[asterisk-commits] translate: Fix transcoding while different in frame size. (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 28 16:24:19 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: translate: Fix transcoding while different in frame size.
......................................................................
translate: Fix transcoding while different in frame size.
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
---
M codecs/codec_gsm.c
M codecs/codec_ilbc.c
M codecs/codec_lpc10.c
M codecs/codec_speex.c
M main/translate.c
5 files changed, 139 insertions(+), 72 deletions(-)
Approvals:
Richard Mudgett: Looks good to me, but someone else must approve
Matt Jordan: Looks good to me, approved; Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c
index 4660048..f80c955 100644
--- a/codecs/codec_gsm.c
+++ b/codecs/codec_gsm.c
@@ -39,6 +39,7 @@
#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 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 8247f24..3e480e8 100644
--- a/codecs/codec_ilbc.c
+++ b/codecs/codec_ilbc.c
@@ -37,6 +37,7 @@
#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 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 49df8f7..e6dcf8c 100644
--- a/codecs/codec_lpc10.c
+++ b/codecs/codec_lpc10.c
@@ -39,6 +39,7 @@
#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 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 c61f7c4..ca48eae 100644
--- a/codecs/codec_speex.c
+++ b/codecs/codec_speex.c
@@ -54,6 +54,8 @@
#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 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 @@
#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) {
+ int datalen = 0; /* output bytes */
- /* 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;
+ 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 @@
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;
}
- /* 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);
+ /* Move the data at the end of the buffer to the front */
+ if (samples) {
+ memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
+ }
+
+ return result;
}
static void speextolin_destroy(struct ast_trans_pvt *arg)
diff --git a/main/translate.c b/main/translate.c
index f13ecf4..334d3b5 100644
--- a/main/translate.c
+++ b/main/translate.c
@@ -44,6 +44,7 @@
#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 @@
}
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 @@
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 @@
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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Gerrit-MessageType: merged
Gerrit-Change-Id: I2e229569d73191d66a4e43fef35432db24000212
Gerrit-PatchSet: 12
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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