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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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291 lines
8.3 KiB
C
291 lines
8.3 KiB
C
/* GStreamer Opus Encoder
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* Copyright (C) <2011> Vincent Penquerc'h <vincent.penquerch@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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/tag/tag.h>
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#include <gst/base/gstbytewriter.h>
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#include "gstopusheader.h"
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static GstBuffer *
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gst_opus_enc_create_id_buffer (gint nchannels, gint n_stereo_streams,
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gint sample_rate, guint8 channel_mapping_family,
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const guint8 * channel_mapping)
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{
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GstBuffer *buffer;
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GstByteWriter bw;
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gboolean hdl = TRUE;
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g_return_val_if_fail (nchannels > 0 && nchannels < 256, NULL);
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g_return_val_if_fail (n_stereo_streams >= 0, NULL);
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g_return_val_if_fail (n_stereo_streams <= nchannels - n_stereo_streams, NULL);
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gst_byte_writer_init (&bw);
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/* See http://wiki.xiph.org/OggOpus */
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hdl &= gst_byte_writer_put_data (&bw, (const guint8 *) "OpusHead", 8);
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hdl &= gst_byte_writer_put_uint8 (&bw, 0x01); /* version number */
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hdl &= gst_byte_writer_put_uint8 (&bw, nchannels);
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hdl &= gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip */
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hdl &= gst_byte_writer_put_uint32_le (&bw, sample_rate);
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hdl &= gst_byte_writer_put_uint16_le (&bw, 0); /* output gain */
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hdl &= gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
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if (channel_mapping_family > 0) {
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hdl &= gst_byte_writer_put_uint8 (&bw, nchannels - n_stereo_streams);
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hdl &= gst_byte_writer_put_uint8 (&bw, n_stereo_streams);
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hdl &= gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
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}
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if (!hdl)
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GST_WARNING ("Error creating header");
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buffer = gst_byte_writer_reset_and_get_buffer (&bw);
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GST_BUFFER_OFFSET (buffer) = 0;
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GST_BUFFER_OFFSET_END (buffer) = 0;
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return buffer;
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}
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static GstBuffer *
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gst_opus_enc_create_metadata_buffer (const GstTagList * tags)
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{
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GstTagList *empty_tags = NULL;
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GstBuffer *comments = NULL;
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GST_DEBUG ("tags = %" GST_PTR_FORMAT, tags);
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if (tags == NULL) {
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/* FIXME: better fix chain of callers to not write metadata at all,
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* if there is none */
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empty_tags = gst_tag_list_new_empty ();
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tags = empty_tags;
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}
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comments =
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gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
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8, "Encoded with GStreamer Opusenc");
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GST_BUFFER_OFFSET (comments) = 0;
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GST_BUFFER_OFFSET_END (comments) = 0;
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if (empty_tags)
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gst_tag_list_unref (empty_tags);
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return comments;
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}
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/*
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* (really really) FIXME: move into core (dixit tpm)
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*/
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/*
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* _gst_caps_set_buffer_array:
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* @caps: (transfer full): a #GstCaps
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* @field: field in caps to set
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* @buf: header buffers
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*
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* Adds given buffers to an array of buffers set as the given @field
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* on the given @caps. List of buffer arguments must be NULL-terminated.
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*
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* Returns: (transfer full): input caps with a streamheader field added, or NULL
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* if some error occurred
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*/
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static GstCaps *
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_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
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GstBuffer * buf, ...)
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{
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GstStructure *structure = NULL;
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va_list va;
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GValue array = { 0 };
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GValue value = { 0 };
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g_return_val_if_fail (caps != NULL, NULL);
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g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
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g_return_val_if_fail (field != NULL, NULL);
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caps = gst_caps_make_writable (caps);
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structure = gst_caps_get_structure (caps, 0);
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g_value_init (&array, GST_TYPE_ARRAY);
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va_start (va, buf);
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/* put buffers in a fixed list */
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while (buf) {
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g_assert (gst_buffer_is_writable (buf));
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/* mark buffer */
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
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g_value_init (&value, GST_TYPE_BUFFER);
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buf = gst_buffer_copy (buf);
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
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gst_value_set_buffer (&value, buf);
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gst_buffer_unref (buf);
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gst_value_array_append_value (&array, &value);
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g_value_unset (&value);
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buf = va_arg (va, GstBuffer *);
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}
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va_end (va);
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gst_structure_set_value (structure, field, &array);
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g_value_unset (&array);
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return caps;
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}
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void
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gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
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GstBuffer * buf1, GstBuffer * buf2)
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{
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int n_streams, family;
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gboolean multistream;
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GstMapInfo map;
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guint8 *data;
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g_return_if_fail (caps);
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g_return_if_fail (headers && !*headers);
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g_return_if_fail (gst_buffer_get_size (buf1) >= 19);
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gst_buffer_map (buf1, &map, GST_MAP_READ);
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data = map.data;
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/* work out the number of streams */
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family = data[18];
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if (family == 0) {
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n_streams = 1;
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} else {
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/* only included in the header for family > 0 */
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if (map.size >= 20)
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n_streams = data[19];
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else {
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g_warning ("family > 0 but header buffer size < 20");
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gst_buffer_unmap (buf1, &map);
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return;
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}
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}
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gst_buffer_unmap (buf1, &map);
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/* mark and put on caps */
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multistream = n_streams > 1;
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*caps = gst_caps_new_simple ("audio/x-opus",
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"multistream", G_TYPE_BOOLEAN, multistream, NULL);
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*caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
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*headers = g_slist_prepend (*headers, gst_buffer_ref (buf2));
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*headers = g_slist_prepend (*headers, gst_buffer_ref (buf1));
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}
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void
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gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
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gint n_stereo_streams, gint sample_rate, guint8 channel_mapping_family,
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const guint8 * channel_mapping, const GstTagList * tags)
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{
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GstBuffer *buf1, *buf2;
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g_return_if_fail (caps);
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g_return_if_fail (headers && !*headers);
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g_return_if_fail (nchannels > 0);
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g_return_if_fail (sample_rate >= 0); /* 0 -> unset */
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g_return_if_fail (channel_mapping_family == 0 || channel_mapping);
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/* Opus streams in Ogg begin with two headers; the initial header (with
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most of the codec setup parameters) which is mandated by the Ogg
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bitstream spec. The second header holds any comment fields. */
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/* create header buffers */
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buf1 =
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gst_opus_enc_create_id_buffer (nchannels, n_stereo_streams, sample_rate,
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channel_mapping_family, channel_mapping);
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buf2 = gst_opus_enc_create_metadata_buffer (tags);
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gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
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gst_buffer_unref (buf2);
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gst_buffer_unref (buf1);
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}
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gboolean
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gst_opus_header_is_header (GstBuffer * buf, const char *magic, guint magic_size)
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{
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return (gst_buffer_get_size (buf) >= magic_size
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&& !gst_buffer_memcmp (buf, 0, magic, magic_size));
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}
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gboolean
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gst_opus_header_is_id_header (GstBuffer * buf)
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{
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gsize size = gst_buffer_get_size (buf);
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guint8 *data = NULL;
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guint8 version, channels, channel_mapping_family, n_streams, n_stereo_streams;
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gboolean ret = FALSE;
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GstMapInfo map;
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if (size < 19)
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goto beach;
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if (!gst_opus_header_is_header (buf, "OpusHead", 8))
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goto beach;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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data = map.data;
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size = map.size;
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version = data[8];
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if (version >= 0x0f) /* major version >=0 is what we grok */
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goto beach;
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channels = data[9];
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if (channels == 0)
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goto beach;
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channel_mapping_family = data[18];
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if (channel_mapping_family == 0) {
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if (channels > 2)
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goto beach;
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} else {
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channels = data[9];
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if (size < 21 + channels)
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goto beach;
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n_streams = data[19];
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n_stereo_streams = data[20];
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if (n_streams == 0)
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goto beach;
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if (n_stereo_streams > n_streams)
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goto beach;
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if (n_streams + n_stereo_streams > 255)
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goto beach;
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}
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ret = TRUE;
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beach:
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if (data)
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gst_buffer_unmap (buf, &map);
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return ret;
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}
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gboolean
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gst_opus_header_is_comment_header (GstBuffer * buf)
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{
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return gst_opus_header_is_header (buf, "OpusTags", 8);
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}
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