
} else {
dtsdec->time = start;
dtsdec->sent_segment = TRUE;
ret = gst_pad_push_event (dtsdec->srcpad, event);
}
gst_segment_set_newsegment (&dtsdec->segment, update, rate, format, start,
end, pos);
break;
}
case GST_EVENT_TAG:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_EOS:
gst_dtsdec_drain (dtsdec);
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_FLUSH_START:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
if (dtsdec->cache) {
gst_buffer_unref (dtsdec->cache);
dtsdec->cache = NULL;
}
clear_queued (dtsdec);
gst_segment_init (&dtsdec->segment, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
default:
ret = gst_pad_push_event (dtsdec->srcpad, event);
break;
}
gst_object_unref (dtsdec);
return ret;
}
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_AUDIO_CODEC, "DTS DCA",
GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (dts), dts->srcpad, taglist);
}
static GstFlowReturn
gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gint channels, i, num_blocks;
gboolean need_renegotiation = FALSE;
/* go over stream properties, renegotiate or update streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
if (flags) {
dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
}
if (bit_rate != dts->bit_rate) {
dts->bit_rate = bit_rate;
gst_dtsdec_update_streaminfo (dts);
}
/* If we haven't had an explicit number of channels chosen through properties
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* downstream element such as audioconvert.
* FIXME: Add the property back in for forcing output channels.
*/
if (dts->request_channels != DCA_CHANNEL) {
flags = dts->request_channels;
} else if (dts->flag_update) {
GstCaps *caps;
dts->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (dts->srcpad);
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
gint channels;
const int dts_channels[6] = {
DCA_MONO,
DCA_STEREO,
DCA_STEREO | DCA_LFE,
DCA_2F2R,
DCA_2F2R | DCA_LFE,
DCA_3F2R | DCA_LFE,
};
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
flags ? gst_dtsdec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
flags = dts_channels[channels - 1];
else
flags = dts_channels[5];
gst_caps_unref (copy);
} else if (flags) {
flags = dts->stream_channels;
} else {
flags = DCA_3F2R | DCA_LFE;
}
if (caps)
gst_caps_unref (caps);
} else {
flags = dts->using_channels;
}
/* process */
flags |= DCA_ADJUST_LEVEL;
dts->level = 1;
if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
GST_WARNING_OBJECT (dts, "dts_frame error");
dts->discont = TRUE;
return GST_FLOW_OK;
}
channels = flags & (DCA_CHANNEL_MASK | DCA_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
/* negotiate if required */
if (need_renegotiation) {
GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
dts->sample_rate, dts->stream_channels, dts->using_channels);
if (!gst_dtsdec_renegotiate (dts)) {
GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
if (dts->dynamic_range_compression == FALSE) {
dca_dynrng (dts->state, NULL, NULL);
}
/* handle decoded data, one block is 256 samples */
num_blocks = dca_blocks_num (dts->state);
for (i = 0; i < num_blocks; i++) {
if (dca_block (dts->state)) {
/* Ignore errors, but mark a discont */
GST_WARNING_OBJECT (dts, "dts_block error %d", i);
dts->discont = TRUE;
} else {
GstFlowReturn ret;
/* push on */
ret = gst_dtsdec_push (dts, dts->srcpad, dts->using_channels,
dts->samples, dts->time);
if (ret != GST_FLOW_OK)
return ret;
}
dts->time += GST_SECOND * 256 / dts->sample_rate;
}
return GST_FLOW_OK;
}
static gboolean
gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad));
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (structure && gst_structure_has_name (structure, "audio/x-private1-dts"))
dts->dvdmode = TRUE;
else
dts->dvdmode = FALSE;
gst_object_unref (dts);
return TRUE;
}
static GstFlowReturn
gst_dtsdec_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad));
gint first_access;
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (dts, "received DISCONT");
gst_dtsdec_drain (dts);
/* clear cache on discont and mark a discont in the element */
if (dts->cache) {
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
dts->discont = TRUE;
}
if (dts->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guint8 *data = GST_BUFFER_DATA (buf);
gint offset, len;
GstBuffer *subbuf;
if (size < 2)
goto not_enough_data;
first_access = (data[0] << 8) | data[1];
/* Skip the first_access header */
offset = 2;
if (first_access > 1) {
/* Length of data before first_access */
len = first_access - 1;
if (len <= 0 || offset + len > size)
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
ret = gst_dtsdec_chain_raw (pad, subbuf);
if (ret != GST_FLOW_OK)
goto done;
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
ret = gst_dtsdec_chain_raw (pad, subbuf);
}
} else {
gst_buffer_ref (buf);
ret = gst_dtsdec_chain_raw (pad, buf);
}
done:
gst_buffer_unref (buf);
return ret;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Insufficient data in buffer. Can't determine first_acess"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
bad_first_access_parameter:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstDtsDec *dts;
guint8 *data;
gint size;
gint length = 0, flags, sample_rate, bit_rate, frame_length;
GstFlowReturn result = GST_FLOW_OK;
dts = GST_DTSDEC (GST_PAD_PARENT (pad));
if (!dts->sent_segment) {
GstSegment segment;
/* Create a basic segment. Usually, we'll get a new-segment sent by
* another element that will know more information (a demuxer). If we're
* just looking at a raw AC3 stream, we won't - so we need to send one
* here, but we don't know much info, so just send a minimal TIME
* new-segment event
*/
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_push_event (dts->srcpad, gst_event_new_new_segment (FALSE,
segment.rate, segment.format, segment.start,
segment.duration, segment.start));
dts->sent_segment = TRUE;
}
/* merge with cache, if any. Also make sure timestamps match */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
dts->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (dts,
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
if (dts->cache) {
buf = gst_buffer_join (dts->cache, buf);
dts->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* find and read header */
bit_rate = dts->bit_rate;
sample_rate = dts->sample_rate;
flags = 0;
while (size >= 7) {
length = dca_syncinfo (dts->state, data, &flags,
&sample_rate, &bit_rate, &frame_length);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: frame size %d", length);
if (flags != dts->prev_flags)
dts->flag_update = TRUE;
dts->prev_flags = flags;
result = gst_dtsdec_handle_frame (dts, data, length,
flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
GST_LOG ("Not enough data available (needed %d had %d)", length, size);
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
dts->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
static GstStateChangeReturn
gst_dtsdec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstDtsDec *dts = GST_DTSDEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstDtsDecClass *klass;
klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts));
dts->state = dca_init (klass->dts_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
dts->samples = dca_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
dts->stream_channels = DCA_CHANNEL;
dts->using_channels = DCA_CHANNEL;
dts->level = 1;
dts->bias = 0;
dts->time = 0;
dts->sent_segment = FALSE;
dts->flag_update = TRUE;
gst_segment_init (&dts->segment, GST_FORMAT_UNDEFINED);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
dts->samples = NULL;
if (dts->cache) {
gst_buffer_unref (dts->cache);
dts->cache = NULL;
}
clear_queued (dts);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
dca_free (dts->state);
dts->state = NULL;
break;
default:
break;
}
return ret;
}
static void
gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
dts->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
g_value_set_boolean (value, dts->dynamic_range_compression);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
#if HAVE_ORC
orc_init ();
#endif
if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
GST_TYPE_DTSDEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"dtsdec",
"Decodes DTS audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
Reference in a new issue View git blame Copy permalink