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) \
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
--- /dev/null
+/* 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},
+};
--- /dev/null
+/* 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__ */
#include <string.h>
#include <gst/tag/tag.h>
#include "gstopusheader.h"
+#include "gstopuscommon.h"
#include "gstopusdec.h"
GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
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")
);
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);
+ }
+ }
+
+ /* 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);
}
- dec->channel_mapping[0] = 0;
- dec->channel_mapping[1] = 1;
+
+ 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;
}
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)
{
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;
}
int n_channels;
guint32 pre_skip;
gint16 r128_gain;
+
+ guint8 n_streams;
+ guint8 n_stereo_streams;
guint8 channel_mapping_family;
guint8 channel_mapping[256];
#include <gst/gsttagsetter.h>
#include <gst/audio/audio.h>
#include "gstopusheader.h"
+#include "gstopuscommon.h"
#include "gstopusenc.h"
GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
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")
);
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)
{
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);
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)
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);
goto done;
}
+ GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
ret =
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 */
GSList *headers;
GstTagList *tags;
+
+ guint8 channel_mapping_family;
+ guint8 channel_mapping[256];
};
struct _GstOpusEncClass {
#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;
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);
}
void
+gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
+ GstBuffer * buf1, GstBuffer * buf2)
+{
+ g_return_if_fail (caps);
+ g_return_if_fail (headers && !*headers);
+
+ /* mark and put on caps */
+ *caps = gst_caps_from_string ("audio/x-opus");
+ *caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
+
+ *headers = g_slist_prepend (*headers, buf2);
+ *headers = g_slist_prepend (*headers, buf1);
+}
+
+void
gst_opus_header_create_caps (GstCaps ** caps, GSList ** headers, gint nchannels,
- gint sample_rate, const GstTagList * tags)
+ gint sample_rate, guint8 channel_mapping_family,
+ const guint8 * channel_mapping, const GstTagList * tags)
{
GstBuffer *buf1, *buf2;
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);
+ buf1 =
+ gst_opus_enc_create_id_buffer (nchannels, sample_rate,
+ channel_mapping_family, channel_mapping);
buf2 = gst_opus_enc_create_metadata_buffer (tags);
- /* mark and put on caps */
- *caps = gst_caps_from_string ("audio/x-opus");
- *caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
-
- *headers = g_slist_prepend (*headers, buf2);
- *headers = g_slist_prepend (*headers, buf1);
+ gst_opus_header_create_caps_from_headers (caps, headers, buf1, buf2);
}
gboolean
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);
+}
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
projects / platform / upstream / gst-plugins-base.git / commitdiff