gstreamer/ext/opus/gstopusdec.c
Vincent Penquerc'h ab1d420f88 opusdec: fix decoding
A simple ... opusenc ! opusdec ... pipeline now works.

https://bugzilla.gnome.org/show_bug.cgi?id=660364
2011-10-03 11:21:37 +02:00

886 lines
24 KiB
C

/* GStreamer
* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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.
*/
/*
* Based on the speexdec element.
*/
/**
* SECTION:element-opusdec
* @see_also: opusenc, oggdemux
*
* This element decodes a OPUS stream to raw integer audio.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink
* ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstopusdec.h"
#include <string.h>
#include <gst/tag/tag.h>
GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
#define GST_CAT_DEFAULT opusdec_debug
#define DEC_MAX_FRAME_SIZE 2000
static GstStaticPadTemplate opus_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
"channels = (int) [ 1, 2 ], "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
);
static GstStaticPadTemplate opus_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus")
);
GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstElement, GST_TYPE_ELEMENT);
static gboolean opus_dec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn opus_dec_chain (GstPad * pad, GstBuffer * buf);
static gboolean opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn opus_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean opus_dec_src_event (GstPad * pad, GstEvent * event);
static gboolean opus_dec_src_query (GstPad * pad, GstQuery * query);
static gboolean opus_dec_sink_query (GstPad * pad, GstQuery * query);
static const GstQueryType *opus_get_src_query_types (GstPad * pad);
static const GstQueryType *opus_get_sink_query_types (GstPad * pad);
static gboolean opus_dec_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value);
static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec,
GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
static GstFlowReturn opus_dec_chain_parse_header (GstOpusDec * dec,
GstBuffer * buf);
#if 0
static GstFlowReturn opus_dec_chain_parse_comments (GstOpusDec * dec,
GstBuffer * buf);
#endif
static void
gst_opus_dec_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 (&opus_dec_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&opus_dec_sink_factory));
gst_element_class_set_details_simple (element_class, "Opus audio decoder",
"Codec/Decoder/Audio",
"decode opus streams to audio",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_opus_dec_class_init (GstOpusDecClass * klass)
{
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass *) klass;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (opus_dec_change_state);
GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
"opus decoding element");
}
static void
gst_opus_dec_reset (GstOpusDec * dec)
{
gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
dec->granulepos = -1;
dec->packetno = 0;
dec->frame_size = 0;
dec->frame_samples = 960;
dec->frame_duration = 0;
if (dec->state) {
opus_decoder_destroy (dec->state);
dec->state = NULL;
}
#if 0
if (dec->mode) {
opus_mode_destroy (dec->mode);
dec->mode = NULL;
}
#endif
gst_buffer_replace (&dec->streamheader, NULL);
gst_buffer_replace (&dec->vorbiscomment, NULL);
g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->extra_headers);
dec->extra_headers = NULL;
#if 0
memset (&dec->header, 0, sizeof (dec->header));
#endif
}
static void
gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
{
dec->sinkpad =
gst_pad_new_from_static_template (&opus_dec_sink_factory, "sink");
gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (opus_dec_chain));
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (opus_dec_sink_event));
gst_pad_set_query_type_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (opus_get_sink_query_types));
gst_pad_set_query_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (opus_dec_sink_query));
gst_pad_set_setcaps_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (opus_dec_sink_setcaps));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad = gst_pad_new_from_static_template (&opus_dec_src_factory, "src");
gst_pad_use_fixed_caps (dec->srcpad);
gst_pad_set_event_function (dec->srcpad,
GST_DEBUG_FUNCPTR (opus_dec_src_event));
gst_pad_set_query_type_function (dec->srcpad,
GST_DEBUG_FUNCPTR (opus_get_src_query_types));
gst_pad_set_query_function (dec->srcpad,
GST_DEBUG_FUNCPTR (opus_dec_src_query));
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->sample_rate = 48000;
dec->n_channels = 2;
gst_opus_dec_reset (dec);
}
static gboolean
opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
GST_DEBUG_OBJECT (pad, "Setting sink caps to %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = opus_dec_chain_parse_header (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->streamheader, buf);
}
#if 0
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = opus_dec_chain_parse_comments (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->vorbiscomment, buf);
}
#endif
g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->extra_headers);
dec->extra_headers = NULL;
if (gst_value_array_get_size (streamheader) > 2) {
gint i, n;
n = gst_value_array_get_size (streamheader);
for (i = 2; i < n; i++) {
header = gst_value_array_get_value (streamheader, i);
buf = gst_value_get_buffer (header);
dec->extra_headers =
g_list_prepend (dec->extra_headers, gst_buffer_ref (buf));
}
}
}
if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
GST_WARNING_OBJECT (dec, "Frame size not included in caps");
}
if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
}
if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
}
switch (dec->frame_size) {
case 2:
dec->frame_samples = dec->sample_rate / 400;
break;
case 5:
dec->frame_samples = dec->sample_rate / 200;
break;
case 10:
dec->frame_samples = dec->sample_rate / 100;
break;
case 20:
dec->frame_samples = dec->sample_rate / 50;
break;
case 40:
dec->frame_samples = dec->sample_rate / 25;
break;
case 60:
dec->frame_samples = 3 * dec->sample_rate / 50;
break;
default:
GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
break;
}
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
GST_SECOND, dec->sample_rate);
GST_INFO_OBJECT (dec,
"Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
done:
gst_object_unref (dec);
return ret;
}
static gboolean
opus_dec_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = TRUE;
GstOpusDec *dec;
guint64 scale = 1;
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
if (dec->packetno < 1) {
res = FALSE;
goto cleanup;
}
if (src_format == *dest_format) {
*dest_value = src_value;
res = TRUE;
goto cleanup;
}
if (pad == dec->sinkpad &&
(src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) {
res = FALSE;
goto cleanup;
}
switch (src_format) {
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = sizeof (gint16) * dec->n_channels;
case GST_FORMAT_DEFAULT:
*dest_value =
gst_util_uint64_scale_int (scale * src_value,
dec->sample_rate, GST_SECOND);
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * sizeof (gint16) * dec->n_channels;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND,
dec->sample_rate);
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / (sizeof (gint16) * dec->n_channels);
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
dec->sample_rate * sizeof (gint16) * dec->n_channels);
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
cleanup:
gst_object_unref (dec);
return res;
}
static const GstQueryType *
opus_get_sink_query_types (GstPad * pad)
{
static const GstQueryType opus_dec_sink_query_types[] = {
GST_QUERY_CONVERT,
0
};
return opus_dec_sink_query_types;
}
static gboolean
opus_dec_sink_query (GstPad * pad, GstQuery * query)
{
GstOpusDec *dec;
gboolean res;
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
res = opus_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (dec);
return res;
}
static const GstQueryType *
opus_get_src_query_types (GstPad * pad)
{
static const GstQueryType opus_dec_src_query_types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
0
};
return opus_dec_src_query_types;
}
static gboolean
opus_dec_src_query (GstPad * pad, GstQuery * query)
{
GstOpusDec *dec;
gboolean res = FALSE;
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:{
GstSegment segment;
GstFormat format;
gint64 cur;
gst_query_parse_position (query, &format, NULL);
GST_PAD_STREAM_LOCK (dec->sinkpad);
segment = dec->segment;
GST_PAD_STREAM_UNLOCK (dec->sinkpad);
if (segment.format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (dec, "segment not initialised yet");
break;
}
if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
segment.last_stop, &format, &cur))) {
gst_query_set_position (query, format, cur);
}
break;
}
case GST_QUERY_DURATION:{
GstFormat format = GST_FORMAT_TIME;
gint64 dur;
/* get duration from demuxer */
if (!gst_pad_query_peer_duration (dec->sinkpad, &format, &dur))
break;
gst_query_parse_duration (query, &format, NULL);
/* and convert it into the requested format */
if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
dur, &format, &dur))) {
gst_query_set_duration (query, format, dur);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (dec);
return res;
}
static gboolean
opus_dec_src_event (GstPad * pad, GstEvent * event)
{
gboolean res = FALSE;
GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
GstFormat format, tformat;
gdouble rate;
GstEvent *real_seek;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 tcur, tstop;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
&stop_type, &stop);
/* we have to ask our peer to seek to time here as we know
* nothing about how to generate a granulepos from the src
* formats or anything.
*
* First bring the requested format to time
*/
tformat = GST_FORMAT_TIME;
if (!(res = opus_dec_convert (pad, format, cur, &tformat, &tcur)))
break;
if (!(res = opus_dec_convert (pad, format, stop, &tformat, &tstop)))
break;
/* then seek with time on the peer */
real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
flags, cur_type, tcur, stop_type, tstop);
GST_LOG_OBJECT (dec, "seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (tcur));
res = gst_pad_push_event (dec->sinkpad, real_seek);
gst_event_unref (event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (dec);
return res;
}
static gboolean
opus_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstOpusDec *dec;
gboolean ret = FALSE;
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
if (rate <= 0.0)
goto newseg_wrong_rate;
if (update) {
/* time progressed without data, see if we can fill the gap with
* some concealment data */
if (dec->segment.last_stop < start) {
GstClockTime duration;
duration = start - dec->segment.last_stop;
opus_dec_chain_parse_data (dec, NULL, dec->segment.last_stop,
duration);
}
}
/* now configure the values */
gst_segment_set_newsegment_full (&dec->segment, update,
rate, arate, GST_FORMAT_TIME, start, stop, time);
dec->granulepos = -1;
GST_DEBUG_OBJECT (dec, "segment now: cur = %" GST_TIME_FORMAT " [%"
GST_TIME_FORMAT " - %" GST_TIME_FORMAT "]",
GST_TIME_ARGS (dec->segment.last_stop),
GST_TIME_ARGS (dec->segment.start),
GST_TIME_ARGS (dec->segment.stop));
ret = gst_pad_push_event (dec->srcpad, event);
break;
}
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (dec);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
gst_object_unref (dec);
return FALSE;
}
newseg_wrong_rate:
{
GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
gst_object_unref (dec);
return FALSE;
}
}
static GstFlowReturn
opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
GstCaps *caps;
int err;
#if 0
dec->samples_per_frame = opus_packet_get_samples_per_frame (
(const unsigned char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
#endif
#if 0
if (memcmp (dec->header.codec_id, "OPUS ", 8) != 0)
goto invalid_header;
#endif
dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
if (!dec->state || err != OPUS_OK)
goto init_failed;
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
GST_SECOND, dec->sample_rate);
/* set caps */
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->n_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
dec->sample_rate, dec->n_channels, dec->frame_size);
if (!gst_pad_set_caps (dec->srcpad, caps))
goto nego_failed;
gst_caps_unref (caps);
return GST_FLOW_OK;
/* ERRORS */
#if 0
invalid_header:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Invalid header"));
return GST_FLOW_ERROR;
}
mode_init_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Mode initialization failed: %d", error));
return GST_FLOW_ERROR;
}
#endif
init_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't initialize decoder"));
return GST_FLOW_ERROR;
}
nego_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't negotiate format"));
gst_caps_unref (caps);
return GST_FLOW_NOT_NEGOTIATED;
}
}
#if 0
static GstFlowReturn
opus_dec_chain_parse_comments (GstOpusDec * dec, GstBuffer * buf)
{
GstTagList *list;
gchar *encoder = NULL;
list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
if (!list) {
GST_WARNING_OBJECT (dec, "couldn't decode comments");
list = gst_tag_list_new ();
}
if (encoder) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
}
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Opus", NULL);
if (dec->header.bytes_per_packet > 0) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) dec->header.bytes_per_packet * 8, NULL);
}
GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
g_free (encoder);
g_free (ver);
return GST_FLOW_OK;
}
#endif
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
GstClockTime timestamp, GstClockTime duration)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
if (timestamp != -1) {
dec->segment.last_stop = timestamp;
dec->granulepos = -1;
}
if (dec->state == NULL) {
GstCaps *caps;
dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
/* set caps */
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->n_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
dec->sample_rate, dec->n_channels, dec->frame_size);
if (!gst_pad_set_caps (dec->srcpad, caps))
GST_ERROR ("nego failure");
gst_caps_unref (caps);
}
if (buf) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
/* copy timestamp */
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
data = NULL;
size = 0;
}
GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data,
48000));
GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data));
res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2,
GST_PAD_CAPS (dec->srcpad), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_LOG_OBJECT (dec, "decoding frame");
n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
return GST_FLOW_ERROR;
}
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
timestamp = gst_util_uint64_scale_int (dec->granulepos - dec->frame_size,
GST_SECOND, dec->sample_rate);
}
GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
if (dec->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->discont = 0;
}
dec->segment.last_stop += dec->frame_duration;
GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (dec->frame_duration));
res = gst_pad_push (dec->srcpad, outbuf);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
}
static GstFlowReturn
opus_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn res;
GstOpusDec *dec;
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (pad,
"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
if (GST_BUFFER_IS_DISCONT (buf)) {
dec->discont = TRUE;
}
res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
GST_BUFFER_DURATION (buf));
//done:
dec->packetno++;
gst_buffer_unref (buf);
gst_object_unref (dec);
return res;
}
static GstStateChangeReturn
opus_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstOpusDec *dec = GST_OPUS_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
default:
break;
}
ret = parent_class->change_state (element, transition);
if (ret != GST_STATE_CHANGE_SUCCESS)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_opus_dec_reset (dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}