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opus: multichannel support

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This commit is contained in:
Vincent Penquerc'h 2011-11-24 13:29:56 +00:00
parent 670c365400
commit d38f4b8a09
9 changed files with 369 additions and 91 deletions
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4
ext/opus/Makefile.am
View file

@ -1,6 +1,6 @@
plugin_LTLIBRARIES = libgstopus.la
libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c
libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
libgstopus_la_CFLAGS = \
-DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) \
@ -15,4 +15,4 @@ libgstopus_la_LIBADD = \
libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
libgstopus_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h
noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h

72
ext/opus/gstopuscommon.c Normal file
View file

@ -0,0 +1,72 @@
/* GStreamer
* Copyright (C) 2009 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.
*/
#include "gstopuscommon.h"
/* http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 */
/* copy of the same structure in the vorbis plugin */
const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
{ /* Mono */
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{ /* Stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* Stereo + Centre */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* Quadraphonic */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
{ /* Stereo + Centre + rear stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
{ /* Full 5.1 Surround */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE,
},
{ /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE},
{ /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE},
};

33
ext/opus/gstopuscommon.h Normal file
View file

@ -0,0 +1,33 @@
/* GStreamer Opus Encoder
* Copyright (C) 2009 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.
*/
#ifndef __GST_OPUS_COMMON_H__
#define __GST_OPUS_COMMON_H__
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
G_BEGIN_DECLS
extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
G_END_DECLS
#endif /* __GST_OPUS_COMMON_H__ */

141
ext/opus/gstopusdec.c
View file

@ -45,6 +45,7 @@
#include <string.h>
#include <gst/tag/tag.h>
#include "gstopusheader.h"
#include "gstopuscommon.h"
#include "gstopusdec.h"
GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
@ -56,7 +57,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
"channels = (int) [ 1, 2 ], "
"channels = (int) [ 1, 8 ], "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
);
@ -217,26 +218,91 @@ gst_opus_dec_get_r128_volume (gint16 r128_gain)
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
g_return_val_if_fail (gst_opus_header_is_header (buf, "OpusHead", 8),
GST_FLOW_ERROR);
g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 19, GST_FLOW_ERROR);
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
GstStructure *s;
const GstAudioChannelPosition *pos = NULL;
dec->pre_skip = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 10);
dec->r128_gain = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 14);
g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
g_return_val_if_fail (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 = GST_BUFFER_DATA (buf)[18];
if (dec->channel_mapping_family != 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
("Decoding error: unsupported channel nmapping family %d",
dec->channel_mapping_family), (NULL));
return GST_FLOW_ERROR;
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);
}
}
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
/* 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_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;
}
@ -248,48 +314,6 @@ gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
return GST_FLOW_OK;
}
static void
gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
{
GstPad *srcpad, *peer;
GstStructure *s;
GstCaps *caps;
const GstCaps *template_caps;
const GstCaps *peer_caps;
srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
peer = gst_pad_get_peer (srcpad);
if (peer) {
template_caps = gst_pad_get_pad_template_caps (srcpad);
peer_caps = gst_pad_get_caps (peer);
GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
caps = gst_caps_intersect (template_caps, peer_caps);
gst_pad_fixate_caps (peer, caps);
GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
dec->n_channels = 2;
GST_WARNING_OBJECT (dec, "Failed to get channels, using default %d",
dec->n_channels);
} else {
GST_DEBUG_OBJECT (dec, "Got channels %d", dec->n_channels);
}
if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
dec->sample_rate = 48000;
GST_WARNING_OBJECT (dec, "Failed to get rate, using default %d",
dec->sample_rate);
} else {
GST_DEBUG_OBJECT (dec, "Got sample rate %d", dec->sample_rate);
}
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
} else {
GST_WARNING_OBJECT (dec, "Failed to get src pad peer");
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
@ -304,12 +328,11 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
GstBuffer *buf;
if (dec->state == NULL) {
gst_opus_dec_setup_from_peer_caps (dec);
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, 1, 1, dec->channel_mapping, &err);
dec->n_channels, dec->n_streams, dec->n_stereo_streams,
dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}

3
ext/opus/gstopusdec.h
View file

@ -55,6 +55,9 @@ struct _GstOpusDec {
int n_channels;
guint32 pre_skip;
gint16 r128_gain;
guint8 n_streams;
guint8 n_stereo_streams;
guint8 channel_mapping_family;
guint8 channel_mapping[256];

100
ext/opus/gstopusenc.c
View file

@ -49,6 +49,7 @@
#include <gst/gsttagsetter.h>
#include <gst/audio/audio.h>
#include "gstopusheader.h"
#include "gstopuscommon.h"
#include "gstopusenc.h"
GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
@ -116,8 +117,8 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
"channels = (int) [ 1, 2 ], "
"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
"channels = (int) [ 1, 8 ], "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
);
@ -419,6 +420,82 @@ gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
return frame_samples;
}
static void
gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
{
#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
int n;
GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
enc->n_channels);
/* Start by setting up a default trivial mapping */
for (n = 0; n < 255; ++n)
enc->channel_mapping[n] = n;
/* For one channel, use the basic RTP mapping */
if (enc->n_channels == 1) {
GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
enc->channel_mapping_family = 0;
enc->channel_mapping[0] = 0;
return;
}
/* For two channels, use the basic RTP mapping if the channels are
mapped as left/right. */
if (enc->n_channels == 2) {
if (MAPS (0, FRONT_LEFT) && MAPS (1, FRONT_RIGHT)) {
GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
enc->channel_mapping_family = 0;
/* The channel mapping is implicit for family 0, that's why we do not
attempt to create one for right/left - this will be mapped to the
Vorbis mapping below. */
} else {
GST_DEBUG_OBJECT (enc, "Stereo, but not canonical mapping, continuing");
}
}
/* For channels between 1 and 8, we use the Vorbis mapping if we can
find a permutation that matches it. Mono will have been taken care
of earlier, but this code also handles it. */
if (enc->n_channels >= 1 && enc->n_channels <= 8) {
GST_DEBUG_OBJECT (enc,
"In range for the Vorbis mapping, checking channel positions");
for (n = 0; n < enc->n_channels; ++n) {
GstAudioChannelPosition pos = GST_AUDIO_INFO_POSITION (info, n);
int c;
GST_DEBUG_OBJECT (enc, "Channel %d has position %d", n, pos);
for (c = 0; c < enc->n_channels; ++c) {
if (gst_opus_channel_positions[enc->n_channels - 1][c] == pos) {
GST_DEBUG_OBJECT (enc, "Found in Vorbis mapping as channel %d", c);
break;
}
}
if (c == enc->n_channels) {
/* We did not find that position, so use undefined */
GST_WARNING_OBJECT (enc,
"Position %d not found in Vorbis mapping, using unknown mapping",
pos);
enc->channel_mapping_family = 255;
return;
}
GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d", c, n);
enc->channel_mapping[c] = n;
}
GST_INFO_OBJECT (enc, "Permutation found, using Vorbis mapping");
enc->channel_mapping_family = 1;
return;
}
/* For other cases, we use undefined, with the default trivial mapping */
GST_WARNING_OBJECT (enc, "Unknown mapping");
enc->channel_mapping_family = 255;
#undef MAPS
}
static gboolean
gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
@ -430,6 +507,7 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
enc->sample_rate = GST_AUDIO_INFO_RATE (info);
gst_opus_enc_setup_channel_mapping (enc, info);
GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
enc->sample_rate);
@ -455,17 +533,12 @@ static gboolean
gst_opus_enc_setup (GstOpusEnc * enc)
{
int error = OPUS_OK;
unsigned char mapping[256];
int n;
GST_DEBUG_OBJECT (enc, "setup");
for (n = 0; n < enc->n_channels; ++n)
mapping[n] = n;
enc->state =
opus_multistream_encoder_create (enc->sample_rate, enc->n_channels,
(enc->n_channels + 1) / 2, enc->n_channels / 2, mapping,
(enc->n_channels + 1) / 2, enc->n_channels / 2, enc->channel_mapping,
enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
&error);
if (!enc->state || error != OPUS_OK)
@ -557,18 +630,19 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
GstBuffer *outbuf;
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, enc->max_payload_size,
GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if (GST_FLOW_OK != ret)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples);
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, GST_BUFFER_DATA (outbuf), enc->max_payload_size);
enc->frame_samples, GST_BUFFER_DATA (outbuf),
enc->max_payload_size * enc->n_channels);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
@ -582,6 +656,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
ret =
@ -621,7 +696,8 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
enc->headers = NULL;
gst_opus_header_create_caps (&caps, &enc->headers, enc->n_channels,
enc->sample_rate, gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
enc->sample_rate, enc->channel_mapping_family, enc->channel_mapping,
gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)));
/* negotiate with these caps */

3
ext/opus/gstopusenc.h
View file

@ -77,6 +77,9 @@ struct _GstOpusEnc {
GSList *headers;
GstTagList *tags;
guint8 channel_mapping_family;
guint8 channel_mapping[256];
};
struct _GstOpusEncClass {

92
ext/opus/gstopusheader.c
View file

@ -27,7 +27,8 @@
#include "gstopusheader.h"
static GstBuffer *
gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate,
guint8 channel_mapping_family, const guint8 * channel_mapping)
{
GstBuffer *buffer;
GstByteWriter bw;
@ -41,7 +42,12 @@ gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate)
gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip *//* TODO: endianness ? */
gst_byte_writer_put_uint32_le (&bw, sample_rate);
gst_byte_writer_put_uint16_le (&bw, 0); /* output gain *//* TODO: endianness ? */
gst_byte_writer_put_uint8 (&bw, 0); /* channel mapping *//* TODO: what is this ? */
gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
if (channel_mapping_family > 0) {
gst_byte_writer_put_uint8 (&bw, (nchannels + 1) / 2);
gst_byte_writer_put_uint8 (&bw, nchannels / 2);
gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
}
buffer = gst_byte_writer_reset_and_get_buffer (&bw);
@ -136,23 +142,11 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
}
void
gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
gint sample_rate, const GstTagList * tags)
gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
GstBuffer * buf1, GstBuffer * buf2)
{
GstBuffer *buf1, *buf2;
g_return_if_fail (caps);
g_return_if_fail (headers && !*headers);
g_return_if_fail (nchannels > 0);
g_return_if_fail (sample_rate >= 0); /* 0 -> unset */
/* Opus streams in Ogg begin with two headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. */
/* create header buffers */
buf1 = gst_opus_enc_create_id_buffer (nchannels, sample_rate);
buf2 = gst_opus_enc_create_metadata_buffer (tags);
/* mark and put on caps */
*caps = gst_caps_from_string ("audio/x-opus");
@ -162,9 +156,75 @@ gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
*headers = g_slist_prepend (*headers, buf1);
}
void
gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
gint sample_rate, guint8 channel_mapping_family,
const guint8 * channel_mapping, const GstTagList * tags)
{
GstBuffer *buf1, *buf2;
g_return_if_fail (caps);
g_return_if_fail (headers && !*headers);
g_return_if_fail (nchannels > 0);
g_return_if_fail (sample_rate >= 0); /* 0 -> unset */
g_return_if_fail (channel_mapping_family == 0 || channel_mapping);
/* Opus streams in Ogg begin with two headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
bitstream spec. The second header holds any comment fields. */
/* create header buffers */
buf1 =
gst_opus_enc_create_id_buffer (nchannels, sample_rate,
channel_mapping_family, channel_mapping);
buf2 = gst_opus_enc_create_metadata_buffer (tags);
gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
}
gboolean
gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size)
{
return (GST_BUFFER_SIZE (buf) >= magic_size
&& !memcmp (magic, GST_BUFFER_DATA (buf), magic_size));
}
gboolean
gst_opus_header_is_id_header (GstBuffer * buf)
{
gsize size = GST_BUFFER_SIZE (buf);
const guint8 *data = GST_BUFFER_DATA (buf);
guint8 channels, channel_mapping_family, n_streams, n_stereo_streams;
if (size < 19)
return FALSE;
if (!gst_opus_header_is_header (buf, "OpusHead", 8))
return FALSE;
channels = data[9];
if (channels == 0)
return FALSE;
channel_mapping_family = data[18];
if (channel_mapping_family == 0) {
if (channels > 2)
return FALSE;
} else {
channels = data[9];
if (size < 21 + channels)
return FALSE;
n_streams = data[19];
n_stereo_streams = data[20];
if (n_streams == 0)
return FALSE;
if (n_stereo_streams > n_streams)
return FALSE;
if (n_streams + n_stereo_streams > 255)
return FALSE;
}
return TRUE;
}
gboolean
gst_opus_header_is_comment_header (GstBuffer * buf)
{
return gst_opus_header_is_header (buf, "OpusTags", 8);
}

12
ext/opus/gstopusheader.h
View file

@ -25,8 +25,16 @@
G_BEGIN_DECLS
extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers, gint nchannels, gint sample_rate, const GstTagList *tags);
extern gboolean gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size);
extern void gst_opus_header_create_caps_from_headers (GstCaps **caps, GSList **headers,
GstBuffer *id_header, GstBuffer *comment_header);
extern void gst_opus_header_create_caps (GstCaps **caps, GSList **headers,
gint nchannels, gint sample_rate,
guint8 channel_mapping_family, const guint8 *channel_mapping,
const GstTagList *tags);
extern gboolean gst_opus_header_is_header (GstBuffer * buf,
const char *magic, guint magic_size);
extern gboolean gst_opus_header_is_id_header (GstBuffer * buf);
extern gboolean gst_opus_header_is_comment_header (GstBuffer * buf);
G_END_DECLS