mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-10 19:31:12 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/a52dec/gsta52dec.c

1000 lines
28 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.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-a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioresample ! audioconvert ! alsasink
* ]| Play audio track from a dvd.
* |[
* gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
* ]| Decode a stand alone file and play it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include "_stdint.h"
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <a52dec/a52.h>
#include <a52dec/mm_accel.h>
#include "gsta52dec.h"
#if HAVE_ORC
#include <orc/orc.h>
#endif
#ifdef LIBA52_DOUBLE
#define SAMPLE_WIDTH 64
#else
#define SAMPLE_WIDTH 32
#endif
GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
#define GST_CAT_DEFAULT (a52dec_debug)
/* A52Dec args */
enum
{
ARG_0,
ARG_DRC,
ARG_MODE,
ARG_LFE,
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-ac3; audio/ac3; audio/x-private1-ac3")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
GstStateChange transition);
static void gst_a52dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_a52dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
#define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type())
static GType
gst_a52dec_mode_get_type (void)
{
static GType a52dec_mode_type = 0;
static const GEnumValue a52dec_modes[] = {
{A52_MONO, "Mono", "mono"},
{A52_STEREO, "Stereo", "stereo"},
{A52_3F, "3 Front", "3f"},
{A52_2F1R, "2 Front, 1 Rear", "2f1r"},
{A52_3F1R, "3 Front, 1 Rear", "3f1r"},
{A52_2F2R, "2 Front, 2 Rear", "2f2r"},
{A52_3F2R, "3 Front, 2 Rear", "3f2r"},
{A52_DOLBY, "Dolby", "dolby"},
{0, NULL, NULL},
};
if (!a52dec_mode_type) {
a52dec_mode_type = g_enum_register_static ("GstA52DecMode", a52dec_modes);
}
return a52dec_mode_type;
}
static void
gst_a52dec_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,
"ATSC A/52 audio decoder", "Codec/Decoder/Audio",
"Decodes ATSC A/52 encoded audio streams",
"David I. Lehn <dlehn@users.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
"AC3/A52 software decoder");
}
static void
gst_a52dec_class_init (GstA52DecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_a52dec_set_property;
gobject_class->get_property = gst_a52dec_get_property;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
/**
* GstA52Dec::drc
*
* Set to true to apply the recommended Dolby Digital 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 | G_PARAM_STATIC_STRINGS));
/**
* GstA52Dec::mode
*
* Force a particular output channel configuration from the decoder. By default,
* the channel downmix (if any) is chosen automatically based on the downstream
* capabilities of the pipeline.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
GST_TYPE_A52DEC_MODE, A52_3F2R,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstA52Dec::lfe
*
* Whether to output the LFE (Low Frequency Emitter) channel of the audio stream.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE,
g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/* If no CPU instruction based acceleration is available, end up using the
* generic software djbfft based one when available in the used liba52 */
#ifdef MM_ACCEL_DJBFFT
klass->a52_cpuflags = MM_ACCEL_DJBFFT;
#else
klass->a52_cpuflags = 0;
#endif
#if HAVE_ORC
cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
if (cpuflags & ORC_TARGET_MMX_MMX)
klass->a52_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & ORC_TARGET_MMX_3DNOW)
klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & ORC_TARGET_MMX_MMXEXT)
klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
#else
cpuflags = 0;
#endif
GST_LOG ("CPU flags: a52=%08x, liboil=%08x", klass->a52_cpuflags, cpuflags);
}
static void
gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
{
/* create the sink and src pads */
a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
gst_pad_set_chain_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_chain));
gst_pad_set_event_function (a52dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
}
static gint
gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
{
gint chans = 0;
GstAudioChannelPosition *pos = NULL;
/* allocated just for safety. Number makes no sense */
if (_pos) {
pos = g_new (GstAudioChannelPosition, 6);
*_pos = pos;
}
if (flags & A52_LFE) {
chans += 1;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE;
}
}
flags &= A52_CHANNEL_MASK;
switch (flags) {
case A52_3F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 5;
break;
case A52_2F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 4;
break;
case A52_3F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 4;
break;
case A52_2F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 3;
break;
case A52_3F:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 3;
break;
case A52_CHANNEL: /* Dual mono. Should really be handled as 2 src pads */
case A52_STEREO:
case A52_DOLBY:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 2;
break;
case A52_MONO:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
}
chans += 1;
break;
default:
/* error, caller should post error message */
g_free (pos);
return 0;
}
return chans;
}
static void
clear_queued (GstA52Dec * dec)
{
g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->queued);
dec->queued = NULL;
}
static GstFlowReturn
flush_queued (GstA52Dec * 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_a52dec_drain (GstA52Dec * 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_a52dec_push (GstA52Dec * a52dec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
GstFlowReturn result;
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), 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 / a52dec->sample_rate;
result = GST_FLOW_OK;
if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
/* set discont when needed */
if (a52dec->discont) {
GST_LOG_OBJECT (a52dec, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
a52dec->discont = FALSE;
}
if (a52dec->segment.rate > 0.0) {
GST_DEBUG_OBJECT (a52dec,
"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 (a52dec, "queued frame");
a52dec->queued = g_list_prepend (a52dec->queued, buf);
}
}
return result;
}
static gboolean
gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
{
GstAudioChannelPosition *pos;
gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
GstCaps *caps = NULL;
gboolean result = FALSE;
if (!channels)
goto done;
GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
channels, a52dec->sample_rate);
caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, SAMPLE_WIDTH,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, a52dec->sample_rate, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
if (!gst_pad_set_caps (pad, caps))
goto done;
result = TRUE;
done:
if (caps)
gst_caps_unref (caps);
return result;
}
static gboolean
gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
{
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
gboolean ret = FALSE;
GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat fmt;
gboolean update;
gint64 start, end, pos;
gdouble rate, arate;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
&start, &end, &pos);
/* drain queued buffers before activating the segment so that we can clip
* against the old segment first */
gst_a52dec_drain (a52dec);
if (fmt != 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 (fmt));
gst_event_unref (event);
a52dec->sent_segment = FALSE;
/* set some dummy values, FIXME: do proper conversion */
a52dec->time = start = pos = 0;
fmt = GST_FORMAT_TIME;
end = -1;
} else {
a52dec->time = start;
a52dec->sent_segment = TRUE;
GST_DEBUG_OBJECT (a52dec,
"Pushing newseg rate %g, applied rate %g, format %d, start %"
G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
G_GINT64_FORMAT, rate, arate, fmt, start, end, pos);
ret = gst_pad_push_event (a52dec->srcpad, event);
}
gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
end, pos);
break;
}
case GST_EVENT_TAG:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_EOS:
gst_a52dec_drain (a52dec);
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_FLUSH_START:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
default:
ret = gst_pad_push_event (a52dec->srcpad, event);
break;
}
gst_object_unref (a52dec);
return ret;
}
static void
gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
GST_PAD (a52dec->srcpad), taglist);
}
static GstFlowReturn
gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gint channels, i;
gboolean need_reneg = FALSE;
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
if (a52dec->sample_rate != sample_rate) {
need_reneg = TRUE;
a52dec->sample_rate = sample_rate;
}
if (flags) {
a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
}
if (bit_rate != a52dec->bit_rate) {
a52dec->bit_rate = bit_rate;
gst_a52dec_update_streaminfo (a52dec);
}
/* 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.
*/
if (a52dec->request_channels != A52_CHANNEL) {
flags = a52dec->request_channels;
} else if (a52dec->flag_update) {
GstCaps *caps;
a52dec->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (a52dec->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 a52_channels[6] = {
A52_MONO,
A52_STEREO,
A52_STEREO | A52_LFE,
A52_2F2R,
A52_2F2R | A52_LFE,
A52_3F2R | A52_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_a52dec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
flags = a52_channels[channels - 1];
else
flags = a52_channels[5];
gst_caps_unref (copy);
} else if (flags)
flags = a52dec->stream_channels;
else
flags = A52_3F2R | A52_LFE;
if (caps)
gst_caps_unref (caps);
} else {
flags = a52dec->using_channels;
}
/* process */
flags |= A52_ADJUST_LEVEL;
a52dec->level = 1;
if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
GST_WARNING ("a52_frame error");
a52dec->discont = TRUE;
return GST_FLOW_OK;
}
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
if (a52dec->using_channels != channels) {
need_reneg = TRUE;
a52dec->using_channels = channels;
}
/* negotiate if required */
if (need_reneg) {
GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
/* each frame consists of 6 blocks */
for (i = 0; i < 6; i++) {
if (a52_block (a52dec->state)) {
/* ignore errors but mark a discont */
GST_WARNING ("a52_block error %d", i);
a52dec->discont = TRUE;
} else {
GstFlowReturn ret;
/* push on */
ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
a52dec->samples, a52dec->time);
if (ret != GST_FLOW_OK)
return ret;
}
a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
}
return GST_FLOW_OK;
}
static gboolean
gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
a52dec->dvdmode = TRUE;
else
a52dec->dvdmode = FALSE;
gst_object_unref (a52dec);
return TRUE;
}
static GstFlowReturn
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
GstFlowReturn ret;
gint first_access;
if (GST_BUFFER_IS_DISCONT (buf)) {
GST_LOG_OBJECT (a52dec, "received DISCONT");
gst_a52dec_drain (a52dec);
/* clear cache on discont and mark a discont in the element */
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
a52dec->discont = TRUE;
}
if (a52dec->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guchar *data = GST_BUFFER_DATA (buf);
gint offset;
gint 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_a52dec_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_a52dec_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_a52dec_chain_raw (pad, subbuf);
}
} else {
gst_buffer_ref (buf);
ret = gst_a52dec_chain_raw (pad, buf);
}
done:
gst_buffer_unref (buf);
return ret;
/* ERRORS */
not_enough_data:
{
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), 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 (a52dec), STREAM, DECODE, (NULL),
("Bad first_access parameter (%d) in buffer", first_access));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec;
guint8 *data;
guint size;
gint length = 0, flags, sample_rate, bit_rate;
GstFlowReturn result = GST_FLOW_OK;
a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
if (!a52dec->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 (a52dec->srcpad, gst_event_new_new_segment (FALSE,
segment.rate, segment.format, segment.start,
segment.duration, segment.start));
a52dec->sent_segment = TRUE;
}
/* merge with cache, if any. Also make sure timestamps match */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (a52dec,
"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 (a52dec->cache) {
buf = gst_buffer_join (a52dec->cache, buf);
a52dec->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* no sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: %d", length);
if (flags != a52dec->prev_flags)
a52dec->flag_update = TRUE;
a52dec->prev_flags = flags;
result = gst_a52dec_handle_frame (a52dec, data,
length, flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
/* not enough data */
GST_LOG ("Not enough data available");
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
a52dec->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
static GstStateChangeReturn
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstA52Dec *a52dec = GST_A52DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstA52DecClass *klass;
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
a52dec->state = a52_init (klass->a52_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->time = 0;
a52dec->sent_segment = FALSE;
a52dec->flag_update = TRUE;
gst_segment_init (&a52dec->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:
a52dec->samples = NULL;
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
a52_free (a52dec->state);
a52dec->state = NULL;
break;
default:
break;
}
return ret;
}
static void
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstA52Dec *src = GST_A52DEC (object);
switch (prop_id) {
case ARG_DRC:
GST_OBJECT_LOCK (src);
src->dynamic_range_compression = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (src);
break;
case ARG_MODE:
GST_OBJECT_LOCK (src);
src->request_channels &= ~A52_CHANNEL_MASK;
src->request_channels |= g_value_get_enum (value);
GST_OBJECT_UNLOCK (src);
break;
case ARG_LFE:
GST_OBJECT_LOCK (src);
src->request_channels &= ~A52_LFE;
src->request_channels |= g_value_get_boolean (value) ? A52_LFE : 0;
GST_OBJECT_UNLOCK (src);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstA52Dec *src = GST_A52DEC (object);
switch (prop_id) {
case ARG_DRC:
GST_OBJECT_LOCK (src);
g_value_set_boolean (value, src->dynamic_range_compression);
GST_OBJECT_UNLOCK (src);
break;
case ARG_MODE:
GST_OBJECT_LOCK (src);
g_value_set_enum (value, src->request_channels & A52_CHANNEL_MASK);
GST_OBJECT_UNLOCK (src);
break;
case ARG_LFE:
GST_OBJECT_LOCK (src);
g_value_set_boolean (value, src->request_channels & A52_LFE);
GST_OBJECT_UNLOCK (src);
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
/* ensure GstAudioChannelPosition type is registered */
if (!gst_audio_channel_position_get_type ())
return FALSE;
if (!gst_element_register (plugin, "a52dec", GST_RANK_SECONDARY,
GST_TYPE_A52DEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"a52dec",
"Decodes ATSC A/52 encoded audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
Reference in a new issue View git blame Copy permalink