gstreamer/ext/opus/gstopusdec.c

615 lines
18 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 <math.h>
#include <string.h>
#include <gst/tag/tag.h>
#include "gstopusheader.h"
#include "gstopuscommon.h"
#include "gstopusdec.h"
GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
#define GST_CAT_DEFAULT opusdec_debug
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) { 48000, 24000, 16000, 12000, 8000 }, "
"channels = (int) [ 1, 8 ], "
"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")
);
#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
#define DEFAULT_USE_INBAND_FEC FALSE
#define DEFAULT_APPLY_GAIN TRUE
enum
{
PROP_0,
PROP_USE_INBAND_FEC,
PROP_APPLY_GAIN
};
GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,
GstBuffer * buf);
static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);
static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
GstCaps * caps);
static void gst_opus_dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_opus_dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void
gst_opus_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class,
&opus_dec_src_factory);
gst_element_class_add_static_pad_template (element_class,
&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)
{
GObjectClass *gobject_class;
GstAudioDecoderClass *adclass;
gobject_class = (GObjectClass *) klass;
adclass = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_opus_dec_set_property;
gobject_class->get_property = gst_opus_dec_get_property;
adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);
adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);
adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);
adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format);
g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC,
g_param_spec_boolean ("use-inband-fec", "Use in-band FEC",
"Use forward error correction if available", DEFAULT_USE_INBAND_FEC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_APPLY_GAIN,
g_param_spec_boolean ("apply-gain", "Apply gain",
"Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
"opus decoding element");
}
static void
gst_opus_dec_reset (GstOpusDec * dec)
{
dec->packetno = 0;
if (dec->state) {
opus_multistream_decoder_destroy (dec->state);
dec->state = NULL;
}
gst_buffer_replace (&dec->streamheader, NULL);
gst_buffer_replace (&dec->vorbiscomment, NULL);
gst_buffer_replace (&dec->last_buffer, NULL);
dec->primed = FALSE;
dec->pre_skip = 0;
dec->r128_gain = 0;
}
static void
gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
{
dec->sample_rate = 0;
dec->n_channels = 0;
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_opus_dec_reset (dec);
}
static gboolean
gst_opus_dec_start (GstAudioDecoder * dec)
{
GstOpusDec *odec = GST_OPUS_DEC (dec);
gst_opus_dec_reset (odec);
/* we know about concealment */
gst_audio_decoder_set_plc_aware (dec, TRUE);
if (odec->use_inband_fec) {
gst_audio_decoder_set_latency (dec, 2 * GST_MSECOND + GST_MSECOND / 2,
120 * GST_MSECOND);
}
return TRUE;
}
static gboolean
gst_opus_dec_stop (GstAudioDecoder * dec)
{
GstOpusDec *odec = GST_OPUS_DEC (dec);
gst_opus_dec_reset (odec);
return TRUE;
}
static double
gst_opus_dec_get_r128_gain (gint16 r128_gain)
{
return r128_gain / (double) (1 << 8);
}
static double
gst_opus_dec_get_r128_volume (gint16 r128_gain)
{
return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
GstStructure *s;
const GstAudioChannelPosition *pos = NULL;
g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
g_return_val_if_fail (dec->n_channels == 0
|| dec->n_channels == data[9], GST_FLOW_ERROR);
dec->n_channels = data[9];
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 14);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
/* negotiate width with downstream */
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", 48000);
gst_structure_get_int (s, "rate", &dec->sample_rate);
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &dec->n_channels);
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
if (pos) {
GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (dec->n_channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
{
return GST_FLOW_OK;
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
if (dec->state == NULL) {
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
dec->state = opus_multistream_decoder_create (dec->sample_rate,
dec->n_channels, dec->n_streams, dec->n_stereo_streams,
dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %u",
GST_BUFFER_SIZE (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = dec->use_inband_fec && dec->last_buffer ? dec->last_buffer : buffer;
if (buf) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
packet_size = samples * dec->n_channels * 2;
res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, packet_size,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &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);
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal delayed decode */
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
GST_BUFFER_SIZE (outbuf) = n * 2 * dec->n_channels;
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
GST_BUFFER_SIZE (outbuf) -= skip * 2 * dec->n_channels;
GST_BUFFER_DATA (outbuf) += skip * 2 * dec->n_channels;
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
if (GST_BUFFER_SIZE (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
}
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
unsigned int i, nsamples = GST_BUFFER_SIZE (outbuf) / 2;
double volume = dec->r128_gain_volume;
gint16 *samples = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
}
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
}
static gboolean
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstOpusDec *dec = GST_OPUS_DEC (bdec);
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
GST_DEBUG_OBJECT (dec, "set_format: %" 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, *vorbiscomment;
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 = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->streamheader, buf);
}
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_opus_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->vorbiscomment, buf);
}
}
done:
return ret;
}
static gboolean
memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
{
gsize size1, size2;
size1 = GST_BUFFER_SIZE (buf1);
size2 = GST_BUFFER_SIZE (buf2);
if (size1 != size2)
return FALSE;
return !memcmp (GST_BUFFER_DATA (buf1), GST_BUFFER_DATA (buf2), size1);
}
static GstFlowReturn
gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
{
GstFlowReturn res;
GstOpusDec *dec;
/* no fancy draining */
if (G_UNLIKELY (!buf))
return GST_FLOW_OK;
dec = GST_OPUS_DEC (adec);
GST_LOG_OBJECT (dec,
"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 we have the streamheader and vorbiscomment from the caps already
* ignore them here */
if (dec->streamheader && dec->vorbiscomment) {
if (memcmp_buffers (dec->streamheader, buf)) {
GST_DEBUG_OBJECT (dec, "found streamheader");
gst_audio_decoder_finish_frame (adec, NULL, 1);
res = GST_FLOW_OK;
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
GST_DEBUG_OBJECT (dec, "found vorbiscomments");
gst_audio_decoder_finish_frame (adec, NULL, 1);
res = GST_FLOW_OK;
} else {
res = opus_dec_chain_parse_data (dec, buf);
}
} else {
/* Otherwise fall back to packet counting and assume that the
* first two packets might be the headers, checking magic. */
switch (dec->packetno) {
case 0:
if (gst_opus_header_is_header (buf, "OpusHead", 8)) {
GST_DEBUG_OBJECT (dec, "found streamheader");
res = gst_opus_dec_parse_header (dec, buf);
gst_audio_decoder_finish_frame (adec, NULL, 1);
} else {
res = opus_dec_chain_parse_data (dec, buf);
}
break;
case 1:
if (gst_opus_header_is_header (buf, "OpusTags", 8)) {
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
res = gst_opus_dec_parse_comments (dec, buf);
gst_audio_decoder_finish_frame (adec, NULL, 1);
} else {
res = opus_dec_chain_parse_data (dec, buf);
}
break;
default:
{
res = opus_dec_chain_parse_data (dec, buf);
break;
}
}
}
dec->packetno++;
return res;
}
static void
gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstOpusDec *dec = GST_OPUS_DEC (object);
switch (prop_id) {
case PROP_USE_INBAND_FEC:
g_value_set_boolean (value, dec->use_inband_fec);
break;
case PROP_APPLY_GAIN:
g_value_set_boolean (value, dec->apply_gain);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_opus_dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpusDec *dec = GST_OPUS_DEC (object);
switch (prop_id) {
case PROP_USE_INBAND_FEC:
dec->use_inband_fec = g_value_get_boolean (value);
break;
case PROP_APPLY_GAIN:
dec->apply_gain = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}