1/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
2 Written by Jean-Marc Valin and Koen Vos */
3/*
4 Redistribution and use in source and binary forms, with or without
5 modification, are permitted provided that the following conditions
6 are met:
7
8 - Redistributions of source code must retain the above copyright
9 notice, this list of conditions and the following disclaimer.
10
11 - Redistributions in binary form must reproduce the above copyright
12 notice, this list of conditions and the following disclaimer in the
13 documentation and/or other materials provided with the distribution.
14
15 THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
16 ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
17 LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
18 A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
19 OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
20 EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
22 PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
23 LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
24 NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
25 SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26*/
27
28/**
29 * @file opus.h
30 * @brief Opus reference implementation API
31 */
32
33#ifndef OPUS_H
34#define OPUS_H
35
36#include "opus_types.h"
37#include "opus_defines.h"
38
39#ifdef __cplusplus
40extern "C" {
41#endif
42
43/**
44 * @mainpage Opus
45 *
46 * The Opus codec is designed for interactive speech and audio transmission over the Internet.
47 * It is designed by the IETF Codec Working Group and incorporates technology from
48 * Skype's SILK codec and Xiph.Org's CELT codec.
49 *
50 * The Opus codec is designed to handle a wide range of interactive audio applications,
51 * including Voice over IP, videoconferencing, in-game chat, and even remote live music
52 * performances. It can scale from low bit-rate narrowband speech to very high quality
53 * stereo music. Its main features are:
54
55 * @li Sampling rates from 8 to 48 kHz
56 * @li Bit-rates from 6 kb/s to 510 kb/s
57 * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
58 * @li Audio bandwidth from narrowband to full-band
59 * @li Support for speech and music
60 * @li Support for mono and stereo
61 * @li Support for multichannel (up to 255 channels)
62 * @li Frame sizes from 2.5 ms to 60 ms
63 * @li Good loss robustness and packet loss concealment (PLC)
64 * @li Floating point and fixed-point implementation
65 *
66 * Documentation sections:
67 * @li @ref opus_encoder
68 * @li @ref opus_decoder
69 * @li @ref opus_repacketizer
70 * @li @ref opus_multistream
71 * @li @ref opus_libinfo
72 * @li @ref opus_custom
73 */
74
75/** @defgroup opus_encoder Opus Encoder
76 * @{
77 *
78 * @brief This page describes the process and functions used to encode Opus.
79 *
80 * Since Opus is a stateful codec, the encoding process starts with creating an encoder
81 * state. This can be done with:
82 *
83 * @code
84 * int error;
85 * OpusEncoder *enc;
86 * enc = opus_encoder_create(Fs, channels, application, &error);
87 * @endcode
88 *
89 * From this point, @c enc can be used for encoding an audio stream. An encoder state
90 * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
91 * state @b must @b not be re-initialized for each frame.
92 *
93 * While opus_encoder_create() allocates memory for the state, it's also possible
94 * to initialize pre-allocated memory:
95 *
96 * @code
97 * int size;
98 * int error;
99 * OpusEncoder *enc;
100 * size = opus_encoder_get_size(channels);
101 * enc = malloc(size);
102 * error = opus_encoder_init(enc, Fs, channels, application);
103 * @endcode
104 *
105 * where opus_encoder_get_size() returns the required size for the encoder state. Note that
106 * future versions of this code may change the size, so no assumptions should be made about it.
107 *
108 * The encoder state is always continuous in memory and only a shallow copy is sufficient
109 * to copy it (e.g. memcpy())
110 *
111 * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
112 * interface. All these settings already default to the recommended value, so they should
113 * only be changed when necessary. The most common settings one may want to change are:
114 *
115 * @code
116 * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
117 * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
118 * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
119 * @endcode
120 *
121 * where
122 *
123 * @arg bitrate is in bits per second (b/s)
124 * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
125 * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
126 *
127 * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
128 *
129 * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
130 * @code
131 * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
132 * @endcode
133 *
134 * where
135 * <ul>
136 * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
137 * <li>frame_size is the duration of the frame in samples (per channel)</li>
138 * <li>packet is the byte array to which the compressed data is written</li>
139 * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
140 * Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
141 * </ul>
142 *
143 * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
144 * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
145 * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
146 *
147 * Once the encoder state if no longer needed, it can be destroyed with
148 *
149 * @code
150 * opus_encoder_destroy(enc);
151 * @endcode
152 *
153 * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
154 * then no action is required aside from potentially freeing the memory that was manually
155 * allocated for it (calling free(enc) for the example above)
156 *
157 */
158
159/** Opus encoder state.
160 * This contains the complete state of an Opus encoder.
161 * It is position independent and can be freely copied.
162 * @see opus_encoder_create,opus_encoder_init
163 */
164typedef struct OpusEncoder OpusEncoder;
165
166/** Gets the size of an <code>OpusEncoder</code> structure.
167 * @param[in] channels <tt>int</tt>: Number of channels.
168 * This must be 1 or 2.
169 * @returns The size in bytes.
170 */
171OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
172
173/**
174 */
175
176/** Allocates and initializes an encoder state.
177 * There are three coding modes:
178 *
179 * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
180 * signals. It enhances the input signal by high-pass filtering and
181 * emphasizing formants and harmonics. Optionally it includes in-band
182 * forward error correction to protect against packet loss. Use this
183 * mode for typical VoIP applications. Because of the enhancement,
184 * even at high bitrates the output may sound different from the input.
185 *
186 * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
187 * non-voice signals like music. Use this mode for music and mixed
188 * (music/voice) content, broadcast, and applications requiring less
189 * than 15 ms of coding delay.
190 *
191 * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
192 * disables the speech-optimized mode in exchange for slightly reduced delay.
193 * This mode can only be set on an newly initialized or freshly reset encoder
194 * because it changes the codec delay.
195 *
196 * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
197 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
198 * This must be one of 8000, 12000, 16000,
199 * 24000, or 48000.
200 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
201 * @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
202 * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
203 * @note Regardless of the sampling rate and number channels selected, the Opus encoder
204 * can switch to a lower audio bandwidth or number of channels if the bitrate
205 * selected is too low. This also means that it is safe to always use 48 kHz stereo input
206 * and let the encoder optimize the encoding.
207 */
208OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
209 opus_int32 Fs,
210 int channels,
211 int application,
212 int *error
213);
214
215/** Initializes a previously allocated encoder state
216 * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
217 * This is intended for applications which use their own allocator instead of malloc.
218 * @see opus_encoder_create(),opus_encoder_get_size()
219 * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
220 * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
221 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
222 * This must be one of 8000, 12000, 16000,
223 * 24000, or 48000.
224 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
225 * @param [in] application <tt>int</tt>: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY)
226 * @retval #OPUS_OK Success or @ref opus_errorcodes
227 */
228OPUS_EXPORT int opus_encoder_init(
229 OpusEncoder *st,
230 opus_int32 Fs,
231 int channels,
232 int application
233) OPUS_ARG_NONNULL(1);
234
235/** Encodes an Opus frame.
236 * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
237 * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
238 * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
239 * input signal.
240 * This must be an Opus frame size for
241 * the encoder's sampling rate.
242 * For example, at 48 kHz the permitted
243 * values are 120, 240, 480, 960, 1920,
244 * and 2880.
245 * Passing in a duration of less than
246 * 10 ms (480 samples at 48 kHz) will
247 * prevent the encoder from using the LPC
248 * or hybrid modes.
249 * @param [out] data <tt>unsigned char*</tt>: Output payload.
250 * This must contain storage for at
251 * least \a max_data_bytes.
252 * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
253 * memory for the output
254 * payload. This may be
255 * used to impose an upper limit on
256 * the instant bitrate, but should
257 * not be used as the only bitrate
258 * control. Use #OPUS_SET_BITRATE to
259 * control the bitrate.
260 * @returns The length of the encoded packet (in bytes) on success or a
261 * negative error code (see @ref opus_errorcodes) on failure.
262 */
263OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
264 OpusEncoder *st,
265 const opus_int16 *pcm,
266 int frame_size,
267 unsigned char *data,
268 opus_int32 max_data_bytes
269) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
270
271/** Encodes an Opus frame from floating point input.
272 * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
273 * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
274 * Samples with a range beyond +/-1.0 are supported but will
275 * be clipped by decoders using the integer API and should
276 * only be used if it is known that the far end supports
277 * extended dynamic range.
278 * length is frame_size*channels*sizeof(float)
279 * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
280 * input signal.
281 * This must be an Opus frame size for
282 * the encoder's sampling rate.
283 * For example, at 48 kHz the permitted
284 * values are 120, 240, 480, 960, 1920,
285 * and 2880.
286 * Passing in a duration of less than
287 * 10 ms (480 samples at 48 kHz) will
288 * prevent the encoder from using the LPC
289 * or hybrid modes.
290 * @param [out] data <tt>unsigned char*</tt>: Output payload.
291 * This must contain storage for at
292 * least \a max_data_bytes.
293 * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
294 * memory for the output
295 * payload. This may be
296 * used to impose an upper limit on
297 * the instant bitrate, but should
298 * not be used as the only bitrate
299 * control. Use #OPUS_SET_BITRATE to
300 * control the bitrate.
301 * @returns The length of the encoded packet (in bytes) on success or a
302 * negative error code (see @ref opus_errorcodes) on failure.
303 */
304OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
305 OpusEncoder *st,
306 const float *pcm,
307 int frame_size,
308 unsigned char *data,
309 opus_int32 max_data_bytes
310) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
311
312/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
313 * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
314 */
315OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
316
317/** Perform a CTL function on an Opus encoder.
318 *
319 * Generally the request and subsequent arguments are generated
320 * by a convenience macro.
321 * @param st <tt>OpusEncoder*</tt>: Encoder state.
322 * @param request This and all remaining parameters should be replaced by one
323 * of the convenience macros in @ref opus_genericctls or
324 * @ref opus_encoderctls.
325 * @see opus_genericctls
326 * @see opus_encoderctls
327 */
328OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
329/**@}*/
330
331/** @defgroup opus_decoder Opus Decoder
332 * @{
333 *
334 * @brief This page describes the process and functions used to decode Opus.
335 *
336 * The decoding process also starts with creating a decoder
337 * state. This can be done with:
338 * @code
339 * int error;
340 * OpusDecoder *dec;
341 * dec = opus_decoder_create(Fs, channels, &error);
342 * @endcode
343 * where
344 * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
345 * @li channels is the number of channels (1 or 2)
346 * @li error will hold the error code in case of failure (or #OPUS_OK on success)
347 * @li the return value is a newly created decoder state to be used for decoding
348 *
349 * While opus_decoder_create() allocates memory for the state, it's also possible
350 * to initialize pre-allocated memory:
351 * @code
352 * int size;
353 * int error;
354 * OpusDecoder *dec;
355 * size = opus_decoder_get_size(channels);
356 * dec = malloc(size);
357 * error = opus_decoder_init(dec, Fs, channels);
358 * @endcode
359 * where opus_decoder_get_size() returns the required size for the decoder state. Note that
360 * future versions of this code may change the size, so no assumptions should be made about it.
361 *
362 * The decoder state is always continuous in memory and only a shallow copy is sufficient
363 * to copy it (e.g. memcpy())
364 *
365 * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
366 * @code
367 * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
368 * @endcode
369 * where
370 *
371 * @li packet is the byte array containing the compressed data
372 * @li len is the exact number of bytes contained in the packet
373 * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
374 * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
375 *
376 * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
377 * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
378 * buffer is too small to hold the decoded audio.
379 *
380 * Opus is a stateful codec with overlapping blocks and as a result Opus
381 * packets are not coded independently of each other. Packets must be
382 * passed into the decoder serially and in the correct order for a correct
383 * decode. Lost packets can be replaced with loss concealment by calling
384 * the decoder with a null pointer and zero length for the missing packet.
385 *
386 * A single codec state may only be accessed from a single thread at
387 * a time and any required locking must be performed by the caller. Separate
388 * streams must be decoded with separate decoder states and can be decoded
389 * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
390 * defined.
391 *
392 */
393
394/** Opus decoder state.
395 * This contains the complete state of an Opus decoder.
396 * It is position independent and can be freely copied.
397 * @see opus_decoder_create,opus_decoder_init
398 */
399typedef struct OpusDecoder OpusDecoder;
400
401/** Opus DRED decoder.
402 * This contains the complete state of an Opus DRED decoder.
403 * It is position independent and can be freely copied.
404 * @see opus_dred_decoder_create,opus_dred_decoder_init
405 */
406typedef struct OpusDREDDecoder OpusDREDDecoder;
407
408
409/** Opus DRED state.
410 * This contains the complete state of an Opus DRED packet.
411 * It is position independent and can be freely copied.
412 * @see opus_dred_create,opus_dred_init
413 */
414typedef struct OpusDRED OpusDRED;
415
416/** Gets the size of an <code>OpusDecoder</code> structure.
417 * @param [in] channels <tt>int</tt>: Number of channels.
418 * This must be 1 or 2.
419 * @returns The size in bytes.
420 */
421OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
422
423/** Allocates and initializes a decoder state.
424 * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
425 * This must be one of 8000, 12000, 16000,
426 * 24000, or 48000.
427 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
428 * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
429 *
430 * Internally Opus stores data at 48000 Hz, so that should be the default
431 * value for Fs. However, the decoder can efficiently decode to buffers
432 * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
433 * data at the full sample rate, or knows the compressed data doesn't
434 * use the full frequency range, it can request decoding at a reduced
435 * rate. Likewise, the decoder is capable of filling in either mono or
436 * interleaved stereo pcm buffers, at the caller's request.
437 */
438OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
439 opus_int32 Fs,
440 int channels,
441 int *error
442);
443
444/** Initializes a previously allocated decoder state.
445 * The state must be at least the size returned by opus_decoder_get_size().
446 * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
447 * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
448 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
449 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
450 * This must be one of 8000, 12000, 16000,
451 * 24000, or 48000.
452 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
453 * @retval #OPUS_OK Success or @ref opus_errorcodes
454 */
455OPUS_EXPORT int opus_decoder_init(
456 OpusDecoder *st,
457 opus_int32 Fs,
458 int channels
459) OPUS_ARG_NONNULL(1);
460
461/** Decode an Opus packet.
462 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
463 * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
464 * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
465 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
466 * is frame_size*channels*sizeof(opus_int16)
467 * @param [in] frame_size Number of samples per channel of available space in \a pcm.
468 * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
469 * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
470 * then frame_size needs to be exactly the duration of audio that is missing, otherwise the
471 * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
472 * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
473 * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
474 * decoded. If no such data is available, the frame is decoded as if it were lost.
475 * @returns Number of decoded samples or @ref opus_errorcodes
476 */
477OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
478 OpusDecoder *st,
479 const unsigned char *data,
480 opus_int32 len,
481 opus_int16 *pcm,
482 int frame_size,
483 int decode_fec
484) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
485
486/** Decode an Opus packet with floating point output.
487 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
488 * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
489 * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
490 * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
491 * is frame_size*channels*sizeof(float)
492 * @param [in] frame_size Number of samples per channel of available space in \a pcm.
493 * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
494 * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
495 * then frame_size needs to be exactly the duration of audio that is missing, otherwise the
496 * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
497 * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
498 * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
499 * decoded. If no such data is available the frame is decoded as if it were lost.
500 * @returns Number of decoded samples or @ref opus_errorcodes
501 */
502OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
503 OpusDecoder *st,
504 const unsigned char *data,
505 opus_int32 len,
506 float *pcm,
507 int frame_size,
508 int decode_fec
509) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
510
511/** Perform a CTL function on an Opus decoder.
512 *
513 * Generally the request and subsequent arguments are generated
514 * by a convenience macro.
515 * @param st <tt>OpusDecoder*</tt>: Decoder state.
516 * @param request This and all remaining parameters should be replaced by one
517 * of the convenience macros in @ref opus_genericctls or
518 * @ref opus_decoderctls.
519 * @see opus_genericctls
520 * @see opus_decoderctls
521 */
522OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
523
524/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
525 * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
526 */
527OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
528
529/** Gets the size of an <code>OpusDREDDecoder</code> structure.
530 * @returns The size in bytes.
531 */
532OPUS_EXPORT int opus_dred_decoder_get_size(void);
533
534/** Allocates and initializes an OpusDREDDecoder state.
535 * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
536 */
537OPUS_EXPORT OpusDREDDecoder *opus_dred_decoder_create(int *error);
538
539/** Initializes an <code>OpusDREDDecoder</code> state.
540 * @param[in] dec <tt>OpusDREDDecoder*</tt>: State to be initialized.
541 */
542OPUS_EXPORT int opus_dred_decoder_init(OpusDREDDecoder *dec);
543
544/** Frees an <code>OpusDREDDecoder</code> allocated by opus_dred_decoder_create().
545 * @param[in] dec <tt>OpusDREDDecoder*</tt>: State to be freed.
546 */
547OPUS_EXPORT void opus_dred_decoder_destroy(OpusDREDDecoder *dec);
548
549/** Perform a CTL function on an Opus DRED decoder.
550 *
551 * Generally the request and subsequent arguments are generated
552 * by a convenience macro.
553 * @param dred_dec <tt>OpusDREDDecoder*</tt>: DRED Decoder state.
554 * @param request This and all remaining parameters should be replaced by one
555 * of the convenience macros in @ref opus_genericctls or
556 * @ref opus_decoderctls.
557 * @see opus_genericctls
558 * @see opus_decoderctls
559 */
560OPUS_EXPORT int opus_dred_decoder_ctl(OpusDREDDecoder *dred_dec, int request, ...);
561
562/** Gets the size of an <code>OpusDRED</code> structure.
563 * @returns The size in bytes.
564 */
565OPUS_EXPORT int opus_dred_get_size(void);
566
567/** Allocates and initializes a DRED state.
568 * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
569 */
570OPUS_EXPORT OpusDRED *opus_dred_alloc(int *error);
571
572/** Frees an <code>OpusDRED</code> allocated by opus_dred_create().
573 * @param[in] dec <tt>OpusDRED*</tt>: State to be freed.
574 */
575OPUS_EXPORT void opus_dred_free(OpusDRED *dec);
576
577/** Decode an Opus DRED packet.
578 * @param [in] dred_dec <tt>OpusDRED*</tt>: DRED Decoder state
579 * @param [in] dred <tt>OpusDRED*</tt>: DRED state
580 * @param [in] data <tt>char*</tt>: Input payload
581 * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
582 * @param [in] max_dred_samples <tt>opus_int32</tt>: Maximum number of DRED samples that may be needed (if available in the packet).
583 * @param [in] sampling_rate <tt>opus_int32</tt>: Sampling rate used for max_dred_samples argument. Needs not match the actual sampling rate of the decoder.
584 * @param [out] dred_end <tt>opus_int32*</tt>: Number of non-encoded (silence) samples between the DRED timestamp and the last DRED sample.
585 * @param [in] defer_processing <tt>int</tt>: Flag (0 or 1). If set to one, the CPU-intensive part of the DRED decoding is deferred until opus_dred_process() is called.
586 * @returns Offset (positive) of the first decoded DRED samples, zero if no DRED is present, or @ref opus_errorcodes
587 */
588OPUS_EXPORT int opus_dred_parse(OpusDREDDecoder *dred_dec, OpusDRED *dred, const unsigned char *data, opus_int32 len, opus_int32 max_dred_samples, opus_int32 sampling_rate, int *dred_end, int defer_processing) OPUS_ARG_NONNULL(1);
589
590/** Finish decoding an Opus DRED packet. The function only needs to be called if opus_dred_parse() was called with defer_processing=1.
591 * The source and destination will often be the same DRED state.
592 * @param [in] dred_dec <tt>OpusDRED*</tt>: DRED Decoder state
593 * @param [in] src <tt>OpusDRED*</tt>: Source DRED state to start the processing from.
594 * @param [out] dst <tt>OpusDRED*</tt>: Destination DRED state to store the updated state after processing.
595 * @returns @ref opus_errorcodes
596 */
597OPUS_EXPORT int opus_dred_process(OpusDREDDecoder *dred_dec, const OpusDRED *src, OpusDRED *dst);
598
599/** Decode audio from an Opus DRED packet with floating point output.
600 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
601 * @param [in] dred <tt>OpusDRED*</tt>: DRED state
602 * @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
603 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
604 * is frame_size*channels*sizeof(opus_int16)
605 * @param [in] frame_size Number of samples per channel to decode in \a pcm.
606 * frame_size <b>must</b> be a multiple of 2.5 ms.
607 * @returns Number of decoded samples or @ref opus_errorcodes
608 */
609OPUS_EXPORT int opus_decoder_dred_decode(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, opus_int16 *pcm, opus_int32 frame_size);
610
611/** Decode audio from an Opus DRED packet with floating point output.
612 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
613 * @param [in] dred <tt>OpusDRED*</tt>: DRED state
614 * @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
615 * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
616 * is frame_size*channels*sizeof(float)
617 * @param [in] frame_size Number of samples per channel to decode in \a pcm.
618 * frame_size <b>must</b> be a multiple of 2.5 ms.
619 * @returns Number of decoded samples or @ref opus_errorcodes
620 */
621OPUS_EXPORT int opus_decoder_dred_decode_float(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, float *pcm, opus_int32 frame_size);
622
623
624/** Parse an opus packet into one or more frames.
625 * Opus_decode will perform this operation internally so most applications do
626 * not need to use this function.
627 * This function does not copy the frames, the returned pointers are pointers into
628 * the input packet.
629 * @param [in] data <tt>char*</tt>: Opus packet to be parsed
630 * @param [in] len <tt>opus_int32</tt>: size of data
631 * @param [out] out_toc <tt>char*</tt>: TOC pointer
632 * @param [out] frames <tt>char*[48]</tt> encapsulated frames
633 * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
634 * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
635 * @returns number of frames
636 */
637OPUS_EXPORT int opus_packet_parse(
638 const unsigned char *data,
639 opus_int32 len,
640 unsigned char *out_toc,
641 const unsigned char *frames[48],
642 opus_int16 size[48],
643 int *payload_offset
644) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
645
646/** Gets the bandwidth of an Opus packet.
647 * @param [in] data <tt>char*</tt>: Opus packet
648 * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
649 * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
650 * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
651 * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
652 * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
653 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
654 */
655OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
656
657/** Gets the number of samples per frame from an Opus packet.
658 * @param [in] data <tt>char*</tt>: Opus packet.
659 * This must contain at least one byte of
660 * data.
661 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
662 * This must be a multiple of 400, or
663 * inaccurate results will be returned.
664 * @returns Number of samples per frame.
665 */
666OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
667
668/** Gets the number of channels from an Opus packet.
669 * @param [in] data <tt>char*</tt>: Opus packet
670 * @returns Number of channels
671 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
672 */
673OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
674
675/** Gets the number of frames in an Opus packet.
676 * @param [in] packet <tt>char*</tt>: Opus packet
677 * @param [in] len <tt>opus_int32</tt>: Length of packet
678 * @returns Number of frames
679 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
680 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
681 */
682OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
683
684/** Gets the number of samples of an Opus packet.
685 * @param [in] packet <tt>char*</tt>: Opus packet
686 * @param [in] len <tt>opus_int32</tt>: Length of packet
687 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
688 * This must be a multiple of 400, or
689 * inaccurate results will be returned.
690 * @returns Number of samples
691 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
692 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
693 */
694OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
695
696/** Checks whether an Opus packet has LBRR.
697 * @param [in] packet <tt>char*</tt>: Opus packet
698 * @param [in] len <tt>opus_int32</tt>: Length of packet
699 * @returns 1 is LBRR is present, 0 otherwise
700 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
701 */
702OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_has_lbrr(const unsigned char packet[], opus_int32 len);
703
704/** Gets the number of samples of an Opus packet.
705 * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
706 * @param [in] packet <tt>char*</tt>: Opus packet
707 * @param [in] len <tt>opus_int32</tt>: Length of packet
708 * @returns Number of samples
709 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
710 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
711 */
712OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
713
714/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
715 * the signal is already in that range, nothing is done. If there are values
716 * outside of [-1,1], then the signal is clipped as smoothly as possible to
717 * both fit in the range and avoid creating excessive distortion in the
718 * process.
719 * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
720 * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
721 * @param [in] channels <tt>int</tt>: Number of channels
722 * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
723 */
724OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
725
726
727/**@}*/
728
729/** @defgroup opus_repacketizer Repacketizer
730 * @{
731 *
732 * The repacketizer can be used to merge multiple Opus packets into a single
733 * packet or alternatively to split Opus packets that have previously been
734 * merged. Splitting valid Opus packets is always guaranteed to succeed,
735 * whereas merging valid packets only succeeds if all frames have the same
736 * mode, bandwidth, and frame size, and when the total duration of the merged
737 * packet is no more than 120 ms. The 120 ms limit comes from the
738 * specification and limits decoder memory requirements at a point where
739 * framing overhead becomes negligible.
740 *
741 * The repacketizer currently only operates on elementary Opus
742 * streams. It will not manipualte multistream packets successfully, except in
743 * the degenerate case where they consist of data from a single stream.
744 *
745 * The repacketizing process starts with creating a repacketizer state, either
746 * by calling opus_repacketizer_create() or by allocating the memory yourself,
747 * e.g.,
748 * @code
749 * OpusRepacketizer *rp;
750 * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
751 * if (rp != NULL)
752 * opus_repacketizer_init(rp);
753 * @endcode
754 *
755 * Then the application should submit packets with opus_repacketizer_cat(),
756 * extract new packets with opus_repacketizer_out() or
757 * opus_repacketizer_out_range(), and then reset the state for the next set of
758 * input packets via opus_repacketizer_init().
759 *
760 * For example, to split a sequence of packets into individual frames:
761 * @code
762 * unsigned char *data;
763 * int len;
764 * while (get_next_packet(&data, &len))
765 * {
766 * unsigned char out[1276];
767 * opus_int32 out_len;
768 * int nb_frames;
769 * int err;
770 * int i;
771 * err = opus_repacketizer_cat(rp, data, len);
772 * if (err != OPUS_OK)
773 * {
774 * release_packet(data);
775 * return err;
776 * }
777 * nb_frames = opus_repacketizer_get_nb_frames(rp);
778 * for (i = 0; i < nb_frames; i++)
779 * {
780 * out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
781 * if (out_len < 0)
782 * {
783 * release_packet(data);
784 * return (int)out_len;
785 * }
786 * output_next_packet(out, out_len);
787 * }
788 * opus_repacketizer_init(rp);
789 * release_packet(data);
790 * }
791 * @endcode
792 *
793 * Alternatively, to combine a sequence of frames into packets that each
794 * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
795 * @code
796 * // The maximum number of packets with duration TARGET_DURATION_MS occurs
797 * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
798 * // packets.
799 * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
800 * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
801 * int nb_packets;
802 * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
803 * opus_int32 out_len;
804 * int prev_toc;
805 * nb_packets = 0;
806 * while (get_next_packet(data+nb_packets, len+nb_packets))
807 * {
808 * int nb_frames;
809 * int err;
810 * nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
811 * if (nb_frames < 1)
812 * {
813 * release_packets(data, nb_packets+1);
814 * return nb_frames;
815 * }
816 * nb_frames += opus_repacketizer_get_nb_frames(rp);
817 * // If adding the next packet would exceed our target, or it has an
818 * // incompatible TOC sequence, output the packets we already have before
819 * // submitting it.
820 * // N.B., The nb_packets > 0 check ensures we've submitted at least one
821 * // packet since the last call to opus_repacketizer_init(). Otherwise a
822 * // single packet longer than TARGET_DURATION_MS would cause us to try to
823 * // output an (invalid) empty packet. It also ensures that prev_toc has
824 * // been set to a valid value. Additionally, len[nb_packets] > 0 is
825 * // guaranteed by the call to opus_packet_get_nb_frames() above, so the
826 * // reference to data[nb_packets][0] should be valid.
827 * if (nb_packets > 0 && (
828 * ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
829 * opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
830 * TARGET_DURATION_MS*48))
831 * {
832 * out_len = opus_repacketizer_out(rp, out, sizeof(out));
833 * if (out_len < 0)
834 * {
835 * release_packets(data, nb_packets+1);
836 * return (int)out_len;
837 * }
838 * output_next_packet(out, out_len);
839 * opus_repacketizer_init(rp);
840 * release_packets(data, nb_packets);
841 * data[0] = data[nb_packets];
842 * len[0] = len[nb_packets];
843 * nb_packets = 0;
844 * }
845 * err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
846 * if (err != OPUS_OK)
847 * {
848 * release_packets(data, nb_packets+1);
849 * return err;
850 * }
851 * prev_toc = data[nb_packets][0];
852 * nb_packets++;
853 * }
854 * // Output the final, partial packet.
855 * if (nb_packets > 0)
856 * {
857 * out_len = opus_repacketizer_out(rp, out, sizeof(out));
858 * release_packets(data, nb_packets);
859 * if (out_len < 0)
860 * return (int)out_len;
861 * output_next_packet(out, out_len);
862 * }
863 * @endcode
864 *
865 * An alternate way of merging packets is to simply call opus_repacketizer_cat()
866 * unconditionally until it fails. At that point, the merged packet can be
867 * obtained with opus_repacketizer_out() and the input packet for which
868 * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
869 * repacketizer state.
870 */
871
872typedef struct OpusRepacketizer OpusRepacketizer;
873
874/** Gets the size of an <code>OpusRepacketizer</code> structure.
875 * @returns The size in bytes.
876 */
877OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
878
879/** (Re)initializes a previously allocated repacketizer state.
880 * The state must be at least the size returned by opus_repacketizer_get_size().
881 * This can be used for applications which use their own allocator instead of
882 * malloc().
883 * It must also be called to reset the queue of packets waiting to be
884 * repacketized, which is necessary if the maximum packet duration of 120 ms
885 * is reached or if you wish to submit packets with a different Opus
886 * configuration (coding mode, audio bandwidth, frame size, or channel count).
887 * Failure to do so will prevent a new packet from being added with
888 * opus_repacketizer_cat().
889 * @see opus_repacketizer_create
890 * @see opus_repacketizer_get_size
891 * @see opus_repacketizer_cat
892 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
893 * (re)initialize.
894 * @returns A pointer to the same repacketizer state that was passed in.
895 */
896OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
897
898/** Allocates memory and initializes the new repacketizer with
899 * opus_repacketizer_init().
900 */
901OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
902
903/** Frees an <code>OpusRepacketizer</code> allocated by
904 * opus_repacketizer_create().
905 * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
906 */
907OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
908
909/** Add a packet to the current repacketizer state.
910 * This packet must match the configuration of any packets already submitted
911 * for repacketization since the last call to opus_repacketizer_init().
912 * This means that it must have the same coding mode, audio bandwidth, frame
913 * size, and channel count.
914 * This can be checked in advance by examining the top 6 bits of the first
915 * byte of the packet, and ensuring they match the top 6 bits of the first
916 * byte of any previously submitted packet.
917 * The total duration of audio in the repacketizer state also must not exceed
918 * 120 ms, the maximum duration of a single packet, after adding this packet.
919 *
920 * The contents of the current repacketizer state can be extracted into new
921 * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
922 *
923 * In order to add a packet with a different configuration or to add more
924 * audio beyond 120 ms, you must clear the repacketizer state by calling
925 * opus_repacketizer_init().
926 * If a packet is too large to add to the current repacketizer state, no part
927 * of it is added, even if it contains multiple frames, some of which might
928 * fit.
929 * If you wish to be able to add parts of such packets, you should first use
930 * another repacketizer to split the packet into pieces and add them
931 * individually.
932 * @see opus_repacketizer_out_range
933 * @see opus_repacketizer_out
934 * @see opus_repacketizer_init
935 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
936 * add the packet.
937 * @param[in] data <tt>const unsigned char*</tt>: The packet data.
938 * The application must ensure
939 * this pointer remains valid
940 * until the next call to
941 * opus_repacketizer_init() or
942 * opus_repacketizer_destroy().
943 * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
944 * @returns An error code indicating whether or not the operation succeeded.
945 * @retval #OPUS_OK The packet's contents have been added to the repacketizer
946 * state.
947 * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
948 * the packet's TOC sequence was not compatible
949 * with previously submitted packets (because
950 * the coding mode, audio bandwidth, frame size,
951 * or channel count did not match), or adding
952 * this packet would increase the total amount of
953 * audio stored in the repacketizer state to more
954 * than 120 ms.
955 */
956OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
957
958
959/** Construct a new packet from data previously submitted to the repacketizer
960 * state via opus_repacketizer_cat().
961 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
962 * construct the new packet.
963 * @param begin <tt>int</tt>: The index of the first frame in the current
964 * repacketizer state to include in the output.
965 * @param end <tt>int</tt>: One past the index of the last frame in the
966 * current repacketizer state to include in the
967 * output.
968 * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
969 * store the output packet.
970 * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
971 * the output buffer. In order to guarantee
972 * success, this should be at least
973 * <code>1276</code> for a single frame,
974 * or for multiple frames,
975 * <code>1277*(end-begin)</code>.
976 * However, <code>1*(end-begin)</code> plus
977 * the size of all packet data submitted to
978 * the repacketizer since the last call to
979 * opus_repacketizer_init() or
980 * opus_repacketizer_create() is also
981 * sufficient, and possibly much smaller.
982 * @returns The total size of the output packet on success, or an error code
983 * on failure.
984 * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
985 * frames (begin < 0, begin >= end, or end >
986 * opus_repacketizer_get_nb_frames()).
987 * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
988 * complete output packet.
989 */
990OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
991
992/** Return the total number of frames contained in packet data submitted to
993 * the repacketizer state so far via opus_repacketizer_cat() since the last
994 * call to opus_repacketizer_init() or opus_repacketizer_create().
995 * This defines the valid range of packets that can be extracted with
996 * opus_repacketizer_out_range() or opus_repacketizer_out().
997 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
998 * frames.
999 * @returns The total number of frames contained in the packet data submitted
1000 * to the repacketizer state.
1001 */
1002OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
1003
1004/** Construct a new packet from data previously submitted to the repacketizer
1005 * state via opus_repacketizer_cat().
1006 * This is a convenience routine that returns all the data submitted so far
1007 * in a single packet.
1008 * It is equivalent to calling
1009 * @code
1010 * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
1011 * data, maxlen)
1012 * @endcode
1013 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
1014 * construct the new packet.
1015 * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
1016 * store the output packet.
1017 * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
1018 * the output buffer. In order to guarantee
1019 * success, this should be at least
1020 * <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
1021 * However,
1022 * <code>1*opus_repacketizer_get_nb_frames(rp)</code>
1023 * plus the size of all packet data
1024 * submitted to the repacketizer since the
1025 * last call to opus_repacketizer_init() or
1026 * opus_repacketizer_create() is also
1027 * sufficient, and possibly much smaller.
1028 * @returns The total size of the output packet on success, or an error code
1029 * on failure.
1030 * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
1031 * complete output packet.
1032 */
1033OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
1034
1035/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
1036 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
1037 * packet to pad.
1038 * @param len <tt>opus_int32</tt>: The size of the packet.
1039 * This must be at least 1.
1040 * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
1041 * This must be at least as large as len.
1042 * @returns an error code
1043 * @retval #OPUS_OK \a on success.
1044 * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
1045 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
1046 */
1047OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
1048
1049/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
1050 * minimize space usage.
1051 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
1052 * packet to strip.
1053 * @param len <tt>opus_int32</tt>: The size of the packet.
1054 * This must be at least 1.
1055 * @returns The new size of the output packet on success, or an error code
1056 * on failure.
1057 * @retval #OPUS_BAD_ARG \a len was less than 1.
1058 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
1059 */
1060OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
1061
1062/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
1063 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
1064 * packet to pad.
1065 * @param len <tt>opus_int32</tt>: The size of the packet.
1066 * This must be at least 1.
1067 * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
1068 * This must be at least 1.
1069 * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
1070 * This must be at least as large as len.
1071 * @returns an error code
1072 * @retval #OPUS_OK \a on success.
1073 * @retval #OPUS_BAD_ARG \a len was less than 1.
1074 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
1075 */
1076OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
1077
1078/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
1079 * minimize space usage.
1080 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
1081 * packet to strip.
1082 * @param len <tt>opus_int32</tt>: The size of the packet.
1083 * This must be at least 1.
1084 * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
1085 * This must be at least 1.
1086 * @returns The new size of the output packet on success, or an error code
1087 * on failure.
1088 * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
1089 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
1090 */
1091OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
1092
1093/**@}*/
1094
1095#ifdef __cplusplus
1096}
1097#endif
1098
1099#endif /* OPUS_H */
1100