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translate: Fix transcoding while different in frame size.

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When Asterisk translates between codecs, each with a different frame size (for
example between iLBC 30 and Speex-WB), too large frames were created by
ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
length, creating several frames when necessary. Affects all transcoding modules
which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.

ASTERISK-25353 #close

Change-Id: I2e229569d73191d66a4e43fef35432db24000212
changes/54/1154/12
Alexander Traud 10 years ago
parent 229b95d253
commit 077adf48b8
5 changed files with 140 additions and 73 deletions
Show Stats Download Patch File Download Diff File

29
codecs/codec_gsm.c
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@ -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)

28
codecs/codec_ilbc.c
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@ -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 = {

41
codecs/codec_lpc10.c
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@ -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;
}

60
codecs/codec_speex.c
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@ -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)

55
main/translate.c
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@ -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) {

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