1 /* GStreamer Opus Encoder
2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
4 * Copyright (C) <2011> Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * Based on the speexenc element
27 * SECTION:element-opusenc
29 * @see_also: opusdec, oggmux
31 * This element encodes raw audio to OPUS.
33 * ## Example pipelines
35 * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg
37 * Encode a test sine signal to Ogg/OPUS.
50 #include <gst/gsttagsetter.h>
51 #include <gst/audio/audio.h>
52 #include <gst/pbutils/pbutils.h>
53 #include <gst/tag/tag.h>
54 #include <gst/glib-compat-private.h>
56 #include "gstopuselements.h"
57 #include "gstopusheader.h"
58 #include "gstopuscommon.h"
59 #include "gstopusenc.h"
61 GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
62 #define GST_CAT_DEFAULT opusenc_debug
64 /* Some arbitrary bounds beyond which it really doesn't make sense.
65 The spec mentions 6 kb/s to 510 kb/s, so 4000 and 650000 ought to be
66 safe as property bounds. */
67 #define LOWEST_BITRATE 4000
68 #define HIGHEST_BITRATE 650000
70 #define GST_OPUS_ENC_TYPE_BANDWIDTH (gst_opus_enc_bandwidth_get_type())
72 gst_opus_enc_bandwidth_get_type (void)
74 static const GEnumValue values[] = {
75 {OPUS_BANDWIDTH_NARROWBAND, "Narrow band", "narrowband"},
76 {OPUS_BANDWIDTH_MEDIUMBAND, "Medium band", "mediumband"},
77 {OPUS_BANDWIDTH_WIDEBAND, "Wide band", "wideband"},
78 {OPUS_BANDWIDTH_SUPERWIDEBAND, "Super wide band", "superwideband"},
79 {OPUS_BANDWIDTH_FULLBAND, "Full band", "fullband"},
80 {OPUS_AUTO, "Auto", "auto"},
85 if (g_once_init_enter ((gsize *) & id)) {
88 _id = g_enum_register_static ("GstOpusEncBandwidth", values);
90 g_once_init_leave ((gsize *) & id, _id);
96 #define GST_OPUS_ENC_TYPE_FRAME_SIZE (gst_opus_enc_frame_size_get_type())
98 gst_opus_enc_frame_size_get_type (void)
100 static const GEnumValue values[] = {
111 if (g_once_init_enter ((gsize *) & id)) {
114 _id = g_enum_register_static ("GstOpusEncFrameSize", values);
116 g_once_init_leave ((gsize *) & id, _id);
122 #define GST_OPUS_ENC_TYPE_AUDIO_TYPE (gst_opus_enc_audio_type_get_type())
124 gst_opus_enc_audio_type_get_type (void)
126 static const GEnumValue values[] = {
127 {OPUS_APPLICATION_AUDIO, "Generic audio", "generic"},
128 {OPUS_APPLICATION_VOIP, "Voice", "voice"},
129 {OPUS_APPLICATION_RESTRICTED_LOWDELAY, "Restricted low delay",
130 "restricted-lowdelay"},
135 if (g_once_init_enter ((gsize *) & id)) {
138 _id = g_enum_register_static ("GstOpusEncAudioType", values);
140 g_once_init_leave ((gsize *) & id, _id);
146 #define GST_OPUS_ENC_TYPE_BITRATE_TYPE (gst_opus_enc_bitrate_type_get_type())
148 gst_opus_enc_bitrate_type_get_type (void)
150 static const GEnumValue values[] = {
151 {BITRATE_TYPE_CBR, "CBR", "cbr"},
152 {BITRATE_TYPE_VBR, "VBR", "vbr"},
153 {BITRATE_TYPE_CONSTRAINED_VBR, "Constrained VBR", "constrained-vbr"},
158 if (g_once_init_enter ((gsize *) & id)) {
161 _id = g_enum_register_static ("GstOpusEncBitrateType", values);
163 g_once_init_leave ((gsize *) & id, _id);
169 #define FORMAT_STR GST_AUDIO_NE(S16)
170 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
173 GST_STATIC_CAPS ("audio/x-raw, "
174 "format = (string) " FORMAT_STR ", "
175 "layout = (string) interleaved, "
176 "rate = (int) 48000, "
177 "channels = (int) [ 1, 8 ]; "
179 "format = (string) " FORMAT_STR ", "
180 "layout = (string) interleaved, "
181 "rate = (int) { 8000, 12000, 16000, 24000 }, "
182 "channels = (int) [ 1, 8 ] ")
185 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
188 GST_STATIC_CAPS ("audio/x-opus")
191 #define DEFAULT_AUDIO TRUE
192 #define DEFAULT_AUDIO_TYPE OPUS_APPLICATION_AUDIO
193 #define DEFAULT_BITRATE 64000
194 #define DEFAULT_BANDWIDTH OPUS_BANDWIDTH_FULLBAND
195 #define DEFAULT_FRAMESIZE 20
196 #define DEFAULT_CBR TRUE
197 #define DEFAULT_CONSTRAINED_VBR TRUE
198 #define DEFAULT_BITRATE_TYPE BITRATE_TYPE_CONSTRAINED_VBR
199 #define DEFAULT_COMPLEXITY 10
200 #define DEFAULT_INBAND_FEC FALSE
201 #define DEFAULT_DTX FALSE
202 #define DEFAULT_PACKET_LOSS_PERCENT 0
203 #define DEFAULT_MAX_PAYLOAD_SIZE 4000
216 PROP_PACKET_LOSS_PERCENT,
217 PROP_MAX_PAYLOAD_SIZE
220 static void gst_opus_enc_finalize (GObject * object);
222 static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc,
224 static GstCaps *gst_opus_enc_sink_getcaps (GstAudioEncoder * benc,
226 static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
228 static void gst_opus_enc_get_property (GObject * object, guint prop_id,
229 GValue * value, GParamSpec * pspec);
230 static void gst_opus_enc_set_property (GObject * object, guint prop_id,
231 const GValue * value, GParamSpec * pspec);
233 static void gst_opus_enc_set_tags (GstOpusEnc * enc);
234 static gboolean gst_opus_enc_start (GstAudioEncoder * benc);
235 static gboolean gst_opus_enc_stop (GstAudioEncoder * benc);
236 static gboolean gst_opus_enc_set_format (GstAudioEncoder * benc,
237 GstAudioInfo * info);
238 static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc,
240 static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
242 static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer);
244 #define gst_opus_enc_parent_class parent_class
245 G_DEFINE_TYPE_WITH_CODE (GstOpusEnc, gst_opus_enc, GST_TYPE_AUDIO_ENCODER,
246 G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
247 G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
248 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (opusenc, "opusenc",
249 GST_RANK_PRIMARY, GST_TYPE_OPUS_ENC, opus_element_init (plugin));
252 gst_opus_enc_set_tags (GstOpusEnc * enc)
256 /* create a taglist and add a bitrate tag to it */
257 taglist = gst_tag_list_new_empty ();
258 gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
259 GST_TAG_BITRATE, enc->bitrate, NULL);
261 gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (enc), taglist,
262 GST_TAG_MERGE_REPLACE);
264 gst_tag_list_unref (taglist);
268 gst_opus_enc_class_init (GstOpusEncClass * klass)
270 GObjectClass *gobject_class;
271 GstAudioEncoderClass *base_class;
272 GstElementClass *gstelement_class;
274 gobject_class = (GObjectClass *) klass;
275 base_class = (GstAudioEncoderClass *) klass;
276 gstelement_class = (GstElementClass *) klass;
278 gobject_class->set_property = gst_opus_enc_set_property;
279 gobject_class->get_property = gst_opus_enc_get_property;
281 gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
282 gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
283 gst_element_class_set_static_metadata (gstelement_class, "Opus audio encoder",
284 "Codec/Encoder/Audio",
285 "Encodes audio in Opus format",
286 "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");
288 base_class->start = GST_DEBUG_FUNCPTR (gst_opus_enc_start);
289 base_class->stop = GST_DEBUG_FUNCPTR (gst_opus_enc_stop);
290 base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format);
291 base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame);
292 base_class->sink_event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
293 base_class->getcaps = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps);
295 g_object_class_install_property (gobject_class, PROP_AUDIO_TYPE,
296 g_param_spec_enum ("audio-type", "What type of audio to optimize for",
297 "What type of audio to optimize for", GST_OPUS_ENC_TYPE_AUDIO_TYPE,
298 DEFAULT_AUDIO_TYPE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
299 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
300 g_param_spec_int ("bitrate", "Encoding Bit-rate",
301 "Specify an encoding bit-rate (in bps).", LOWEST_BITRATE,
302 HIGHEST_BITRATE, DEFAULT_BITRATE,
303 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
304 GST_PARAM_MUTABLE_PLAYING));
305 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
306 g_param_spec_enum ("bandwidth", "Band Width", "Audio Band Width",
307 GST_OPUS_ENC_TYPE_BANDWIDTH, DEFAULT_BANDWIDTH,
308 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
309 GST_PARAM_MUTABLE_PLAYING));
310 g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
311 g_param_spec_enum ("frame-size", "Frame Size",
312 "The duration of an audio frame, in ms", GST_OPUS_ENC_TYPE_FRAME_SIZE,
314 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
315 GST_PARAM_MUTABLE_PLAYING));
316 g_object_class_install_property (gobject_class, PROP_BITRATE_TYPE,
317 g_param_spec_enum ("bitrate-type", "Bitrate type", "Bitrate type",
318 GST_OPUS_ENC_TYPE_BITRATE_TYPE, DEFAULT_BITRATE_TYPE,
319 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
320 GST_PARAM_MUTABLE_PLAYING));
321 g_object_class_install_property (gobject_class, PROP_COMPLEXITY,
322 g_param_spec_int ("complexity", "Complexity", "Complexity", 0, 10,
324 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
325 GST_PARAM_MUTABLE_PLAYING));
326 g_object_class_install_property (gobject_class, PROP_INBAND_FEC,
327 g_param_spec_boolean ("inband-fec", "In-band FEC",
328 "Enable in-band forward error correction (use in combination with "
329 "the packet-loss-percentage property)", DEFAULT_INBAND_FEC,
330 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
331 GST_PARAM_MUTABLE_PLAYING));
332 g_object_class_install_property (gobject_class, PROP_DTX,
333 g_param_spec_boolean ("dtx", "DTX", "DTX", DEFAULT_DTX,
334 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
335 GST_PARAM_MUTABLE_PLAYING));
336 g_object_class_install_property (G_OBJECT_CLASS (klass),
337 PROP_PACKET_LOSS_PERCENT, g_param_spec_int ("packet-loss-percentage",
338 "Loss percentage", "Packet loss percentage", 0, 100,
339 DEFAULT_PACKET_LOSS_PERCENT,
340 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
341 GST_PARAM_MUTABLE_PLAYING));
342 g_object_class_install_property (G_OBJECT_CLASS (klass),
343 PROP_MAX_PAYLOAD_SIZE, g_param_spec_uint ("max-payload-size",
344 "Max payload size", "Maximum payload size in bytes", 2, 4000,
345 DEFAULT_MAX_PAYLOAD_SIZE,
346 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
347 GST_PARAM_MUTABLE_PLAYING));
349 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize);
351 GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder");
353 gst_type_mark_as_plugin_api (GST_OPUS_ENC_TYPE_AUDIO_TYPE, 0);
354 gst_type_mark_as_plugin_api (GST_OPUS_ENC_TYPE_BANDWIDTH, 0);
355 gst_type_mark_as_plugin_api (GST_OPUS_ENC_TYPE_BITRATE_TYPE, 0);
356 gst_type_mark_as_plugin_api (GST_OPUS_ENC_TYPE_FRAME_SIZE, 0);
360 gst_opus_enc_finalize (GObject * object)
364 enc = GST_OPUS_ENC (object);
366 g_mutex_clear (&enc->property_lock);
368 G_OBJECT_CLASS (parent_class)->finalize (object);
372 gst_opus_enc_init (GstOpusEnc * enc)
374 GST_DEBUG_OBJECT (enc, "init");
376 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
378 g_mutex_init (&enc->property_lock);
380 enc->n_channels = -1;
381 enc->sample_rate = -1;
382 enc->frame_samples = 0;
383 enc->unpositioned = FALSE;
385 enc->bitrate = DEFAULT_BITRATE;
386 enc->bandwidth = DEFAULT_BANDWIDTH;
387 enc->frame_size = DEFAULT_FRAMESIZE;
388 enc->bitrate_type = DEFAULT_BITRATE_TYPE;
389 enc->complexity = DEFAULT_COMPLEXITY;
390 enc->inband_fec = DEFAULT_INBAND_FEC;
391 enc->dtx = DEFAULT_DTX;
392 enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT;
393 enc->max_payload_size = DEFAULT_MAX_PAYLOAD_SIZE;
394 enc->audio_type = DEFAULT_AUDIO_TYPE;
398 gst_opus_enc_start (GstAudioEncoder * benc)
400 GstOpusEnc *enc = GST_OPUS_ENC (benc);
402 GST_DEBUG_OBJECT (enc, "start");
403 enc->encoded_samples = 0;
404 enc->consumed_samples = 0;
410 gst_opus_enc_stop (GstAudioEncoder * benc)
412 GstOpusEnc *enc = GST_OPUS_ENC (benc);
414 GST_DEBUG_OBJECT (enc, "stop");
416 opus_multistream_encoder_destroy (enc->state);
419 gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
425 gst_opus_enc_get_latency (GstOpusEnc * enc)
427 gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
429 GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
434 gst_opus_enc_setup_base_class (GstOpusEnc * enc, GstAudioEncoder * benc)
436 gst_audio_encoder_set_latency (benc,
437 gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
438 gst_audio_encoder_set_frame_samples_min (benc, enc->frame_samples);
439 gst_audio_encoder_set_frame_samples_max (benc, enc->frame_samples);
440 gst_audio_encoder_set_frame_max (benc, 1);
444 gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
446 gint frame_samples = 0;
447 switch (enc->frame_size) {
449 frame_samples = enc->sample_rate / 400;
452 frame_samples = enc->sample_rate / 200;
455 frame_samples = enc->sample_rate / 100;
458 frame_samples = enc->sample_rate / 50;
461 frame_samples = enc->sample_rate / 25;
464 frame_samples = 3 * enc->sample_rate / 50;
467 GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size);
471 return frame_samples;
475 gst_opus_enc_setup_trivial_mapping (GstOpusEnc * enc, guint8 mapping[256])
479 for (n = 0; n < 255; ++n)
484 gst_opus_enc_find_channel_position (GstOpusEnc * enc, const GstAudioInfo * info,
485 GstAudioChannelPosition position)
488 for (n = 0; n < enc->n_channels; ++n) {
489 if (GST_AUDIO_INFO_POSITION (info, n) == position) {
497 gst_opus_enc_find_channel_position_in_vorbis_order (GstOpusEnc * enc,
498 GstAudioChannelPosition position)
502 for (c = 0; c < enc->n_channels; ++c) {
503 if (gst_opus_channel_positions[enc->n_channels - 1][c] == position) {
504 GST_INFO_OBJECT (enc,
505 "Channel position %s maps to index %d in Vorbis order",
506 gst_opus_channel_names[position], c);
510 GST_WARNING_OBJECT (enc,
511 "Channel position %s is not representable in Vorbis order",
512 gst_opus_channel_names[position]);
517 gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
518 const GstAudioInfo * info)
520 #define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
524 GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
527 /* Start by setting up a default trivial mapping */
528 enc->n_stereo_streams = 0;
529 gst_opus_enc_setup_trivial_mapping (enc, enc->encoding_channel_mapping);
530 gst_opus_enc_setup_trivial_mapping (enc, enc->decoding_channel_mapping);
532 /* For one channel, use the basic RTP mapping */
533 if (enc->n_channels == 1 && !enc->unpositioned) {
534 GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
535 enc->channel_mapping_family = 0;
536 /* implicit mapping for family 0 */
540 /* For two channels, use the basic RTP mapping if the channels are
541 mapped as left/right. */
542 if (enc->n_channels == 2 && !enc->unpositioned) {
543 GST_INFO_OBJECT (enc, "Stereo, trivial RTP mapping");
544 enc->channel_mapping_family = 0;
545 enc->n_stereo_streams = 1;
546 /* implicit mapping for family 0 */
550 /* For channels between 3 and 8, we use the Vorbis mapping if we can
551 find a permutation that matches it. Mono and stereo will have been taken
552 care of earlier, but this code also handles it. There are two mappings.
553 One maps the input channels to an ordering which has the natural pairs
554 first so they can benefit from the Opus stereo channel coupling, and the
555 other maps this ordering to the Vorbis ordering. */
556 if (enc->n_channels >= 3 && enc->n_channels <= 8 && !enc->unpositioned) {
557 int c0, c1, c0v, c1v;
559 gboolean positions_done[256];
560 static const GstAudioChannelPosition pairs[][2] = {
561 {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
562 GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
563 {GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
564 GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
565 {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
566 GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
567 {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
568 GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
569 {GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
570 GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
571 {GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
572 GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
576 GST_DEBUG_OBJECT (enc,
577 "In range for the Vorbis mapping, building channel mapping tables");
579 enc->n_stereo_streams = 0;
581 for (n = 0; n < 256; ++n)
582 positions_done[n] = FALSE;
584 /* First, find any natural pairs, and move them to the front */
585 for (pair = 0; pair < G_N_ELEMENTS (pairs); ++pair) {
586 GstAudioChannelPosition p0 = pairs[pair][0];
587 GstAudioChannelPosition p1 = pairs[pair][1];
588 c0 = gst_opus_enc_find_channel_position (enc, info, p0);
589 c1 = gst_opus_enc_find_channel_position (enc, info, p1);
590 if (c0 >= 0 && c1 >= 0) {
591 /* We found a natural pair */
592 GST_DEBUG_OBJECT (enc, "Natural pair '%s/%s' found at %d %d",
593 gst_opus_channel_names[p0], gst_opus_channel_names[p1], c0, c1);
594 /* Find where they map in Vorbis order */
595 c0v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p0);
596 c1v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p1);
597 if (c0v < 0 || c1v < 0) {
598 GST_WARNING_OBJECT (enc,
599 "Cannot map channel positions to Vorbis order, using unknown mapping");
600 enc->channel_mapping_family = 255;
601 enc->n_stereo_streams = 0;
605 enc->encoding_channel_mapping[mapped] = c0;
606 enc->encoding_channel_mapping[mapped + 1] = c1;
607 enc->decoding_channel_mapping[c0v] = mapped;
608 enc->decoding_channel_mapping[c1v] = mapped + 1;
609 enc->n_stereo_streams++;
611 positions_done[p0] = positions_done[p1] = TRUE;
615 /* Now add all other input channels as mono streams */
616 for (n = 0; n < enc->n_channels; ++n) {
617 GstAudioChannelPosition position = GST_AUDIO_INFO_POSITION (info, n);
619 /* if we already mapped it while searching for pairs, nothing else
621 if (!positions_done[position]) {
623 GST_DEBUG_OBJECT (enc, "Channel position %s is not mapped yet, adding",
624 gst_opus_channel_names[position]);
625 cv = gst_opus_enc_find_channel_position_in_vorbis_order (enc, position);
627 g_assert_not_reached ();
628 enc->encoding_channel_mapping[mapped] = n;
629 enc->decoding_channel_mapping[cv] = mapped;
634 #ifndef GST_DISABLE_GST_DEBUG
635 GST_INFO_OBJECT (enc,
636 "Mapping tables built: %d channels, %d stereo streams", enc->n_channels,
637 enc->n_stereo_streams);
638 gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
639 "Encoding mapping table", enc->n_channels,
640 enc->encoding_channel_mapping);
641 gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
642 "Decoding mapping table", enc->n_channels,
643 enc->decoding_channel_mapping);
646 enc->channel_mapping_family = 1;
650 /* More than 8 channels, if future mappings are added for those */
652 /* For other cases, we use undefined, with the default trivial mapping
653 and all mono streams */
654 if (!enc->unpositioned)
655 GST_WARNING_OBJECT (enc, "Unknown mapping");
657 GST_INFO_OBJECT (enc, "Unpositioned mapping, all channels mono");
659 enc->channel_mapping_family = 255;
660 enc->n_stereo_streams = 0;
666 gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
670 enc = GST_OPUS_ENC (benc);
672 g_mutex_lock (&enc->property_lock);
674 enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
675 enc->unpositioned = GST_AUDIO_INFO_IS_UNPOSITIONED (info);
676 enc->sample_rate = GST_AUDIO_INFO_RATE (info);
677 gst_opus_enc_setup_channel_mappings (enc, info);
678 GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
681 /* handle reconfigure */
683 opus_multistream_encoder_destroy (enc->state);
686 if (!gst_opus_enc_setup (enc)) {
687 g_mutex_unlock (&enc->property_lock);
691 /* update the tags */
692 gst_opus_enc_set_tags (enc);
694 enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
696 /* feedback to base class */
697 gst_opus_enc_setup_base_class (enc, benc);
699 g_mutex_unlock (&enc->property_lock);
705 gst_opus_enc_setup (GstOpusEnc * enc)
711 const GstTagList *tags;
712 GstTagList *empty_tags = NULL;
713 GstBuffer *header, *comments;
715 #ifndef GST_DISABLE_GST_DEBUG
716 GST_DEBUG_OBJECT (enc,
717 "setup: %d Hz, %d channels, %d stereo streams, family %d",
718 enc->sample_rate, enc->n_channels, enc->n_stereo_streams,
719 enc->channel_mapping_family);
720 GST_INFO_OBJECT (enc, "Mapping tables built: %d channels, %d stereo streams",
721 enc->n_channels, enc->n_stereo_streams);
722 gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
723 "Encoding mapping table", enc->n_channels, enc->encoding_channel_mapping);
724 gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
725 "Decoding mapping table", enc->n_channels, enc->decoding_channel_mapping);
728 enc->state = opus_multistream_encoder_create (enc->sample_rate,
729 enc->n_channels, enc->n_channels - enc->n_stereo_streams,
730 enc->n_stereo_streams, enc->encoding_channel_mapping,
731 enc->audio_type, &error);
732 if (!enc->state || error != OPUS_OK)
733 goto encoder_creation_failed;
735 opus_multistream_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
736 opus_multistream_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth),
738 opus_multistream_encoder_ctl (enc->state,
739 OPUS_SET_VBR (enc->bitrate_type != BITRATE_TYPE_CBR), 0);
740 opus_multistream_encoder_ctl (enc->state,
741 OPUS_SET_VBR_CONSTRAINT (enc->bitrate_type ==
742 BITRATE_TYPE_CONSTRAINED_VBR), 0);
743 opus_multistream_encoder_ctl (enc->state,
744 OPUS_SET_COMPLEXITY (enc->complexity), 0);
745 opus_multistream_encoder_ctl (enc->state,
746 OPUS_SET_INBAND_FEC (enc->inband_fec), 0);
747 opus_multistream_encoder_ctl (enc->state, OPUS_SET_DTX (enc->dtx), 0);
748 opus_multistream_encoder_ctl (enc->state,
749 OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
751 opus_multistream_encoder_ctl (enc->state, OPUS_GET_LOOKAHEAD (&lookahead), 0);
753 GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
756 /* lookahead is samples, the Opus header wants it in 48kHz samples */
757 lookahead = lookahead * 48000 / enc->sample_rate;
758 enc->lookahead = enc->pending_lookahead = lookahead;
760 header = gst_codec_utils_opus_create_header (enc->sample_rate,
761 enc->n_channels, enc->channel_mapping_family,
762 enc->n_channels - enc->n_stereo_streams, enc->n_stereo_streams,
763 enc->decoding_channel_mapping, lookahead, 0);
764 tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
766 tags = empty_tags = gst_tag_list_new_empty ();
768 gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags",
769 8, "Encoded with GStreamer opusenc");
770 caps = gst_codec_utils_opus_create_caps_from_header (header, comments);
772 gst_tag_list_unref (empty_tags);
773 gst_buffer_unref (header);
774 gst_buffer_unref (comments);
776 /* negotiate with these caps */
777 GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
779 ret = gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
780 gst_caps_unref (caps);
784 encoder_creation_failed:
785 GST_ERROR_OBJECT (enc, "Encoder creation failed");
790 gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
794 enc = GST_OPUS_ENC (benc);
796 GST_DEBUG_OBJECT (enc, "sink event: %s", GST_EVENT_TYPE_NAME (event));
797 switch (GST_EVENT_TYPE (event)) {
801 GstTagSetter *setter = GST_TAG_SETTER (enc);
802 const GstTagMergeMode mode = gst_tag_setter_get_tag_merge_mode (setter);
804 gst_event_parse_tag (event, &list);
805 gst_tag_setter_merge_tags (setter, list, mode);
808 case GST_EVENT_SEGMENT:
809 enc->encoded_samples = 0;
810 enc->consumed_samples = 0;
817 return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
821 gst_opus_enc_get_sink_template_caps (void)
823 static gsize init = 0;
824 static GstCaps *caps = NULL;
826 if (g_once_init_enter (&init)) {
827 GValue rate_array = G_VALUE_INIT;
828 GValue v = G_VALUE_INIT;
829 GstStructure *s1, *s2, *s;
832 caps = gst_caps_new_empty ();
834 /* The caps is cached */
835 GST_MINI_OBJECT_FLAG_SET (caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
837 /* Generate our two template structures */
838 g_value_init (&rate_array, GST_TYPE_LIST);
839 g_value_init (&v, G_TYPE_INT);
840 g_value_set_int (&v, 8000);
841 gst_value_list_append_value (&rate_array, &v);
842 g_value_set_int (&v, 12000);
843 gst_value_list_append_value (&rate_array, &v);
844 g_value_set_int (&v, 16000);
845 gst_value_list_append_value (&rate_array, &v);
846 g_value_set_int (&v, 24000);
847 gst_value_list_append_value (&rate_array, &v);
849 s1 = gst_structure_new ("audio/x-raw",
850 "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
851 "layout", G_TYPE_STRING, "interleaved",
852 "rate", G_TYPE_INT, 48000, NULL);
853 s2 = gst_structure_new ("audio/x-raw",
854 "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
855 "layout", G_TYPE_STRING, "interleaved", NULL);
856 gst_structure_set_value (s2, "rate", &rate_array);
857 g_value_unset (&rate_array);
860 /* Stereo and further */
861 for (i = 8; i >= 2; i--) {
862 guint64 channel_mask = 0;
863 const GstAudioChannelPosition *pos = gst_opus_channel_positions[i - 1];
865 for (c = 0; c < i; c++) {
866 channel_mask |= G_GUINT64_CONSTANT (1) << pos[c];
869 s = gst_structure_copy (s1);
870 gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
871 GST_TYPE_BITMASK, channel_mask, NULL);
872 gst_caps_append_structure (caps, s);
874 s = gst_structure_copy (s2);
875 gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
876 GST_TYPE_BITMASK, channel_mask, NULL);
877 gst_caps_append_structure (caps, s);
879 /* We also allow unpositioned channels, input will be
880 * treated as a set of individual mono channels */
881 s = gst_structure_copy (s2);
882 gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
883 GST_TYPE_BITMASK, G_GUINT64_CONSTANT (0), NULL);
884 gst_caps_append_structure (caps, s);
886 s = gst_structure_copy (s1);
887 gst_structure_set (s, "channels", G_TYPE_INT, i, "channel-mask",
888 GST_TYPE_BITMASK, G_GUINT64_CONSTANT (0), NULL);
889 gst_caps_append_structure (caps, s);
893 s = gst_structure_copy (s1);
894 gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
895 gst_caps_append_structure (caps, s);
897 s = gst_structure_copy (s2);
898 gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
899 gst_caps_append_structure (caps, s);
900 gst_structure_free (s1);
901 gst_structure_free (s2);
903 g_once_init_leave (&init, 1);
910 gst_opus_enc_sink_getcaps (GstAudioEncoder * benc, GstCaps * filter)
915 enc = GST_OPUS_ENC (benc);
917 GST_DEBUG_OBJECT (enc, "sink getcaps");
919 caps = gst_opus_enc_get_sink_template_caps ();
920 caps = gst_audio_encoder_proxy_getcaps (benc, caps, filter);
922 GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
928 gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
930 guint8 *bdata = NULL, *data, *mdata = NULL;
933 gint ret = GST_FLOW_OK;
938 guint64 trim_start = 0, trim_end = 0;
940 guint max_payload_size;
941 gint frame_samples, input_samples, output_samples;
943 g_mutex_lock (&enc->property_lock);
945 bytes = enc->frame_samples * enc->n_channels * 2;
946 max_payload_size = enc->max_payload_size;
947 frame_samples = input_samples = enc->frame_samples;
949 g_mutex_unlock (&enc->property_lock);
951 if (G_LIKELY (buf)) {
952 gst_buffer_map (buf, &map, GST_MAP_READ);
956 if (G_UNLIKELY (bsize % bytes)) {
959 GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
960 g_assert (bsize < bytes);
962 input_samples = bsize / (enc->n_channels * 2);
964 (enc->encoded_samples + frame_samples) - (enc->consumed_samples +
967 GST_DEBUG_OBJECT (enc,
968 "%" G_GINT64_FORMAT " extra samples of padding in this frame",
970 output_samples = frame_samples - diff;
971 trim_end = diff * 48000 / enc->sample_rate;
973 GST_DEBUG_OBJECT (enc,
974 "Need to add %" G_GINT64_FORMAT " extra samples in the next frame",
976 output_samples = frame_samples;
979 size = ((bsize / bytes) + 1) * bytes;
980 mdata = g_malloc0 (size);
981 /* FIXME: Instead of silence, use LPC with the last real samples.
982 * Otherwise we will create a discontinuity here, which will distort the
983 * last few encoded samples
985 memcpy (mdata, bdata, bsize);
991 /* Adjust for lookahead here */
992 if (enc->pending_lookahead) {
993 guint scaled_lookahead =
994 enc->pending_lookahead * enc->sample_rate / 48000;
996 if (input_samples > scaled_lookahead) {
997 output_samples = input_samples - scaled_lookahead;
998 trim_start = enc->pending_lookahead;
999 enc->pending_lookahead = 0;
1001 trim_start = ((guint64) input_samples) * 48000 / enc->sample_rate;
1002 enc->pending_lookahead -= trim_start;
1006 output_samples = input_samples;
1010 if (enc->encoded_samples < enc->consumed_samples) {
1011 /* FIXME: Instead of silence, use LPC with the last real samples.
1012 * Otherwise we will create a discontinuity here, which will distort the
1013 * last few encoded samples
1015 data = mdata = g_malloc0 (bytes);
1017 output_samples = enc->consumed_samples - enc->encoded_samples;
1019 GST_DEBUG_OBJECT (enc, "draining %d samples", output_samples);
1021 ((guint64) frame_samples - output_samples) * 48000 / enc->sample_rate;
1022 } else if (enc->encoded_samples == enc->consumed_samples) {
1023 GST_DEBUG_OBJECT (enc, "nothing to drain");
1026 g_assert_not_reached ();
1031 g_assert (size == bytes);
1034 gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER (enc),
1035 max_payload_size * enc->n_channels);
1039 GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
1040 frame_samples, (int) bytes);
1042 if (trim_start || trim_end) {
1043 GST_DEBUG_OBJECT (enc,
1044 "Adding trim-start %" G_GUINT64_FORMAT " trim-end %" G_GUINT64_FORMAT,
1045 trim_start, trim_end);
1046 gst_buffer_add_audio_clipping_meta (outbuf, GST_FORMAT_DEFAULT, trim_start,
1050 gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
1053 opus_multistream_encode (enc->state, (const gint16 *) data,
1054 frame_samples, omap.data, max_payload_size * enc->n_channels);
1056 gst_buffer_unmap (outbuf, &omap);
1059 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
1060 ("Encoding failed (%d): %s", outsize, opus_strerror (outsize)));
1061 ret = GST_FLOW_ERROR;
1063 } else if (outsize > max_payload_size) {
1064 GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
1065 ("Encoded size %d is higher than max payload size (%d bytes)",
1066 outsize, max_payload_size));
1067 ret = GST_FLOW_ERROR;
1071 GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
1072 gst_buffer_set_size (outbuf, outsize);
1076 gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
1078 enc->encoded_samples += output_samples;
1079 enc->consumed_samples += input_samples;
1084 gst_buffer_unmap (buf, &map);
1091 static GstFlowReturn
1092 gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
1095 GstFlowReturn ret = GST_FLOW_OK;
1097 enc = GST_OPUS_ENC (benc);
1098 GST_DEBUG_OBJECT (enc, "handle_frame");
1099 GST_DEBUG_OBJECT (enc, "received buffer %p of %" G_GSIZE_FORMAT " bytes", buf,
1100 buf ? gst_buffer_get_size (buf) : 0);
1102 ret = gst_opus_enc_encode (enc, buf);
1108 gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value,
1113 enc = GST_OPUS_ENC (object);
1115 g_mutex_lock (&enc->property_lock);
1118 case PROP_AUDIO_TYPE:
1119 g_value_set_enum (value, enc->audio_type);
1122 g_value_set_int (value, enc->bitrate);
1124 case PROP_BANDWIDTH:
1125 g_value_set_enum (value, enc->bandwidth);
1127 case PROP_FRAME_SIZE:
1128 g_value_set_enum (value, enc->frame_size);
1130 case PROP_BITRATE_TYPE:
1131 g_value_set_enum (value, enc->bitrate_type);
1133 case PROP_COMPLEXITY:
1134 g_value_set_int (value, enc->complexity);
1136 case PROP_INBAND_FEC:
1137 g_value_set_boolean (value, enc->inband_fec);
1140 g_value_set_boolean (value, enc->dtx);
1142 case PROP_PACKET_LOSS_PERCENT:
1143 g_value_set_int (value, enc->packet_loss_percentage);
1145 case PROP_MAX_PAYLOAD_SIZE:
1146 g_value_set_uint (value, enc->max_payload_size);
1149 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1153 g_mutex_unlock (&enc->property_lock);
1157 gst_opus_enc_set_property (GObject * object, guint prop_id,
1158 const GValue * value, GParamSpec * pspec)
1162 enc = GST_OPUS_ENC (object);
1164 #define GST_OPUS_UPDATE_PROPERTY(prop,type,ctl) do { \
1165 g_mutex_lock (&enc->property_lock); \
1166 enc->prop = g_value_get_##type (value); \
1168 opus_multistream_encoder_ctl (enc->state, OPUS_SET_##ctl (enc->prop)); \
1170 g_mutex_unlock (&enc->property_lock); \
1174 case PROP_AUDIO_TYPE:
1175 enc->audio_type = g_value_get_enum (value);
1178 GST_OPUS_UPDATE_PROPERTY (bitrate, int, BITRATE);
1180 case PROP_BANDWIDTH:
1181 GST_OPUS_UPDATE_PROPERTY (bandwidth, enum, BANDWIDTH);
1183 case PROP_FRAME_SIZE:
1184 g_mutex_lock (&enc->property_lock);
1185 enc->frame_size = g_value_get_enum (value);
1186 enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
1187 gst_opus_enc_setup_base_class (enc, GST_AUDIO_ENCODER (enc));
1188 g_mutex_unlock (&enc->property_lock);
1190 case PROP_BITRATE_TYPE:
1191 /* this one has an opposite meaning to the opus ctl... */
1192 g_mutex_lock (&enc->property_lock);
1193 enc->bitrate_type = g_value_get_enum (value);
1195 opus_multistream_encoder_ctl (enc->state,
1196 OPUS_SET_VBR (enc->bitrate_type != BITRATE_TYPE_CBR));
1197 opus_multistream_encoder_ctl (enc->state,
1198 OPUS_SET_VBR_CONSTRAINT (enc->bitrate_type ==
1199 BITRATE_TYPE_CONSTRAINED_VBR), 0);
1201 g_mutex_unlock (&enc->property_lock);
1203 case PROP_COMPLEXITY:
1204 GST_OPUS_UPDATE_PROPERTY (complexity, int, COMPLEXITY);
1206 case PROP_INBAND_FEC:
1207 GST_OPUS_UPDATE_PROPERTY (inband_fec, boolean, INBAND_FEC);
1210 GST_OPUS_UPDATE_PROPERTY (dtx, boolean, DTX);
1212 case PROP_PACKET_LOSS_PERCENT:
1213 GST_OPUS_UPDATE_PROPERTY (packet_loss_percentage, int, PACKET_LOSS_PERC);
1215 case PROP_MAX_PAYLOAD_SIZE:
1216 g_mutex_lock (&enc->property_lock);
1217 enc->max_payload_size = g_value_get_uint (value);
1218 g_mutex_unlock (&enc->property_lock);
1221 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1225 #undef GST_OPUS_UPDATE_PROPERTY