diff options
-rw-r--r-- | codecs/codec_gsm.c | 29 | ||||
-rw-r--r-- | codecs/codec_ilbc.c | 28 | ||||
-rw-r--r-- | codecs/codec_lpc10.c | 41 | ||||
-rw-r--r-- | codecs/codec_speex.c | 60 | ||||
-rw-r--r-- | main/translate.c | 55 |
5 files changed, 140 insertions, 73 deletions
diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c index 4660048c8..f80c955a6 100644 --- a/codecs/codec_gsm.c +++ b/codecs/codec_gsm.c @@ -39,6 +39,7 @@ ASTERISK_REGISTER_FILE() #include "asterisk/config.h" #include "asterisk/module.h" #include "asterisk/utils.h" +#include "asterisk/linkedlists.h" #ifdef HAVE_GSM_HEADER #include "gsm.h" @@ -139,25 +140,35 @@ static int lintogsm_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) static struct ast_frame *lintogsm_frameout(struct ast_trans_pvt *pvt) { struct gsm_translator_pvt *tmp = pvt->pvt; - int datalen = 0; - int samples = 0; + struct ast_frame *result = NULL; + struct ast_frame *last = NULL; + int samples = 0; /* output samples */ - /* We can't work on anything less than a frame in size */ - if (pvt->samples < GSM_SAMPLES) - return NULL; while (pvt->samples >= GSM_SAMPLES) { + struct ast_frame *current; + /* Encode a frame of data */ - gsm_encode(tmp->gsm, tmp->buf + samples, (gsm_byte *) pvt->outbuf.c + datalen); - datalen += GSM_FRAME_LEN; + gsm_encode(tmp->gsm, tmp->buf + samples, (gsm_byte *) pvt->outbuf.c); samples += GSM_SAMPLES; pvt->samples -= GSM_SAMPLES; + + current = ast_trans_frameout(pvt, GSM_FRAME_LEN, GSM_SAMPLES); + if (!current) { + continue; + } else if (last) { + AST_LIST_NEXT(last, frame_list) = current; + } else { + result = current; + } + last = current; } /* Move the data at the end of the buffer to the front */ - if (pvt->samples) + if (samples) { memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2); + } - return ast_trans_frameout(pvt, datalen, samples); + return result; } static void gsm_destroy_stuff(struct ast_trans_pvt *pvt) diff --git a/codecs/codec_ilbc.c b/codecs/codec_ilbc.c index 8247f2473..3e480e8fd 100644 --- a/codecs/codec_ilbc.c +++ b/codecs/codec_ilbc.c @@ -37,6 +37,7 @@ ASTERISK_REGISTER_FILE() #include "asterisk/translate.h" #include "asterisk/module.h" #include "asterisk/utils.h" +#include "asterisk/linkedlists.h" #ifdef ILBC_WEBRTC #include <ilbc.h> @@ -150,31 +151,40 @@ static int lintoilbc_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) static struct ast_frame *lintoilbc_frameout(struct ast_trans_pvt *pvt) { struct ilbc_coder_pvt *tmp = pvt->pvt; - int datalen = 0; - int samples = 0; + struct ast_frame *result = NULL; + struct ast_frame *last = NULL; + int samples = 0; /* output samples */ - /* We can't work on anything less than a frame in size */ - if (pvt->samples < ILBC_SAMPLES) - return NULL; while (pvt->samples >= ILBC_SAMPLES) { + struct ast_frame *current; ilbc_block tmpf[ILBC_SAMPLES]; int i; /* Encode a frame of data */ for (i = 0 ; i < ILBC_SAMPLES ; i++) tmpf[i] = tmp->buf[samples + i]; - iLBC_encode( (ilbc_bytes*)pvt->outbuf.BUF_TYPE + datalen, tmpf, &tmp->enc); + iLBC_encode((ilbc_bytes *) pvt->outbuf.BUF_TYPE, tmpf, &tmp->enc); - datalen += ILBC_FRAME_LEN; samples += ILBC_SAMPLES; pvt->samples -= ILBC_SAMPLES; + + current = ast_trans_frameout(pvt, ILBC_FRAME_LEN, ILBC_SAMPLES); + if (!current) { + continue; + } else if (last) { + AST_LIST_NEXT(last, frame_list) = current; + } else { + result = current; + } + last = current; } /* Move the data at the end of the buffer to the front */ - if (pvt->samples) + if (samples) { memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2); + } - return ast_trans_frameout(pvt, datalen, samples); + return result; } static struct ast_translator ilbctolin = { diff --git a/codecs/codec_lpc10.c b/codecs/codec_lpc10.c index 49df8f753..e6dcf8c99 100644 --- a/codecs/codec_lpc10.c +++ b/codecs/codec_lpc10.c @@ -39,6 +39,7 @@ ASTERISK_REGISTER_FILE() #include "asterisk/config.h" #include "asterisk/module.h" #include "asterisk/utils.h" +#include "asterisk/linkedlists.h" #include "lpc10/lpc10.h" @@ -160,31 +161,45 @@ static int lintolpc10_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) static struct ast_frame *lintolpc10_frameout(struct ast_trans_pvt *pvt) { struct lpc10_coder_pvt *tmp = pvt->pvt; - int x; - int datalen = 0; /* output frame */ - int samples = 0; /* output samples */ - float tmpbuf[LPC10_SAMPLES_PER_FRAME]; - INT32 bits[LPC10_BITS_IN_COMPRESSED_FRAME]; /* XXX what ??? */ - /* We can't work on anything less than a frame in size */ - if (pvt->samples < LPC10_SAMPLES_PER_FRAME) - return NULL; - while (pvt->samples >= LPC10_SAMPLES_PER_FRAME) { + struct ast_frame *result = NULL; + struct ast_frame *last = NULL; + int samples = 0; /* output samples */ + + while (pvt->samples >= LPC10_SAMPLES_PER_FRAME) { + struct ast_frame *current; + float tmpbuf[LPC10_SAMPLES_PER_FRAME]; + INT32 bits[LPC10_BITS_IN_COMPRESSED_FRAME]; /* XXX what ??? */ + int x; + /* Encode a frame of data */ for (x=0;x<LPC10_SAMPLES_PER_FRAME;x++) tmpbuf[x] = (float)tmp->buf[x + samples] / 32768.0; lpc10_encode(tmpbuf, bits, tmp->lpc10.enc); - build_bits(pvt->outbuf.uc + datalen, bits); - datalen += LPC10_BYTES_IN_COMPRESSED_FRAME; + build_bits(pvt->outbuf.uc, bits); + samples += LPC10_SAMPLES_PER_FRAME; pvt->samples -= LPC10_SAMPLES_PER_FRAME; /* Use one of the two left over bits to record if this is a 22 or 23 ms frame... important for IAX use */ tmp->longer = 1 - tmp->longer; + + current = ast_trans_frameout(pvt, LPC10_BYTES_IN_COMPRESSED_FRAME, LPC10_SAMPLES_PER_FRAME); + if (!current) { + continue; + } else if (last) { + AST_LIST_NEXT(last, frame_list) = current; + } else { + result = current; + } + last = current; } + /* Move the data at the end of the buffer to the front */ - if (pvt->samples) + if (samples) { memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2); - return ast_trans_frameout(pvt, datalen, samples); + } + + return result; } diff --git a/codecs/codec_speex.c b/codecs/codec_speex.c index c61f7c4f4..ca48eae62 100644 --- a/codecs/codec_speex.c +++ b/codecs/codec_speex.c @@ -54,6 +54,8 @@ ASTERISK_REGISTER_FILE() #include "asterisk/module.h" #include "asterisk/config.h" #include "asterisk/utils.h" +#include "asterisk/frame.h" +#include "asterisk/linkedlists.h" /* codec variables */ static int quality = 3; @@ -259,15 +261,16 @@ static int lintospeex_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt) { struct speex_coder_pvt *tmp = pvt->pvt; - int is_speech=1; - int datalen = 0; /* output bytes */ - int samples = 0; /* output samples */ + struct ast_frame *result = NULL; + struct ast_frame *last = NULL; + int samples = 0; /* output samples */ - /* We can't work on anything less than a frame in size */ - if (pvt->samples < tmp->framesize) - return NULL; - speex_bits_reset(&tmp->bits); while (pvt->samples >= tmp->framesize) { + struct ast_frame *current; + int is_speech = 1; + + speex_bits_reset(&tmp->bits); + #ifdef _SPEEX_TYPES_H /* Preprocess audio */ if (preproc) @@ -293,18 +296,18 @@ static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt) #endif samples += tmp->framesize; pvt->samples -= tmp->framesize; - } - /* Move the data at the end of the buffer to the front */ - if (pvt->samples) - memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2); - - /* Use AST_FRAME_CNG to signify the start of any silence period */ - if (is_speech) { - tmp->silent_state = 0; - } else { - if (tmp->silent_state) { - return NULL; + /* Use AST_FRAME_CNG to signify the start of any silence period */ + if (is_speech) { + int datalen = 0; /* output bytes */ + + tmp->silent_state = 0; + /* Terminate bit stream */ + speex_bits_pack(&tmp->bits, 15, 5); + datalen = speex_bits_write(&tmp->bits, pvt->outbuf.c, pvt->t->buf_size); + current = ast_trans_frameout(pvt, datalen, tmp->framesize); + } else if (tmp->silent_state) { + current = NULL; } else { struct ast_frame frm = { .frametype = AST_FRAME_CNG, @@ -320,14 +323,25 @@ static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt) tmp->silent_state = 1; /* XXX what now ? format etc... */ - return ast_frisolate(&frm); + current = ast_frisolate(&frm); } + + if (!current) { + continue; + } else if (last) { + AST_LIST_NEXT(last, frame_list) = current; + } else { + result = current; + } + last = current; + } + + /* Move the data at the end of the buffer to the front */ + if (samples) { + memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2); } - /* Terminate bit stream */ - speex_bits_pack(&tmp->bits, 15, 5); - datalen = speex_bits_write(&tmp->bits, pvt->outbuf.c, pvt->t->buf_size); - return ast_trans_frameout(pvt, datalen, samples); + return result; } static void speextolin_destroy(struct ast_trans_pvt *arg) diff --git a/main/translate.c b/main/translate.c index f13ecf456..334d3b550 100644 --- a/main/translate.c +++ b/main/translate.c @@ -44,6 +44,7 @@ ASTERISK_REGISTER_FILE() #include "asterisk/cli.h" #include "asterisk/term.h" #include "asterisk/format.h" +#include "asterisk/linkedlists.h" /*! \todo * TODO: sample frames for each supported input format. @@ -547,7 +548,12 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, } delivery = f->delivery; for (out = f; out && p ; p = p->next) { - framein(p, out); + struct ast_frame *current = out; + + do { + framein(p, current); + current = AST_LIST_NEXT(current, frame_list); + } while (current); if (out != f) { ast_frfree(out); } @@ -556,22 +562,33 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, if (out) { /* we have a frame, play with times */ if (!ast_tvzero(delivery)) { - /* Regenerate prediction after a discontinuity */ - if (ast_tvzero(path->nextout)) { - path->nextout = ast_tvnow(); - } + struct ast_frame *current = out; - /* Use next predicted outgoing timestamp */ - out->delivery = path->nextout; + do { + /* Regenerate prediction after a discontinuity */ + if (ast_tvzero(path->nextout)) { + path->nextout = ast_tvnow(); + } - /* Predict next outgoing timestamp from samples in this - frame. */ - path->nextout = ast_tvadd(path->nextout, ast_samp2tv( - out->samples, ast_format_get_sample_rate(out->subclass.format))); - if (f->samples != out->samples && ast_test_flag(out, AST_FRFLAG_HAS_TIMING_INFO)) { - ast_debug(4, "Sample size different %d vs %d\n", f->samples, out->samples); - ast_clear_flag(out, AST_FRFLAG_HAS_TIMING_INFO); - } + /* Use next predicted outgoing timestamp */ + current->delivery = path->nextout; + + /* Invalidate prediction if we're entering a silence period */ + if (current->frametype == AST_FRAME_CNG) { + path->nextout = ast_tv(0, 0); + /* Predict next outgoing timestamp from samples in this + frame. */ + } else { + path->nextout = ast_tvadd(path->nextout, ast_samp2tv( + current->samples, ast_format_get_sample_rate(current->subclass.format))); + } + + if (f->samples != current->samples && ast_test_flag(current, AST_FRFLAG_HAS_TIMING_INFO)) { + ast_debug(4, "Sample size different %d vs %d\n", f->samples, current->samples); + ast_clear_flag(current, AST_FRFLAG_HAS_TIMING_INFO); + } + current = AST_LIST_NEXT(current, frame_list); + } while (current); } else { out->delivery = ast_tv(0, 0); ast_set2_flag(out, has_timing_info, AST_FRFLAG_HAS_TIMING_INFO); @@ -580,10 +597,10 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, out->len = len; out->seqno = seqno; } - } - /* Invalidate prediction if we're entering a silence period */ - if (out->frametype == AST_FRAME_CNG) { - path->nextout = ast_tv(0, 0); + /* Invalidate prediction if we're entering a silence period */ + if (out->frametype == AST_FRAME_CNG) { + path->nextout = ast_tv(0, 0); + } } } if (consume) { |