/* $NetBSD: auconv.c,v 1.20 2007/03/01 17:31:35 thorpej Exp $ */ /* * Copyright (c) 1996 The NetBSD Foundation, Inc. * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. All advertising materials mentioning features or use of this software * must display the following acknowledgement: * This product includes software developed by the Computer Systems * Engineering Group at Lawrence Berkeley Laboratory. * 4. Neither the name of the University nor of the Laboratory may be used * to endorse or promote products derived from this software without * specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF * SUCH DAMAGE. * */ #include __KERNEL_RCSID(0, "$NetBSD: auconv.c,v 1.20 2007/03/01 17:31:35 thorpej Exp $"); #include #include #include #include #include #include #include #include #include #include #ifndef _KERNEL #include #include #include #include #include #endif #include /* generated by config(8) */ #include /* generated by config(8) */ /* #define AUCONV_DEBUG */ #ifdef AUCONV_DEBUG # define DPRINTF(x) printf x #else # define DPRINTF(x) #endif #if NAURATECONV > 0 static int auconv_rateconv_supportable(u_int, u_int, u_int); static int auconv_rateconv_check_channels(const struct audio_format *, int, int, const audio_params_t *, stream_filter_list_t *); static int auconv_rateconv_check_rates(const struct audio_format *, int, int, const audio_params_t *, audio_params_t *, stream_filter_list_t *); #endif #ifdef AUCONV_DEBUG static void auconv_dump_formats(const struct audio_format *, int); #endif static void auconv_dump_params(const audio_params_t *); static int auconv_exact_match(const struct audio_format *, int, int, const struct audio_params *); static u_int auconv_normalize_encoding(u_int, u_int); static int auconv_is_supported_rate(const struct audio_format *, u_int); static int auconv_add_encoding(int, int, int, struct audio_encoding_set **, int *); #ifdef _KERNEL #define AUCONV_MALLOC(size) malloc(size, M_DEVBUF, M_NOWAIT) #define AUCONV_REALLOC(p, size) realloc(p, size, M_DEVBUF, M_NOWAIT) #define AUCONV_FREE(p) free(p, M_DEVBUF) #else #define AUCONV_MALLOC(size) malloc(size) #define AUCONV_REALLOC(p, size) realloc(p, size) #define AUCONV_FREE(p) free(p) #endif struct audio_encoding_set { int size; audio_encoding_t items[1]; }; #define ENCODING_SET_SIZE(n) (offsetof(struct audio_encoding_set, items) \ + sizeof(audio_encoding_t) * (n)) struct conv_table { u_int encoding; u_int validbits; u_int precision; stream_filter_factory_t *play_conv; stream_filter_factory_t *rec_conv; }; /* * SLINEAR-16 or SLINEAR-24 should precede in a table because * aurateconv supports only SLINEAR. */ static const struct conv_table s8_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {AUDIO_ENCODING_SLINEAR_BE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {AUDIO_ENCODING_ULINEAR_LE, 8, 8, change_sign8, change_sign8}, {0, 0, 0, NULL, NULL}}; static const struct conv_table u8_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {AUDIO_ENCODING_SLINEAR_BE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {AUDIO_ENCODING_SLINEAR_LE, 8, 8, change_sign8, change_sign8}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, linear8_to_linear16, linear16_to_linear8}, {0, 0, 0, NULL, NULL}}; static const struct conv_table s16le_table[] = { {AUDIO_ENCODING_SLINEAR_BE, 16, 16, swap_bytes, swap_bytes}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, change_sign16, change_sign16}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, swap_bytes_change_sign16, swap_bytes_change_sign16}, {0, 0, 0, NULL, NULL}}; static const struct conv_table s16be_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, swap_bytes, swap_bytes}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, change_sign16, change_sign16}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, swap_bytes_change_sign16, swap_bytes_change_sign16}, {0, 0, 0, NULL, NULL}}; static const struct conv_table u16le_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, change_sign16, change_sign16}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, swap_bytes, swap_bytes}, {AUDIO_ENCODING_SLINEAR_BE, 16, 16, swap_bytes_change_sign16, swap_bytes_change_sign16}, {0, 0, 0, NULL, NULL}}; static const struct conv_table u16be_table[] = { {AUDIO_ENCODING_SLINEAR_BE, 16, 16, change_sign16, change_sign16}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, swap_bytes, swap_bytes}, {AUDIO_ENCODING_SLINEAR_LE, 16, 16, swap_bytes_change_sign16, swap_bytes_change_sign16}, {0, 0, 0, NULL, NULL}}; #if NMULAW > 0 static const struct conv_table mulaw_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, mulaw_to_linear16, linear16_to_mulaw}, {AUDIO_ENCODING_SLINEAR_BE, 16, 16, mulaw_to_linear16, linear16_to_mulaw}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, mulaw_to_linear16, linear16_to_mulaw}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, mulaw_to_linear16, linear16_to_mulaw}, {AUDIO_ENCODING_SLINEAR_LE, 8, 8, mulaw_to_linear8, linear8_to_mulaw}, {AUDIO_ENCODING_ULINEAR_LE, 8, 8, mulaw_to_linear8, linear8_to_mulaw}, {0, 0, 0, NULL, NULL}}; static const struct conv_table alaw_table[] = { {AUDIO_ENCODING_SLINEAR_LE, 16, 16, alaw_to_linear16, linear16_to_alaw}, {AUDIO_ENCODING_SLINEAR_BE, 16, 16, alaw_to_linear16, linear16_to_alaw}, {AUDIO_ENCODING_ULINEAR_LE, 16, 16, alaw_to_linear16, linear16_to_alaw}, {AUDIO_ENCODING_ULINEAR_BE, 16, 16, alaw_to_linear16, linear16_to_alaw}, {AUDIO_ENCODING_SLINEAR_LE, 8, 8, alaw_to_linear8, linear8_to_alaw}, {AUDIO_ENCODING_ULINEAR_LE, 8, 8, alaw_to_linear8, linear8_to_alaw}, {0, 0, 0, NULL, NULL}}; #endif #ifdef AUCONV_DEBUG static const char *encoding_dbg_names[] = { "none", AudioEmulaw, AudioEalaw, "pcm16", "pcm8", AudioEadpcm, AudioEslinear_le, AudioEslinear_be, AudioEulinear_le, AudioEulinear_be, AudioEslinear, AudioEulinear, AudioEmpeg_l1_stream, AudioEmpeg_l1_packets, AudioEmpeg_l1_system, AudioEmpeg_l2_stream, AudioEmpeg_l2_packets, AudioEmpeg_l2_system }; #endif void stream_filter_set_fetcher(stream_filter_t *this, stream_fetcher_t *p) { this->prev = p; } void stream_filter_set_inputbuffer(stream_filter_t *this, audio_stream_t *stream) { this->src = stream; } stream_filter_t * auconv_nocontext_filter_factory( int (*fetch_to)(stream_fetcher_t *, audio_stream_t *, int)) { stream_filter_t *this; this = AUCONV_MALLOC(sizeof(stream_filter_t)); if (this == NULL) return NULL; this->base.fetch_to = fetch_to; this->dtor = auconv_nocontext_filter_dtor; this->set_fetcher = stream_filter_set_fetcher; this->set_inputbuffer = stream_filter_set_inputbuffer; this->prev = NULL; this->src = NULL; return this; } void auconv_nocontext_filter_dtor(struct stream_filter *this) { if (this != NULL) AUCONV_FREE(this); } #define DEFINE_FILTER(name) \ static int \ name##_fetch_to(stream_fetcher_t *, audio_stream_t *, int); \ stream_filter_t * \ name(struct audio_softc *sc, const audio_params_t *from, \ const audio_params_t *to) \ { \ return auconv_nocontext_filter_factory(name##_fetch_to); \ } \ static int \ name##_fetch_to(stream_fetcher_t *self, audio_stream_t *dst, int max_used) DEFINE_FILTER(change_sign8) { stream_filter_t *this; int m, err; this = (stream_filter_t *)self; if ((err = this->prev->fetch_to(this->prev, this->src, max_used))) return err; m = dst->end - dst->start; m = min(m, max_used); FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) { *d = *s ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); return 0; } DEFINE_FILTER(change_sign16) { stream_filter_t *this; int m, err, enc; this = (stream_filter_t *)self; max_used = (max_used + 1) & ~1; /* round up to even */ if ((err = this->prev->fetch_to(this->prev, this->src, max_used))) return err; m = (dst->end - dst->start) & ~1; m = min(m, max_used); enc = dst->param.encoding; if (enc == AUDIO_ENCODING_SLINEAR_LE || enc == AUDIO_ENCODING_ULINEAR_LE) { FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { d[0] = s[0]; d[1] = s[1] ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); } else { FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { d[0] = s[0] ^ 0x80; d[1] = s[1]; } FILTER_LOOP_EPILOGUE(this->src, dst); } return 0; } DEFINE_FILTER(swap_bytes) { stream_filter_t *this; int m, err; this = (stream_filter_t *)self; max_used = (max_used + 1) & ~1; /* round up to even */ if ((err = this->prev->fetch_to(this->prev, this->src, max_used))) return err; m = (dst->end - dst->start) & ~1; m = min(m, max_used); FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { d[0] = s[1]; d[1] = s[0]; } FILTER_LOOP_EPILOGUE(this->src, dst); return 0; } DEFINE_FILTER(swap_bytes_change_sign16) { stream_filter_t *this; int m, err, enc; this = (stream_filter_t *)self; max_used = (max_used + 1) & ~1; /* round up to even */ if ((err = this->prev->fetch_to(this->prev, this->src, max_used))) return err; m = (dst->end - dst->start) & ~1; m = min(m, max_used); enc = dst->param.encoding; if (enc == AUDIO_ENCODING_SLINEAR_LE || enc == AUDIO_ENCODING_ULINEAR_LE) { FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { d[0] = s[1]; d[1] = s[0] ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); } else { FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { d[0] = s[1] ^ 0x80; d[1] = s[0]; } FILTER_LOOP_EPILOGUE(this->src, dst); } return 0; } DEFINE_FILTER(linear8_to_linear16) { stream_filter_t *this; int m, err, enc_dst, enc_src; this = (stream_filter_t *)self; max_used = (max_used + 1) & ~1; /* round up to even */ if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 2))) return err; m = (dst->end - dst->start) & ~1; m = min(m, max_used); enc_dst = dst->param.encoding; enc_src = this->src->param.encoding; if ((enc_src == AUDIO_ENCODING_SLINEAR_LE && enc_dst == AUDIO_ENCODING_SLINEAR_LE) || (enc_src == AUDIO_ENCODING_ULINEAR_LE && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { /* * slinear8 -> slinear16_le * ulinear8 -> ulinear16_le */ FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { d[0] = 0; d[1] = s[0]; } FILTER_LOOP_EPILOGUE(this->src, dst); } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE && enc_dst == AUDIO_ENCODING_SLINEAR_BE) || (enc_src == AUDIO_ENCODING_ULINEAR_LE && enc_dst == AUDIO_ENCODING_ULINEAR_BE)) { /* * slinear8 -> slinear16_be * ulinear8 -> ulinear16_be */ FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { d[0] = s[0]; d[1] = 0; } FILTER_LOOP_EPILOGUE(this->src, dst); } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE && enc_dst == AUDIO_ENCODING_ULINEAR_LE) || (enc_src == AUDIO_ENCODING_ULINEAR_LE && enc_dst == AUDIO_ENCODING_SLINEAR_LE)) { /* * slinear8 -> ulinear16_le * ulinear8 -> slinear16_le */ FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { d[0] = 0; d[1] = s[0] ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); } else { /* * slinear8 -> ulinear16_be * ulinear8 -> slinear16_be */ FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { d[0] = s[0] ^ 0x80; d[1] = 0; } FILTER_LOOP_EPILOGUE(this->src, dst); } return 0; } DEFINE_FILTER(linear16_to_linear8) { stream_filter_t *this; int m, err, enc_src, enc_dst; this = (stream_filter_t *)self; if ((err = this->prev->fetch_to(this->prev, this->src, max_used * 2))) return err; m = dst->end - dst->start; m = min(m, max_used); enc_dst = dst->param.encoding; enc_src = this->src->param.encoding; if ((enc_src == AUDIO_ENCODING_SLINEAR_LE && enc_dst == AUDIO_ENCODING_SLINEAR_LE) || (enc_src == AUDIO_ENCODING_ULINEAR_LE && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { /* * slinear16_le -> slinear8 * ulinear16_le -> ulinear8 */ FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { d[0] = s[1]; } FILTER_LOOP_EPILOGUE(this->src, dst); } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE && enc_dst == AUDIO_ENCODING_ULINEAR_LE) || (enc_src == AUDIO_ENCODING_ULINEAR_LE && enc_dst == AUDIO_ENCODING_SLINEAR_LE)) { /* * slinear16_le -> ulinear8 * ulinear16_le -> slinear8 */ FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { d[0] = s[1] ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); } else if ((enc_src == AUDIO_ENCODING_SLINEAR_BE && enc_dst == AUDIO_ENCODING_SLINEAR_LE) || (enc_src == AUDIO_ENCODING_ULINEAR_BE && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { /* * slinear16_be -> slinear8 * ulinear16_be -> ulinear8 */ FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { d[0] = s[0]; } FILTER_LOOP_EPILOGUE(this->src, dst); } else { /* * slinear16_be -> ulinear8 * ulinear16_be -> slinear8 */ FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { d[0] = s[0] ^ 0x80; } FILTER_LOOP_EPILOGUE(this->src, dst); } return 0; } /** * Set appropriate parameters in `param,' and return the index in * the hardware capability array `formats.' * * @param formats [IN] An array of formats which a hardware can support. * @param nformats [IN] The number of elements of the array. * @param mode [IN] Either AUMODE_PLAY or AUMODE_RECORD. * @param param [IN] Requested format. param->sw_code may be set. * @param rateconv [IN] true if aurateconv may be used. * @param list [OUT] stream_filters required for param. * @return The index of selected audio_format entry. -1 if the device * can not support the specified param. */ int auconv_set_converter(const struct audio_format *formats, int nformats, int mode, const audio_params_t *param, int rateconv, stream_filter_list_t *list) { audio_params_t work; const struct conv_table *table; stream_filter_factory_t *conv; int enc; int i, j; #ifdef AUCONV_DEBUG DPRINTF(("%s: ENTER rateconv=%d\n", __func__, rateconv)); auconv_dump_formats(formats, nformats); #endif enc = auconv_normalize_encoding(param->encoding, param->precision); /* check support by native format */ i = auconv_exact_match(formats, nformats, mode, param); if (i >= 0) { DPRINTF(("%s: LEAVE with %d (exact)\n", __func__, i)); return i; } #if NAURATECONV > 0 /* native format with aurateconv */ DPRINTF(("%s: native with aurateconv\n", __func__)); if (rateconv && auconv_rateconv_supportable(enc, param->precision, param->validbits)) { i = auconv_rateconv_check_channels(formats, nformats, mode, param, list); if (i >= 0) { DPRINTF(("%s: LEAVE with %d (aurateconv1)\n", __func__, i)); return i; } } #endif /* check for emulation */ DPRINTF(("%s: encoding emulation\n", __func__)); table = NULL; switch (enc) { case AUDIO_ENCODING_SLINEAR_LE: if (param->precision == 8) table = s8_table; else if (param->precision == 16) table = s16le_table; break; case AUDIO_ENCODING_SLINEAR_BE: if (param->precision == 8) table = s8_table; else if (param->precision == 16) table = s16be_table; break; case AUDIO_ENCODING_ULINEAR_LE: if (param->precision == 8) table = u8_table; else if (param->precision == 16) table = u16le_table; break; case AUDIO_ENCODING_ULINEAR_BE: if (param->precision == 8) table = u8_table; else if (param->precision == 16) table = u16be_table; break; #if NMULAW > 0 case AUDIO_ENCODING_ULAW: table = mulaw_table; break; case AUDIO_ENCODING_ALAW: table = alaw_table; break; #endif } if (table == NULL) { DPRINTF(("%s: LEAVE with -1 (no-emultable)\n", __func__)); return -1; } work = *param; for (j = 0; table[j].precision != 0; j++) { work.encoding = table[j].encoding; work.precision = table[j].precision; work.validbits = table[j].validbits; i = auconv_exact_match(formats, nformats, mode, &work); if (i >= 0) { conv = mode == AUMODE_PLAY ? table[j].play_conv : table[j].rec_conv; list->append(list, conv, &work); DPRINTF(("%s: LEAVE with %d (emultable)\n", __func__, i)); return i; } } /* not found */ #if NAURATECONV > 0 /* emulation with aurateconv */ DPRINTF(("%s: encoding emulation with aurateconv\n", __func__)); if (!rateconv) { DPRINTF(("%s: LEAVE with -1 (no-rateconv)\n", __func__)); return -1; } work = *param; for (j = 0; table[j].precision != 0; j++) { if (!auconv_rateconv_supportable(table[j].encoding, table[j].precision, table[j].validbits)) continue; work.encoding = table[j].encoding; work.precision = table[j].precision; work.validbits = table[j].validbits; i = auconv_rateconv_check_channels(formats, nformats, mode, &work, list); if (i >= 0) { /* work<=>hw conversion is already registered */ conv = mode == AUMODE_PLAY ? table[j].play_conv : table[j].rec_conv; /* register userland<=>work conversion */ list->append(list, conv, &work); DPRINTF(("%s: LEAVE with %d (rateconv2)\n", __func__, i)); return i; } } #endif DPRINTF(("%s: LEAVE with -1 (bottom)\n", __func__)); return -1; } #if NAURATECONV > 0 static int auconv_rateconv_supportable(u_int encoding, u_int precision, u_int validbits) { if (encoding != AUDIO_ENCODING_SLINEAR_LE && encoding != AUDIO_ENCODING_SLINEAR_BE) return false; if (precision != 16 && precision != 24 && precision != 32) return false; if (precision < validbits) return false; return true; } static int auconv_rateconv_check_channels(const struct audio_format *formats, int nformats, int mode, const audio_params_t *param, stream_filter_list_t *list) { audio_params_t hw_param; int ind, n; hw_param = *param; /* check for the specified number of channels */ ind = auconv_rateconv_check_rates(formats, nformats, mode, param, &hw_param, list); if (ind >= 0) return ind; /* check for larger numbers */ for (n = param->channels + 1; n <= AUDIO_MAX_CHANNELS; n++) { hw_param.channels = n; ind = auconv_rateconv_check_rates(formats, nformats, mode, param, &hw_param, list); if (ind >= 0) return ind; } /* check for stereo:monaural conversion */ if (param->channels == 2) { hw_param.channels = 1; ind = auconv_rateconv_check_rates(formats, nformats, mode, param, &hw_param, list); if (ind >= 0) return ind; } return -1; } static int auconv_rateconv_check_rates(const struct audio_format *formats, int nformats, int mode, const audio_params_t *param, audio_params_t *hw_param, stream_filter_list_t *list) { int ind, i, j, enc, f_enc; u_int rate, minrate, maxrate, orig_rate;; /* exact match */ ind = auconv_exact_match(formats, nformats, mode, hw_param); if (ind >= 0) goto found; /* determine min/max of specified encoding/precision/channels */ minrate = UINT_MAX; maxrate = 0; enc = auconv_normalize_encoding(param->encoding, param->precision); for (i = 0; i < nformats; i++) { if (!AUFMT_IS_VALID(&formats[i])) continue; if ((formats[i].mode & mode) == 0) continue; f_enc = auconv_normalize_encoding(formats[i].encoding, formats[i].precision); if (f_enc != enc) continue; if (formats[i].validbits != hw_param->validbits) continue; if (formats[i].precision != hw_param->precision) continue; if (formats[i].channels != hw_param->channels) continue; if (formats[i].frequency_type == 0) { if (formats[i].frequency[0] < minrate) minrate = formats[i].frequency[0]; if (formats[i].frequency[1] > maxrate) maxrate = formats[i].frequency[1]; } else { for (j = 0; j < formats[i].frequency_type; j++) { if (formats[i].frequency[j] < minrate) minrate = formats[i].frequency[j]; if (formats[i].frequency[j] > maxrate) maxrate = formats[i].frequency[j]; } } } if (maxrate == 0) return -1; /* try multiples of sample_rate */ orig_rate = hw_param->sample_rate; for (i = 2; (rate = param->sample_rate * i) <= maxrate; i++) { hw_param->sample_rate = rate; ind = auconv_exact_match(formats, nformats, mode, hw_param); if (ind >= 0) goto found; } hw_param->sample_rate = param->sample_rate >= minrate ? maxrate : minrate; ind = auconv_exact_match(formats, nformats, mode, hw_param); if (ind >= 0) goto found; hw_param->sample_rate = orig_rate; return -1; found: list->append(list, aurateconv, hw_param); return ind; } #endif /* NAURATECONV */ #ifdef AUCONV_DEBUG static void auconv_dump_formats(const struct audio_format *formats, int nformats) { const struct audio_format *f; int i, j; for (i = 0; i < nformats; i++) { f = &formats[i]; printf("[%2d]: mode=", i); if (!AUFMT_IS_VALID(f)) { printf("INVALID"); } else if (f->mode == AUMODE_PLAY) { printf("PLAY"); } else if (f->mode == AUMODE_RECORD) { printf("RECORD"); } else if (f->mode == (AUMODE_PLAY | AUMODE_RECORD)) { printf("PLAY|RECORD"); } else { printf("0x%x", f->mode); } printf(" enc=%s", encoding_dbg_names[f->encoding]); printf(" %u/%ubit", f->validbits, f->precision); printf(" %uch", f->channels); printf(" channel_mask="); if (f->channel_mask == AUFMT_MONAURAL) { printf("MONAURAL"); } else if (f->channel_mask == AUFMT_STEREO) { printf("STEREO"); } else if (f->channel_mask == AUFMT_SURROUND4) { printf("SURROUND4"); } else if (f->channel_mask == AUFMT_DOLBY_5_1) { printf("DOLBY5.1"); } else { printf("0x%x", f->channel_mask); } if (f->frequency_type == 0) { printf(" %uHz-%uHz", f->frequency[0], f->frequency[1]); } else { printf(" %uHz", f->frequency[0]); for (j = 1; j < f->frequency_type; j++) printf(",%uHz", f->frequency[j]); } printf("\n"); } } static void auconv_dump_params(const audio_params_t *p) { printf("enc=%s", encoding_dbg_names[p->encoding]); printf(" %u/%ubit", p->validbits, p->precision); printf(" %uch", p->channels); printf(" %uHz", p->sample_rate); printf("\n"); } #else static void auconv_dump_params(const audio_params_t *p) { } #endif /* AUCONV_DEBUG */ /** * a sub-routine for auconv_set_converter() */ static int auconv_exact_match(const struct audio_format *formats, int nformats, int mode, const audio_params_t *param) { int i, enc, f_enc; DPRINTF(("%s: ENTER: mode=0x%x target:", __func__, mode)); auconv_dump_params(param); enc = auconv_normalize_encoding(param->encoding, param->precision); DPRINTF(("%s: target normalized: %s\n", __func__, encoding_dbg_names[enc])); for (i = 0; i < nformats; i++) { if (!AUFMT_IS_VALID(&formats[i])) continue; if ((formats[i].mode & mode) == 0) continue; f_enc = auconv_normalize_encoding(formats[i].encoding, formats[i].precision); DPRINTF(("%s: format[%d] normalized: %s\n", __func__, i, encoding_dbg_names[f_enc])); if (f_enc != enc) continue; /** * XXX we need encoding-dependent check. * XXX Is to check precision/channels meaningful for * MPEG encodings? */ if (formats[i].validbits != param->validbits) continue; if (formats[i].precision != param->precision) continue; if (formats[i].channels != param->channels) continue; if (!auconv_is_supported_rate(&formats[i], param->sample_rate)) continue; return i; } return -1; } /** * a sub-routine for auconv_set_converter() * SLINEAR ==> SLINEAR_ * ULINEAR ==> ULINEAR_ * SLINEAR_BE 8bit ==> SLINEAR_LE 8bit * ULINEAR_BE 8bit ==> ULINEAR_LE 8bit * This should be the same rule as audio_check_params() */ static u_int auconv_normalize_encoding(u_int encoding, u_int precision) { int enc; enc = encoding; if (enc == AUDIO_ENCODING_SLINEAR_LE) return enc; if (enc == AUDIO_ENCODING_ULINEAR_LE) return enc; #if BYTE_ORDER == LITTLE_ENDIAN if (enc == AUDIO_ENCODING_SLINEAR) return AUDIO_ENCODING_SLINEAR_LE; else if (enc == AUDIO_ENCODING_ULINEAR) return AUDIO_ENCODING_ULINEAR_LE; #else if (enc == AUDIO_ENCODING_SLINEAR) enc = AUDIO_ENCODING_SLINEAR_BE; else if (enc == AUDIO_ENCODING_ULINEAR) enc = AUDIO_ENCODING_ULINEAR_BE; #endif if (precision == 8 && enc == AUDIO_ENCODING_SLINEAR_BE) return AUDIO_ENCODING_SLINEAR_LE; if (precision == 8 && enc == AUDIO_ENCODING_ULINEAR_BE) return AUDIO_ENCODING_ULINEAR_LE; return enc; } /** * a sub-routine for auconv_set_converter() */ static int auconv_is_supported_rate(const struct audio_format *format, u_int rate) { u_int i; if (format->frequency_type == 0) { return format->frequency[0] <= rate && rate <= format->frequency[1]; } for (i = 0; i < format->frequency_type; i++) { if (format->frequency[i] == rate) return true; } return false; } /** * Create an audio_encoding_set besed on hardware capability represented * by audio_format. * * Usage: * foo_attach(...) { * : * if (auconv_create_encodings(formats, nformats, * &sc->sc_encodings) != 0) { * // attach failure * } * * @param formats [IN] An array of formats which a hardware can support. * @param nformats [IN] The number of elements of the array. * @param encodings [OUT] receives an address of an audio_encoding_set. * @return errno; 0 for success. */ int auconv_create_encodings(const struct audio_format *formats, int nformats, struct audio_encoding_set **encodings) { struct audio_encoding_set *buf; int capacity; int i; int err; #define ADD_ENCODING(enc, prec, flags) do { \ err = auconv_add_encoding(enc, prec, flags, &buf, &capacity); \ if (err != 0) goto err_exit; \ } while (/*CONSTCOND*/0) capacity = 10; buf = AUCONV_MALLOC(ENCODING_SET_SIZE(capacity)); if (buf == NULL) { err = ENOMEM; goto err_exit; } buf->size = 0; for (i = 0; i < nformats; i++) { if (!AUFMT_IS_VALID(&formats[i])) continue; switch (formats[i].encoding) { case AUDIO_ENCODING_SLINEAR_LE: ADD_ENCODING(formats[i].encoding, formats[i].precision, 0); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); #if NMULAW > 0 if (formats[i].precision == 8 || formats[i].precision == 16) { ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, AUDIO_ENCODINGFLAG_EMULATED); } #endif break; case AUDIO_ENCODING_SLINEAR_BE: ADD_ENCODING(formats[i].encoding, formats[i].precision, 0); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); #if NMULAW > 0 if (formats[i].precision == 8 || formats[i].precision == 16) { ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, AUDIO_ENCODINGFLAG_EMULATED); } #endif break; case AUDIO_ENCODING_ULINEAR_LE: ADD_ENCODING(formats[i].encoding, formats[i].precision, 0); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); #if NMULAW > 0 if (formats[i].precision == 8 || formats[i].precision == 16) { ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, AUDIO_ENCODINGFLAG_EMULATED); } #endif break; case AUDIO_ENCODING_ULINEAR_BE: ADD_ENCODING(formats[i].encoding, formats[i].precision, 0); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, formats[i].precision, AUDIO_ENCODINGFLAG_EMULATED); #if NMULAW > 0 if (formats[i].precision == 8 || formats[i].precision == 16) { ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, AUDIO_ENCODINGFLAG_EMULATED); ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, AUDIO_ENCODINGFLAG_EMULATED); } #endif break; case AUDIO_ENCODING_ULAW: case AUDIO_ENCODING_ALAW: case AUDIO_ENCODING_ADPCM: case AUDIO_ENCODING_MPEG_L1_STREAM: case AUDIO_ENCODING_MPEG_L1_PACKETS: case AUDIO_ENCODING_MPEG_L1_SYSTEM: case AUDIO_ENCODING_MPEG_L2_STREAM: case AUDIO_ENCODING_MPEG_L2_PACKETS: case AUDIO_ENCODING_MPEG_L2_SYSTEM: ADD_ENCODING(formats[i].encoding, formats[i].precision, 0); break; case AUDIO_ENCODING_SLINEAR: case AUDIO_ENCODING_ULINEAR: case AUDIO_ENCODING_LINEAR: case AUDIO_ENCODING_LINEAR8: default: printf("%s: invalid encoding value " "for audio_format: %d\n", __func__, formats[i].encoding); break; } } *encodings = buf; return 0; err_exit: if (buf != NULL) AUCONV_FREE(buf); *encodings = NULL; return err; } /** * a sub-routine for auconv_create_encodings() */ static int auconv_add_encoding(int enc, int prec, int flags, struct audio_encoding_set **buf, int *capacity) { static const char *encoding_names[] = { NULL, AudioEmulaw, AudioEalaw, NULL, NULL, AudioEadpcm, AudioEslinear_le, AudioEslinear_be, AudioEulinear_le, AudioEulinear_be, AudioEslinear, AudioEulinear, AudioEmpeg_l1_stream, AudioEmpeg_l1_packets, AudioEmpeg_l1_system, AudioEmpeg_l2_stream, AudioEmpeg_l2_packets, AudioEmpeg_l2_system }; struct audio_encoding_set *set; struct audio_encoding_set *new_buf; audio_encoding_t *e; int i; set = *buf; /* already has the same encoding? */ e = set->items; for (i = 0; i < set->size; i++, e++) { if (e->encoding == enc && e->precision == prec) { /* overwrite EMULATED flag */ if ((e->flags & AUDIO_ENCODINGFLAG_EMULATED) && (flags & AUDIO_ENCODINGFLAG_EMULATED) == 0) { e->flags &= ~AUDIO_ENCODINGFLAG_EMULATED; } return 0; } } /* We don't have the specified one. */ if (set->size >= *capacity) { new_buf = AUCONV_REALLOC(set, ENCODING_SET_SIZE(*capacity + 10)); if (new_buf == NULL) return ENOMEM; *buf = new_buf; set = new_buf; *capacity += 10; } e = &set->items[set->size]; e->index = 0; strlcpy(e->name, encoding_names[enc], MAX_AUDIO_DEV_LEN); e->encoding = enc; e->precision = prec; e->flags = flags; set->size += 1; return 0; } /** * Delete an audio_encoding_set created by auconv_create_encodings(). * * Usage: * foo_detach(...) { * : * auconv_delete_encodings(sc->sc_encodings); * : * } * * @param encodings [IN] An audio_encoding_set which was created by * auconv_create_encodings(). * @return errno; 0 for success. */ int auconv_delete_encodings(struct audio_encoding_set *encodings) { if (encodings != NULL) AUCONV_FREE(encodings); return 0; } /** * Copy the element specified by aep->index. * * Usage: * int foo_query_encoding(void *v, audio_encoding_t *aep) { * struct foo_softc *sc = (struct foo_softc *)v; * return auconv_query_encoding(sc->sc_encodings, aep); * } * * @param encodings [IN] An audio_encoding_set created by * auconv_create_encodings(). * @param aep [IN/OUT] resultant audio_encoding_t. */ int auconv_query_encoding(const struct audio_encoding_set *encodings, audio_encoding_t *aep) { if (aep->index >= encodings->size) return EINVAL; strlcpy(aep->name, encodings->items[aep->index].name, MAX_AUDIO_DEV_LEN); aep->encoding = encodings->items[aep->index].encoding; aep->precision = encodings->items[aep->index].precision; aep->flags = encodings->items[aep->index].flags; return 0; }