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Diffstat (limited to 'main/codec_builtin.c')
| -rw-r--r-- | main/codec_builtin.c | 845 |
1 files changed, 845 insertions, 0 deletions
diff --git a/main/codec_builtin.c b/main/codec_builtin.c new file mode 100644 index 000000000..594074aa4 --- /dev/null +++ b/main/codec_builtin.c @@ -0,0 +1,845 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2014, Digium, Inc. + * + * Joshua Colp <jcolp@digium.com> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! \file + * + * \brief Built-in supported codecs + * + * \author Joshua Colp <jcolp@digium.com> + */ + +/*** MODULEINFO + <support_level>core</support_level> + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include "asterisk/logger.h" +#include "asterisk/astobj2.h" +#include "asterisk/codec.h" +#include "asterisk/format.h" +#include "asterisk/format_cache.h" +#include "asterisk/frame.h" + +enum frame_type { + TYPE_HIGH, /* 0x0 */ + TYPE_LOW, /* 0x1 */ + TYPE_SILENCE, /* 0x2 */ + TYPE_DONTSEND /* 0x3 */ +}; + +#define TYPE_MASK 0x3 + +static int g723_len(unsigned char buf) +{ + enum frame_type type = buf & TYPE_MASK; + + switch(type) { + case TYPE_DONTSEND: + return 0; + break; + case TYPE_SILENCE: + return 4; + break; + case TYPE_HIGH: + return 24; + break; + case TYPE_LOW: + return 20; + break; + default: + ast_log(LOG_WARNING, "Badly encoded frame (%u)\n", type); + } + return -1; +} + +static int g723_samples(struct ast_frame *frame) +{ + unsigned char *buf = frame->data.ptr; + int pos = 0, samples = 0, res; + + while(pos < frame->datalen) { + res = g723_len(buf[pos]); + if (res <= 0) + break; + samples += 240; + pos += res; + } + + return samples; +} + +static int g723_length(unsigned int samples) +{ + return (samples / 240) * 20; +} + +static struct ast_codec g723 = { + .name = "g723", + .description = "G.723.1", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 30, + .maximum_ms = 300, + .default_ms = 30, + .minimum_bytes = 20, + .samples_count = g723_samples, + .get_length = g723_length, +}; + +static int none_samples(struct ast_frame *frame) +{ + return frame->datalen; +} + +static int none_length(unsigned int samples) { + return samples; +} + +static struct ast_codec none = { + .name = "none", + .description = "<Null> codec", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, /* This must have some sample rate to prevent divide by 0 */ + .minimum_ms = 10, + .maximum_ms = 150, + .default_ms = 20, + .minimum_bytes = 20, + .samples_count = none_samples, + .get_length = none_length, +}; + +static int ulaw_samples(struct ast_frame *frame) +{ + return frame->datalen; +} + +static int ulaw_length(unsigned int samples) +{ + return samples; +} + +static struct ast_codec ulaw = { + .name = "ulaw", + .description = "G.711 u-law", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 150, + .default_ms = 20, + .minimum_bytes = 80, + .samples_count = ulaw_samples, + .get_length = ulaw_length, + .smooth = 1, +}; + +static struct ast_codec alaw = { + .name = "alaw", + .description = "G.711 a-law", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 150, + .default_ms = 20, + .minimum_bytes = 80, + .samples_count = ulaw_samples, + .get_length = ulaw_length, + .smooth = 1, +}; + +static int gsm_samples(struct ast_frame *frame) +{ + return 160 * (frame->datalen / 33); +} + +static int gsm_length(unsigned int samples) +{ + return (samples / 160) * 33; +} + +static struct ast_codec gsm = { + .name = "gsm", + .description = "GSM", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 20, + .maximum_ms = 300, + .default_ms = 20, + .minimum_bytes = 33, + .samples_count = gsm_samples, + .get_length = gsm_length, + .smooth = 1, +}; + +static int g726_samples(struct ast_frame *frame) +{ + return frame->datalen * 2; +} + +static int g726_length(unsigned int samples) +{ + return samples / 2; +} + +static struct ast_codec g726rfc3551 = { + .name = "g726", + .description = "G.726 RFC3551", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 300, + .default_ms = 20, + .minimum_bytes = 40, + .samples_count = g726_samples, + .get_length = g726_length, + .smooth = 1, +}; + +static struct ast_codec g726aal2 = { + .name = "g726aal2", + .description = "G.726 AAL2", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 300, + .default_ms = 20, + .minimum_bytes = 40, + .samples_count = g726_samples, + .get_length = g726_length, + .smooth = 1, +}; + +static struct ast_codec adpcm = { + .name = "adpcm", + .description = "Dialogic ADPCM", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 300, + .default_ms = 20, + .minimum_bytes = 40, + .samples_count = g726_samples, + .get_length = g726_length, + .smooth = 1, +}; + +static int slin_samples(struct ast_frame *frame) +{ + return frame->datalen / 2; +} + +static int slin_length(unsigned int samples) +{ + return samples * 2; +} + +static struct ast_codec slin8 = { + .name = "slin", + .description = "16 bit Signed Linear PCM", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 160, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin12 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (12kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 12000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 240, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin16 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (16kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 16000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 320, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin24 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (24kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 24000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 480, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin32 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (32kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 32000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 640, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin44 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (44kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 44100, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 882, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin48 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (48kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 48000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 960, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin96 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (96kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 96000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 1920, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static struct ast_codec slin192 = { + .name = "slin", + .description = "16 bit Signed Linear PCM (192kHz)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 192000, + .minimum_ms = 10, + .maximum_ms = 70, + .default_ms = 20, + .minimum_bytes = 3840, + .samples_count = slin_samples, + .get_length = slin_length, + .smooth = 1, +}; + +static int lpc10_samples(struct ast_frame *frame) +{ + int samples = 22 * 8; + + /* assumes that the RTP packet contains one LPC10 frame */ + samples += (((char *)(frame->data.ptr))[7] & 0x1) * 8; + + return samples; +} + +static struct ast_codec lpc10 = { + .name = "lpc10", + .description = "LPC10", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 20, + .maximum_ms = 20, + .default_ms = 20, + .minimum_bytes = 7, + .samples_count = lpc10_samples, + .smooth = 1, +}; + +static int g729_samples(struct ast_frame *frame) +{ + return frame->datalen * 8; +} + +static int g729_length(unsigned int samples) +{ + return samples / 8; +} + +static struct ast_codec g729a = { + .name = "g729", + .description = "G.729A", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 230, + .default_ms = 20, + .minimum_bytes = 10, + .samples_count = g729_samples, + .get_length = g729_length, + .smooth = 1, +}; + +static unsigned char get_n_bits_at(unsigned char *data, int n, int bit) +{ + int byte = bit / 8; /* byte containing first bit */ + int rem = 8 - (bit % 8); /* remaining bits in first byte */ + unsigned char ret = 0; + + if (n <= 0 || n > 8) + return 0; + + if (rem < n) { + ret = (data[byte] << (n - rem)); + ret |= (data[byte + 1] >> (8 - n + rem)); + } else { + ret = (data[byte] >> (rem - n)); + } + + return (ret & (0xff >> (8 - n))); +} + +static int speex_get_wb_sz_at(unsigned char *data, int len, int bit) +{ + static const int SpeexWBSubModeSz[] = { + 4, 36, 112, 192, + 352, 0, 0, 0 }; + int off = bit; + unsigned char c; + + /* skip up to two wideband frames */ + if (((len * 8 - off) >= 5) && + get_n_bits_at(data, 1, off)) { + c = get_n_bits_at(data, 3, off + 1); + off += SpeexWBSubModeSz[c]; + + if (((len * 8 - off) >= 5) && + get_n_bits_at(data, 1, off)) { + c = get_n_bits_at(data, 3, off + 1); + off += SpeexWBSubModeSz[c]; + + if (((len * 8 - off) >= 5) && + get_n_bits_at(data, 1, off)) { + ast_log(LOG_WARNING, "Encountered corrupt speex frame; too many wideband frames in a row.\n"); + return -1; + } + } + + } + return off - bit; +} + +static int speex_samples(unsigned char *data, int len) +{ + static const int SpeexSubModeSz[] = { + 5, 43, 119, 160, + 220, 300, 364, 492, + 79, 0, 0, 0, + 0, 0, 0, 0 }; + static const int SpeexInBandSz[] = { + 1, 1, 4, 4, + 4, 4, 4, 4, + 8, 8, 16, 16, + 32, 32, 64, 64 }; + int bit = 0; + int cnt = 0; + int off; + unsigned char c; + + while ((len * 8 - bit) >= 5) { + /* skip wideband frames */ + off = speex_get_wb_sz_at(data, len, bit); + if (off < 0) { + ast_log(LOG_WARNING, "Had error while reading wideband frames for speex samples\n"); + break; + } + bit += off; + + if ((len * 8 - bit) < 5) + break; + + /* get control bits */ + c = get_n_bits_at(data, 5, bit); + bit += 5; + + if (c == 15) { + /* terminator */ + break; + } else if (c == 14) { + /* in-band signal; next 4 bits contain signal id */ + c = get_n_bits_at(data, 4, bit); + bit += 4; + bit += SpeexInBandSz[c]; + } else if (c == 13) { + /* user in-band; next 4 bits contain msg len */ + c = get_n_bits_at(data, 4, bit); + bit += 4; + /* after which it's 5-bit signal id + c bytes of data */ + bit += 5 + c * 8; + } else if (c > 8) { + /* unknown */ + ast_log(LOG_WARNING, "Unknown speex control frame %d\n", c); + break; + } else { + /* skip number bits for submode (less the 5 control bits) */ + bit += SpeexSubModeSz[c] - 5; + cnt += 160; /* new frame */ + } + } + return cnt; +} + +static int speex8_samples(struct ast_frame *frame) +{ + return speex_samples(frame->data.ptr, frame->datalen); +} + +static struct ast_codec speex8 = { + .name = "speex", + .description = "SpeeX", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 60, + .default_ms = 20, + .minimum_bytes = 10, + .samples_count = speex8_samples, +}; + +static int speex16_samples(struct ast_frame *frame) +{ + return 2 * speex_samples(frame->data.ptr, frame->datalen); +} + +static struct ast_codec speex16 = { + .name = "speex", + .description = "SpeeX 16khz", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 16000, + .minimum_ms = 10, + .maximum_ms = 60, + .default_ms = 20, + .minimum_bytes = 10, + .samples_count = speex16_samples, +}; + +static int speex32_samples(struct ast_frame *frame) +{ + return 4 * speex_samples(frame->data.ptr, frame->datalen); +} + +static struct ast_codec speex32 = { + .name = "speex", + .description = "SpeeX 32khz", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 32000, + .minimum_ms = 10, + .maximum_ms = 60, + .default_ms = 20, + .minimum_bytes = 10, + .samples_count = speex32_samples, +}; + +static int ilbc_samples(struct ast_frame *frame) +{ + return 240 * (frame->datalen / 50); +} + +static struct ast_codec ilbc = { + .name = "ilbc", + .description = "iLBC", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 30, + .maximum_ms = 30, + .default_ms = 30, + .minimum_bytes = 50, + .samples_count = ilbc_samples, + .smooth = 1, +}; + +static struct ast_codec g722 = { + .name = "g722", + .description = "G722", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 16000, + .minimum_ms = 10, + .maximum_ms = 150, + .default_ms = 20, + .minimum_bytes = 80, + .samples_count = g726_samples, + .get_length = g726_length, + .smooth = 1, +}; + +static int siren7_samples(struct ast_frame *frame) +{ + return frame->datalen * (16000 / 4000); +} + +static int siren7_length(unsigned int samples) +{ + return samples / (16000 / 4000); +} + +static struct ast_codec siren7 = { + .name = "siren7", + .description = "ITU G.722.1 (Siren7, licensed from Polycom)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 16000, + .minimum_ms = 20, + .maximum_ms = 80, + .default_ms = 20, + .minimum_bytes = 80, + .samples_count = siren7_samples, + .get_length = siren7_length, +}; + +static int siren14_samples(struct ast_frame *frame) +{ + return (int) frame->datalen * ((float) 32000 / 6000); +} + +static int siren14_length(unsigned int samples) +{ + return (int) samples / ((float) 32000 / 6000);; +} + +static struct ast_codec siren14 = { + .name = "siren14", + .description = "ITU G.722.1 Annex C, (Siren14, licensed from Polycom)", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 32000, + .minimum_ms = 20, + .maximum_ms = 80, + .default_ms = 20, + .minimum_bytes = 120, + .samples_count = siren14_samples, + .get_length = siren14_length, +}; + +static struct ast_codec testlaw = { + .name = "testlaw", + .description = "G.711 test-law", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 8000, + .minimum_ms = 10, + .maximum_ms = 150, + .default_ms = 20, + .minimum_bytes = 80, + .samples_count = ulaw_samples, + .get_length = ulaw_length, + .smooth = 1, +}; + +static int g719_samples(struct ast_frame *frame) +{ + return (int) frame->datalen * ((float) 48000 / 8000); +} + +static int g719_length(unsigned int samples) +{ + return (int) samples / ((float) 48000 / 8000); +} + +static struct ast_codec g719 = { + .name = "g719", + .description = "ITU G.719", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 48000, + .minimum_ms = 20, + .maximum_ms = 80, + .default_ms = 20, + .minimum_bytes = 160, + .samples_count = g719_samples, + .get_length = g719_length, +}; + +static struct ast_codec opus = { + .name = "opus", + .description = "Opus Codec", + .type = AST_MEDIA_TYPE_AUDIO, + .sample_rate = 48000, + .minimum_ms = 20, + .maximum_ms = 60, + .default_ms = 20, + .minimum_bytes = 10, +}; + +static struct ast_codec jpeg = { + .name = "jpeg", + .description = "JPEG image", + .type = AST_MEDIA_TYPE_IMAGE, +}; + +static struct ast_codec png = { + .name = "png", + .description = "PNG Image", + .type = AST_MEDIA_TYPE_IMAGE, +}; + +static struct ast_codec h261 = { + .name = "h261", + .description = "H.261 video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec h263 = { + .name = "h263", + .description = "H.263 video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec h263p = { + .name = "h263p", + .description = "H.263+ video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec h264 = { + .name = "h264", + .description = "H.264 video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec mpeg4 = { + .name = "mpeg4", + .description = "MPEG4 video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec vp8 = { + .name = "vp8", + .description = "VP8 video", + .type = AST_MEDIA_TYPE_VIDEO, +}; + +static struct ast_codec t140red = { + .name = "red", + .description = "T.140 Realtime Text with redundancy", + .type = AST_MEDIA_TYPE_TEXT, +}; + +static struct ast_codec t140 = { + .name = "t140", + .description = "Passthrough T.140 Realtime Text", + .type = AST_MEDIA_TYPE_TEXT, +}; + +#define CODEC_REGISTER_AND_CACHE(codec) \ + ({ \ + int __res_ ## __LINE__ = 0; \ + struct ast_format *__fmt_ ## __LINE__; \ + struct ast_codec *__codec_ ## __LINE__; \ + res |= __ast_codec_register(&(codec), NULL); \ + __codec_ ## __LINE__ = ast_codec_get((codec).name, (codec).type, (codec).sample_rate); \ + __fmt_ ## __LINE__ = ast_format_create(__codec_ ## __LINE__); \ + res |= ast_format_cache_set(__fmt_ ## __LINE__); \ + ao2_ref(__fmt_ ## __LINE__, -1); \ + ao2_ref(__codec_ ## __LINE__, -1); \ + __res_ ## __LINE__; \ + }) + +#define CODEC_REGISTER_AND_CACHE_NAMED(format_name, codec) \ + ({ \ + int __res_ ## __LINE__ = 0; \ + struct ast_format *__fmt_ ## __LINE__; \ + struct ast_codec *__codec_ ## __LINE__; \ + res |= __ast_codec_register(&(codec), NULL); \ + __codec_ ## __LINE__ = ast_codec_get((codec).name, (codec).type, (codec).sample_rate); \ + __fmt_ ## __LINE__ = ast_format_create_named((format_name), __codec_ ## __LINE__); \ + res |= ast_format_cache_set(__fmt_ ## __LINE__); \ + ao2_ref(__fmt_ ## __LINE__, -1); \ + ao2_ref(__codec_ ## __LINE__, -1); \ + __res_ ## __LINE__; \ + }) + +int ast_codec_builtin_init(void) +{ + int res = 0; + + res |= CODEC_REGISTER_AND_CACHE(g723); + res |= CODEC_REGISTER_AND_CACHE(ulaw); + res |= CODEC_REGISTER_AND_CACHE(alaw); + res |= CODEC_REGISTER_AND_CACHE(gsm); + res |= CODEC_REGISTER_AND_CACHE(g726rfc3551); + res |= CODEC_REGISTER_AND_CACHE(g726aal2); + res |= CODEC_REGISTER_AND_CACHE(adpcm); + res |= CODEC_REGISTER_AND_CACHE(slin8); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin12", slin12); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin16", slin16); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin24", slin24); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin32", slin32); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin44", slin44); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin48", slin48); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin96", slin96); + res |= CODEC_REGISTER_AND_CACHE_NAMED("slin192", slin192); + res |= CODEC_REGISTER_AND_CACHE(lpc10); + res |= CODEC_REGISTER_AND_CACHE(g729a); + res |= CODEC_REGISTER_AND_CACHE(speex8); + res |= CODEC_REGISTER_AND_CACHE_NAMED("speex16", speex16); + res |= CODEC_REGISTER_AND_CACHE_NAMED("speex32", speex32); + res |= CODEC_REGISTER_AND_CACHE(ilbc); + res |= CODEC_REGISTER_AND_CACHE(g722); + res |= CODEC_REGISTER_AND_CACHE(siren7); + res |= CODEC_REGISTER_AND_CACHE(siren14); + res |= CODEC_REGISTER_AND_CACHE(testlaw); + res |= CODEC_REGISTER_AND_CACHE(g719); + res |= CODEC_REGISTER_AND_CACHE(opus); + res |= CODEC_REGISTER_AND_CACHE(jpeg); + res |= CODEC_REGISTER_AND_CACHE(png); + res |= CODEC_REGISTER_AND_CACHE(h261); + res |= CODEC_REGISTER_AND_CACHE(h263); + res |= CODEC_REGISTER_AND_CACHE(h263p); + res |= CODEC_REGISTER_AND_CACHE(h264); + res |= CODEC_REGISTER_AND_CACHE(mpeg4); + res |= CODEC_REGISTER_AND_CACHE(vp8); + res |= CODEC_REGISTER_AND_CACHE(t140red); + res |= CODEC_REGISTER_AND_CACHE(t140); + res |= CODEC_REGISTER_AND_CACHE(none); + + return res; +} |