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media/libopus/include/opus.h

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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
40 extern "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 assuptions 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 1 byte, 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 */
164 typedef 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 */
171 OPUS_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 (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@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 */
208 OPUS_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 (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
226 * @retval #OPUS_OK Success or @ref opus_errorcodes
227 */
228 OPUS_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 */
263 OPUS_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 */
304 OPUS_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 */
315 OPUS_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 */
328 OPUS_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 assuptions 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 */
399 typedef struct OpusDecoder OpusDecoder;
400
401 /** Gets the size of an <code>OpusDecoder</code> structure.
402 * @param [in] channels <tt>int</tt>: Number of channels.
403 * This must be 1 or 2.
404 * @returns The size in bytes.
405 */
406 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
407
408 /** Allocates and initializes a decoder state.
409 * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
410 * This must be one of 8000, 12000, 16000,
411 * 24000, or 48000.
412 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
413 * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
414 *
415 * Internally Opus stores data at 48000 Hz, so that should be the default
416 * value for Fs. However, the decoder can efficiently decode to buffers
417 * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
418 * data at the full sample rate, or knows the compressed data doesn't
419 * use the full frequency range, it can request decoding at a reduced
420 * rate. Likewise, the decoder is capable of filling in either mono or
421 * interleaved stereo pcm buffers, at the caller's request.
422 */
423 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
424 opus_int32 Fs,
425 int channels,
426 int *error
427 );
428
429 /** Initializes a previously allocated decoder state.
430 * The state must be at least the size returned by opus_decoder_get_size().
431 * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
432 * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
433 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
434 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
435 * This must be one of 8000, 12000, 16000,
436 * 24000, or 48000.
437 * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
438 * @retval #OPUS_OK Success or @ref opus_errorcodes
439 */
440 OPUS_EXPORT int opus_decoder_init(
441 OpusDecoder *st,
442 opus_int32 Fs,
443 int channels
444 ) OPUS_ARG_NONNULL(1);
445
446 /** Decode an Opus packet.
447 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
448 * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
449 * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
450 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
451 * is frame_size*channels*sizeof(opus_int16)
452 * @param [in] frame_size Number of samples per channel of available space in \a pcm.
453 * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
454 * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
455 * then frame_size needs to be exactly the duration of audio that is missing, otherwise the
456 * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
457 * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
458 * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
459 * decoded. If no such data is available, the frame is decoded as if it were lost.
460 * @returns Number of decoded samples or @ref opus_errorcodes
461 */
462 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
463 OpusDecoder *st,
464 const unsigned char *data,
465 opus_int32 len,
466 opus_int16 *pcm,
467 int frame_size,
468 int decode_fec
469 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
470
471 /** Decode an Opus packet with floating point output.
472 * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
473 * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
474 * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
475 * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
476 * is frame_size*channels*sizeof(float)
477 * @param [in] frame_size Number of samples per channel of available space in \a pcm.
478 * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
479 * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
480 * then frame_size needs to be exactly the duration of audio that is missing, otherwise the
481 * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
482 * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
483 * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
484 * decoded. If no such data is available the frame is decoded as if it were lost.
485 * @returns Number of decoded samples or @ref opus_errorcodes
486 */
487 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
488 OpusDecoder *st,
489 const unsigned char *data,
490 opus_int32 len,
491 float *pcm,
492 int frame_size,
493 int decode_fec
494 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
495
496 /** Perform a CTL function on an Opus decoder.
497 *
498 * Generally the request and subsequent arguments are generated
499 * by a convenience macro.
500 * @param st <tt>OpusDecoder*</tt>: Decoder state.
501 * @param request This and all remaining parameters should be replaced by one
502 * of the convenience macros in @ref opus_genericctls or
503 * @ref opus_decoderctls.
504 * @see opus_genericctls
505 * @see opus_decoderctls
506 */
507 OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
508
509 /** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
510 * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
511 */
512 OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
513
514 /** Parse an opus packet into one or more frames.
515 * Opus_decode will perform this operation internally so most applications do
516 * not need to use this function.
517 * This function does not copy the frames, the returned pointers are pointers into
518 * the input packet.
519 * @param [in] data <tt>char*</tt>: Opus packet to be parsed
520 * @param [in] len <tt>opus_int32</tt>: size of data
521 * @param [out] out_toc <tt>char*</tt>: TOC pointer
522 * @param [out] frames <tt>char*[48]</tt> encapsulated frames
523 * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
524 * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
525 * @returns number of frames
526 */
527 OPUS_EXPORT int opus_packet_parse(
528 const unsigned char *data,
529 opus_int32 len,
530 unsigned char *out_toc,
531 const unsigned char *frames[48],
532 opus_int16 size[48],
533 int *payload_offset
534 ) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
535
536 /** Gets the bandwidth of an Opus packet.
537 * @param [in] data <tt>char*</tt>: Opus packet
538 * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
539 * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
540 * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
541 * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
542 * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
543 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
544 */
545 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
546
547 /** Gets the number of samples per frame from an Opus packet.
548 * @param [in] data <tt>char*</tt>: Opus packet.
549 * This must contain at least one byte of
550 * data.
551 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
552 * This must be a multiple of 400, or
553 * inaccurate results will be returned.
554 * @returns Number of samples per frame.
555 */
556 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
557
558 /** Gets the number of channels from an Opus packet.
559 * @param [in] data <tt>char*</tt>: Opus packet
560 * @returns Number of channels
561 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
562 */
563 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
564
565 /** Gets the number of frames in an Opus packet.
566 * @param [in] packet <tt>char*</tt>: Opus packet
567 * @param [in] len <tt>opus_int32</tt>: Length of packet
568 * @returns Number of frames
569 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
570 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
571 */
572 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
573
574 /** Gets the number of samples of an Opus packet.
575 * @param [in] packet <tt>char*</tt>: Opus packet
576 * @param [in] len <tt>opus_int32</tt>: Length of packet
577 * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
578 * This must be a multiple of 400, or
579 * inaccurate results will be returned.
580 * @returns Number of samples
581 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
582 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
583 */
584 OPUS_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);
585
586 /** Gets the number of samples of an Opus packet.
587 * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
588 * @param [in] packet <tt>char*</tt>: Opus packet
589 * @param [in] len <tt>opus_int32</tt>: Length of packet
590 * @returns Number of samples
591 * @retval OPUS_BAD_ARG Insufficient data was passed to the function
592 * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
593 */
594 OPUS_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);
595
596 /** Applies soft-clipping to bring a float signal within the [-1,1] range. If
597 * the signal is already in that range, nothing is done. If there are values
598 * outside of [-1,1], then the signal is clipped as smoothly as possible to
599 * both fit in the range and avoid creating excessive distortion in the
600 * process.
601 * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
602 * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
603 * @param [in] channels <tt>int</tt>: Number of channels
604 * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
605 */
606 OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
607
608
609 /**@}*/
610
611 /** @defgroup opus_repacketizer Repacketizer
612 * @{
613 *
614 * The repacketizer can be used to merge multiple Opus packets into a single
615 * packet or alternatively to split Opus packets that have previously been
616 * merged. Splitting valid Opus packets is always guaranteed to succeed,
617 * whereas merging valid packets only succeeds if all frames have the same
618 * mode, bandwidth, and frame size, and when the total duration of the merged
619 * packet is no more than 120 ms.
620 * The repacketizer currently only operates on elementary Opus
621 * streams. It will not manipualte multistream packets successfully, except in
622 * the degenerate case where they consist of data from a single stream.
623 *
624 * The repacketizing process starts with creating a repacketizer state, either
625 * by calling opus_repacketizer_create() or by allocating the memory yourself,
626 * e.g.,
627 * @code
628 * OpusRepacketizer *rp;
629 * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
630 * if (rp != NULL)
631 * opus_repacketizer_init(rp);
632 * @endcode
633 *
634 * Then the application should submit packets with opus_repacketizer_cat(),
635 * extract new packets with opus_repacketizer_out() or
636 * opus_repacketizer_out_range(), and then reset the state for the next set of
637 * input packets via opus_repacketizer_init().
638 *
639 * For example, to split a sequence of packets into individual frames:
640 * @code
641 * unsigned char *data;
642 * int len;
643 * while (get_next_packet(&data, &len))
644 * {
645 * unsigned char out[1276];
646 * opus_int32 out_len;
647 * int nb_frames;
648 * int err;
649 * int i;
650 * err = opus_repacketizer_cat(rp, data, len);
651 * if (err != OPUS_OK)
652 * {
653 * release_packet(data);
654 * return err;
655 * }
656 * nb_frames = opus_repacketizer_get_nb_frames(rp);
657 * for (i = 0; i < nb_frames; i++)
658 * {
659 * out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
660 * if (out_len < 0)
661 * {
662 * release_packet(data);
663 * return (int)out_len;
664 * }
665 * output_next_packet(out, out_len);
666 * }
667 * opus_repacketizer_init(rp);
668 * release_packet(data);
669 * }
670 * @endcode
671 *
672 * Alternatively, to combine a sequence of frames into packets that each
673 * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
674 * @code
675 * // The maximum number of packets with duration TARGET_DURATION_MS occurs
676 * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
677 * // packets.
678 * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
679 * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
680 * int nb_packets;
681 * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
682 * opus_int32 out_len;
683 * int prev_toc;
684 * nb_packets = 0;
685 * while (get_next_packet(data+nb_packets, len+nb_packets))
686 * {
687 * int nb_frames;
688 * int err;
689 * nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
690 * if (nb_frames < 1)
691 * {
692 * release_packets(data, nb_packets+1);
693 * return nb_frames;
694 * }
695 * nb_frames += opus_repacketizer_get_nb_frames(rp);
696 * // If adding the next packet would exceed our target, or it has an
697 * // incompatible TOC sequence, output the packets we already have before
698 * // submitting it.
699 * // N.B., The nb_packets > 0 check ensures we've submitted at least one
700 * // packet since the last call to opus_repacketizer_init(). Otherwise a
701 * // single packet longer than TARGET_DURATION_MS would cause us to try to
702 * // output an (invalid) empty packet. It also ensures that prev_toc has
703 * // been set to a valid value. Additionally, len[nb_packets] > 0 is
704 * // guaranteed by the call to opus_packet_get_nb_frames() above, so the
705 * // reference to data[nb_packets][0] should be valid.
706 * if (nb_packets > 0 && (
707 * ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
708 * opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
709 * TARGET_DURATION_MS*48))
710 * {
711 * out_len = opus_repacketizer_out(rp, out, sizeof(out));
712 * if (out_len < 0)
713 * {
714 * release_packets(data, nb_packets+1);
715 * return (int)out_len;
716 * }
717 * output_next_packet(out, out_len);
718 * opus_repacketizer_init(rp);
719 * release_packets(data, nb_packets);
720 * data[0] = data[nb_packets];
721 * len[0] = len[nb_packets];
722 * nb_packets = 0;
723 * }
724 * err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
725 * if (err != OPUS_OK)
726 * {
727 * release_packets(data, nb_packets+1);
728 * return err;
729 * }
730 * prev_toc = data[nb_packets][0];
731 * nb_packets++;
732 * }
733 * // Output the final, partial packet.
734 * if (nb_packets > 0)
735 * {
736 * out_len = opus_repacketizer_out(rp, out, sizeof(out));
737 * release_packets(data, nb_packets);
738 * if (out_len < 0)
739 * return (int)out_len;
740 * output_next_packet(out, out_len);
741 * }
742 * @endcode
743 *
744 * An alternate way of merging packets is to simply call opus_repacketizer_cat()
745 * unconditionally until it fails. At that point, the merged packet can be
746 * obtained with opus_repacketizer_out() and the input packet for which
747 * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
748 * repacketizer state.
749 */
750
751 typedef struct OpusRepacketizer OpusRepacketizer;
752
753 /** Gets the size of an <code>OpusRepacketizer</code> structure.
754 * @returns The size in bytes.
755 */
756 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
757
758 /** (Re)initializes a previously allocated repacketizer state.
759 * The state must be at least the size returned by opus_repacketizer_get_size().
760 * This can be used for applications which use their own allocator instead of
761 * malloc().
762 * It must also be called to reset the queue of packets waiting to be
763 * repacketized, which is necessary if the maximum packet duration of 120 ms
764 * is reached or if you wish to submit packets with a different Opus
765 * configuration (coding mode, audio bandwidth, frame size, or channel count).
766 * Failure to do so will prevent a new packet from being added with
767 * opus_repacketizer_cat().
768 * @see opus_repacketizer_create
769 * @see opus_repacketizer_get_size
770 * @see opus_repacketizer_cat
771 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
772 * (re)initialize.
773 * @returns A pointer to the same repacketizer state that was passed in.
774 */
775 OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
776
777 /** Allocates memory and initializes the new repacketizer with
778 * opus_repacketizer_init().
779 */
780 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
781
782 /** Frees an <code>OpusRepacketizer</code> allocated by
783 * opus_repacketizer_create().
784 * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
785 */
786 OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
787
788 /** Add a packet to the current repacketizer state.
789 * This packet must match the configuration of any packets already submitted
790 * for repacketization since the last call to opus_repacketizer_init().
791 * This means that it must have the same coding mode, audio bandwidth, frame
792 * size, and channel count.
793 * This can be checked in advance by examining the top 6 bits of the first
794 * byte of the packet, and ensuring they match the top 6 bits of the first
795 * byte of any previously submitted packet.
796 * The total duration of audio in the repacketizer state also must not exceed
797 * 120 ms, the maximum duration of a single packet, after adding this packet.
798 *
799 * The contents of the current repacketizer state can be extracted into new
800 * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
801 *
802 * In order to add a packet with a different configuration or to add more
803 * audio beyond 120 ms, you must clear the repacketizer state by calling
804 * opus_repacketizer_init().
805 * If a packet is too large to add to the current repacketizer state, no part
806 * of it is added, even if it contains multiple frames, some of which might
807 * fit.
808 * If you wish to be able to add parts of such packets, you should first use
809 * another repacketizer to split the packet into pieces and add them
810 * individually.
811 * @see opus_repacketizer_out_range
812 * @see opus_repacketizer_out
813 * @see opus_repacketizer_init
814 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
815 * add the packet.
816 * @param[in] data <tt>const unsigned char*</tt>: The packet data.
817 * The application must ensure
818 * this pointer remains valid
819 * until the next call to
820 * opus_repacketizer_init() or
821 * opus_repacketizer_destroy().
822 * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
823 * @returns An error code indicating whether or not the operation succeeded.
824 * @retval #OPUS_OK The packet's contents have been added to the repacketizer
825 * state.
826 * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
827 * the packet's TOC sequence was not compatible
828 * with previously submitted packets (because
829 * the coding mode, audio bandwidth, frame size,
830 * or channel count did not match), or adding
831 * this packet would increase the total amount of
832 * audio stored in the repacketizer state to more
833 * than 120 ms.
834 */
835 OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
836
837
838 /** Construct a new packet from data previously submitted to the repacketizer
839 * state via opus_repacketizer_cat().
840 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
841 * construct the new packet.
842 * @param begin <tt>int</tt>: The index of the first frame in the current
843 * repacketizer state to include in the output.
844 * @param end <tt>int</tt>: One past the index of the last frame in the
845 * current repacketizer state to include in the
846 * output.
847 * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
848 * store the output packet.
849 * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
850 * the output buffer. In order to guarantee
851 * success, this should be at least
852 * <code>1276</code> for a single frame,
853 * or for multiple frames,
854 * <code>1277*(end-begin)</code>.
855 * However, <code>1*(end-begin)</code> plus
856 * the size of all packet data submitted to
857 * the repacketizer since the last call to
858 * opus_repacketizer_init() or
859 * opus_repacketizer_create() is also
860 * sufficient, and possibly much smaller.
861 * @returns The total size of the output packet on success, or an error code
862 * on failure.
863 * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
864 * frames (begin < 0, begin >= end, or end >
865 * opus_repacketizer_get_nb_frames()).
866 * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
867 * complete output packet.
868 */
869 OPUS_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);
870
871 /** Return the total number of frames contained in packet data submitted to
872 * the repacketizer state so far via opus_repacketizer_cat() since the last
873 * call to opus_repacketizer_init() or opus_repacketizer_create().
874 * This defines the valid range of packets that can be extracted with
875 * opus_repacketizer_out_range() or opus_repacketizer_out().
876 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
877 * frames.
878 * @returns The total number of frames contained in the packet data submitted
879 * to the repacketizer state.
880 */
881 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
882
883 /** Construct a new packet from data previously submitted to the repacketizer
884 * state via opus_repacketizer_cat().
885 * This is a convenience routine that returns all the data submitted so far
886 * in a single packet.
887 * It is equivalent to calling
888 * @code
889 * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
890 * data, maxlen)
891 * @endcode
892 * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
893 * construct the new packet.
894 * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
895 * store the output packet.
896 * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
897 * the output buffer. In order to guarantee
898 * success, this should be at least
899 * <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
900 * However,
901 * <code>1*opus_repacketizer_get_nb_frames(rp)</code>
902 * plus the size of all packet data
903 * submitted to the repacketizer since the
904 * last call to opus_repacketizer_init() or
905 * opus_repacketizer_create() is also
906 * sufficient, and possibly much smaller.
907 * @returns The total size of the output packet on success, or an error code
908 * on failure.
909 * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
910 * complete output packet.
911 */
912 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
913
914 /** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
915 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
916 * packet to pad.
917 * @param len <tt>opus_int32</tt>: The size of the packet.
918 * This must be at least 1.
919 * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
920 * This must be at least as large as len.
921 * @returns an error code
922 * @retval #OPUS_OK \a on success.
923 * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
924 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
925 */
926 OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
927
928 /** Remove all padding from a given Opus packet and rewrite the TOC sequence to
929 * minimize space usage.
930 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
931 * packet to strip.
932 * @param len <tt>opus_int32</tt>: The size of the packet.
933 * This must be at least 1.
934 * @returns The new size of the output packet on success, or an error code
935 * on failure.
936 * @retval #OPUS_BAD_ARG \a len was less than 1.
937 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
938 */
939 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
940
941 /** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
942 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
943 * packet to pad.
944 * @param len <tt>opus_int32</tt>: The size of the packet.
945 * This must be at least 1.
946 * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
947 * This must be at least 1.
948 * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
949 * This must be at least as large as len.
950 * @returns an error code
951 * @retval #OPUS_OK \a on success.
952 * @retval #OPUS_BAD_ARG \a len was less than 1.
953 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
954 */
955 OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
956
957 /** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
958 * minimize space usage.
959 * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
960 * packet to strip.
961 * @param len <tt>opus_int32</tt>: The size of the packet.
962 * This must be at least 1.
963 * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
964 * This must be at least 1.
965 * @returns The new size of the output packet on success, or an error code
966 * on failure.
967 * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
968 * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
969 */
970 OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
971
972 /**@}*/
973
974 #ifdef __cplusplus
975 }
976 #endif
977
978 #endif /* OPUS_H */

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