mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
opus: multichannel support
This commit is contained in:
Vincent Penquerc'h
parent
670c365400
commit
d38f4b8a09
9 changed files with 369 additions and 91 deletions
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@ -1,6 +1,6 @@
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plugin_LTLIBRARIES = libgstopus.la
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libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c
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libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c gstopusparse.c gstopusheader.c gstopuscommon.c
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libgstopus_la_CFLAGS = \
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-DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) \
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@ -15,4 +15,4 @@ libgstopus_la_LIBADD = \
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libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM)
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libgstopus_la_LIBTOOLFLAGS = --tag=disable-static
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noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h
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noinst_HEADERS = gstopusenc.h gstopusdec.h gstopusparse.h gstopusheader.h gstopuscommon.h
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72
ext/opus/gstopuscommon.c
Normal file
72
ext/opus/gstopuscommon.c
Normal file
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@ -0,0 +1,72 @@
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/* GStreamer
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* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include "gstopuscommon.h"
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/* http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9 */
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/* copy of the same structure in the vorbis plugin */
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const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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{ /* Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* Stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Stereo + Centre */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* Quadraphonic */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Stereo + Centre + rear stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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},
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{ /* Full 5.1 Surround */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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},
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{ /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE},
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{ /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE},
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};
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33
ext/opus/gstopuscommon.h
Normal file
33
ext/opus/gstopuscommon.h
Normal file
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@ -0,0 +1,33 @@
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/* GStreamer Opus Encoder
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* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_OPUS_COMMON_H__
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#define __GST_OPUS_COMMON_H__
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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G_BEGIN_DECLS
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extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
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G_END_DECLS
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#endif /* __GST_OPUS_COMMON_H__ */
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@ -45,6 +45,7 @@
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#include <string.h>
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#include <gst/tag/tag.h>
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#include "gstopusheader.h"
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#include "gstopuscommon.h"
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#include "gstopusdec.h"
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GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
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@ -56,7 +57,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
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"channels = (int) [ 1, 2 ], "
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"channels = (int) [ 1, 8 ], "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
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);
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@ -217,26 +218,91 @@ gst_opus_dec_get_r128_volume (gint16 r128_gain)
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static GstFlowReturn
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gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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g_return_val_if_fail (gst_opus_header_is_header (buf, "OpusHead", 8),
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GST_FLOW_ERROR);
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g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 19, GST_FLOW_ERROR);
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const guint8 *data = GST_BUFFER_DATA (buf);
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GstCaps *caps;
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GstStructure *s;
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const GstAudioChannelPosition *pos = NULL;
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dec->pre_skip = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 10);
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dec->r128_gain = GST_READ_UINT16_LE (GST_BUFFER_DATA (buf) + 14);
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g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
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g_return_val_if_fail (dec->n_channels != data[9], GST_FLOW_ERROR);
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dec->n_channels = data[9];
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dec->pre_skip = GST_READ_UINT16_LE (data + 10);
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dec->r128_gain = GST_READ_UINT16_LE (data + 14);
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dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
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GST_INFO_OBJECT (dec,
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"Found pre-skip of %u samples, R128 gain %d (volume %f)",
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dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
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dec->channel_mapping_family = GST_BUFFER_DATA (buf)[18];
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if (dec->channel_mapping_family != 0) {
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GST_ELEMENT_ERROR (dec, STREAM, DECODE,
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("Decoding error: unsupported channel nmapping family %d",
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dec->channel_mapping_family), (NULL));
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return GST_FLOW_ERROR;
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}
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dec->channel_mapping_family = data[18];
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if (dec->channel_mapping_family == 0) {
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/* implicit mapping */
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GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
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dec->n_streams = dec->n_stereo_streams = 1;
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dec->channel_mapping[0] = 0;
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dec->channel_mapping[1] = 1;
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} else {
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dec->n_streams = data[19];
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dec->n_stereo_streams = data[20];
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memcpy (dec->channel_mapping, data + 21, dec->n_channels);
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if (dec->channel_mapping_family == 1) {
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GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
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switch (dec->n_channels) {
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case 1:
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case 2:
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/* nothing */
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break;
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case 3:
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case 4:
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case 5:
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case 6:
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case 7:
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case 8:
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pos = gst_opus_channel_positions[dec->n_channels - 1];
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break;
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default:{
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gint i;
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GstAudioChannelPosition *posn =
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g_new (GstAudioChannelPosition, dec->n_channels);
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GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("Using NONE channel layout for more than 8 channels"));
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for (i = 0; i < dec->n_channels; i++)
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posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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pos = posn;
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}
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}
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} else {
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GST_INFO_OBJECT (dec, "Channel mapping family %d",
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dec->channel_mapping_family);
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}
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}
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/* negotiate width with downstream */
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caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
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s = gst_caps_get_structure (caps, 0);
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gst_structure_fixate_field_nearest_int (s, "rate", 48000);
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gst_structure_get_int (s, "rate", &dec->sample_rate);
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gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
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gst_structure_get_int (s, "channels", &dec->n_channels);
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GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
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dec->sample_rate);
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if (pos) {
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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}
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if (dec->n_channels > 8) {
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g_free ((GstAudioChannelPosition *) pos);
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}
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GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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return GST_FLOW_OK;
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}
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@ -248,48 +314,6 @@ gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
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return GST_FLOW_OK;
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}
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static void
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gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
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{
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GstPad *srcpad, *peer;
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GstStructure *s;
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GstCaps *caps;
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const GstCaps *template_caps;
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const GstCaps *peer_caps;
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srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
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peer = gst_pad_get_peer (srcpad);
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if (peer) {
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template_caps = gst_pad_get_pad_template_caps (srcpad);
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peer_caps = gst_pad_get_caps (peer);
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GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
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caps = gst_caps_intersect (template_caps, peer_caps);
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gst_pad_fixate_caps (peer, caps);
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GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
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dec->n_channels = 2;
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GST_WARNING_OBJECT (dec, "Failed to get channels, using default %d",
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dec->n_channels);
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} else {
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GST_DEBUG_OBJECT (dec, "Got channels %d", dec->n_channels);
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}
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if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
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dec->sample_rate = 48000;
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GST_WARNING_OBJECT (dec, "Failed to get rate, using default %d",
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dec->sample_rate);
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} else {
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GST_DEBUG_OBJECT (dec, "Got sample rate %d", dec->sample_rate);
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}
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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} else {
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GST_WARNING_OBJECT (dec, "Failed to get src pad peer");
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}
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}
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static GstFlowReturn
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opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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{
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@ -304,12 +328,11 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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GstBuffer *buf;
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if (dec->state == NULL) {
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gst_opus_dec_setup_from_peer_caps (dec);
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GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
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dec->n_channels, dec->sample_rate);
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dec->state = opus_multistream_decoder_create (dec->sample_rate,
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dec->n_channels, 1, 1, dec->channel_mapping, &err);
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dec->n_channels, dec->n_streams, dec->n_stereo_streams,
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dec->channel_mapping, &err);
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if (!dec->state || err != OPUS_OK)
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goto creation_failed;
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}
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|
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@ -55,6 +55,9 @@ struct _GstOpusDec {
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int n_channels;
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guint32 pre_skip;
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gint16 r128_gain;
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guint8 n_streams;
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guint8 n_stereo_streams;
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guint8 channel_mapping_family;
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guint8 channel_mapping[256];
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@ -49,6 +49,7 @@
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#include <gst/gsttagsetter.h>
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#include <gst/audio/audio.h>
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#include "gstopusheader.h"
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#include "gstopuscommon.h"
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#include "gstopusenc.h"
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GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
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@ -116,8 +117,8 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
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"channels = (int) [ 1, 8 ], "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
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);
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@ -419,6 +420,82 @@ gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
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return frame_samples;
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}
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static void
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gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
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{
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#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
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int n;
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GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
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enc->n_channels);
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/* Start by setting up a default trivial mapping */
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for (n = 0; n < 255; ++n)
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enc->channel_mapping[n] = n;
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/* For one channel, use the basic RTP mapping */
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if (enc->n_channels == 1) {
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GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
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enc->channel_mapping_family = 0;
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enc->channel_mapping[0] = 0;
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return;
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}
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/* For two channels, use the basic RTP mapping if the channels are
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mapped as left/right. */
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if (enc->n_channels == 2) {
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if (MAPS (0, FRONT_LEFT) && MAPS (1, FRONT_RIGHT)) {
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GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
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enc->channel_mapping_family = 0;
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/* The channel mapping is implicit for family 0, that's why we do not
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attempt to create one for right/left - this will be mapped to the
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Vorbis mapping below. */
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} else {
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GST_DEBUG_OBJECT (enc, "Stereo, but not canonical mapping, continuing");
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}
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}
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/* For channels between 1 and 8, we use the Vorbis mapping if we can
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find a permutation that matches it. Mono will have been taken care
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of earlier, but this code also handles it. */
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if (enc->n_channels >= 1 && enc->n_channels <= 8) {
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GST_DEBUG_OBJECT (enc,
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"In range for the Vorbis mapping, checking channel positions");
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for (n = 0; n < enc->n_channels; ++n) {
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GstAudioChannelPosition pos = GST_AUDIO_INFO_POSITION (info, n);
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int c;
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GST_DEBUG_OBJECT (enc, "Channel %d has position %d", n, pos);
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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 */
|
||||
|
|
|
|||
|
|
@ -77,6 +77,9 @@ struct _GstOpusEnc {
|
|||
GSList *headers;
|
||||
|
||||
GstTagList *tags;
|
||||
|
||||
guint8 channel_mapping_family;
|
||||
guint8 channel_mapping[256];
|
||||
};
|
||||
|
||||
struct _GstOpusEncClass {
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in a new issue