path: root/rcaudio/opus.h (plain)
blob: e029f710037cd1a9f5cae8cfa5818b9fc3a74913

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