[Asterisk-code-review] transcode: Fix transcoding while different in frame size. (asterisk[13])
Alexander Traud
asteriskteam at digium.com
Fri Aug 28 15:50:16 CDT 2015
Alexander Traud has uploaded a new change for review.
https://gerrit.asterisk.org/1155
Change subject: transcode: Fix transcoding while different in frame size.
......................................................................
transcode: Fix transcoding while different in frame size.
When Asterisk transcoded 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, 111 insertions(+), 81 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/55/1155/1
diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c
index 8cb4943..cc9aa46 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 +140,30 @@
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;
- /* 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 = NULL;
+
/* Encode a frame of data */
- gsm_encode(tmp->gsm, tmp->buf + samples, (gsm_byte *) pvt->outbuf.c + datalen);
- datalen += GSM_FRAME_LEN;
- samples += GSM_SAMPLES;
+ gsm_encode(tmp->gsm, tmp->buf, (gsm_byte *) pvt->outbuf.c);
+
+ /* Move the data at the end of the buffer to the front */
pvt->samples -= GSM_SAMPLES;
+ if (pvt->samples) {
+ memmove(tmp->buf, tmp->buf + GSM_SAMPLES, pvt->samples * 2);
+ }
+
+ current = ast_trans_frameout(pvt, GSM_FRAME_LEN, GSM_SAMPLES);
+ 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)
- 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 af23b90..dbb6fe2 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 +151,34 @@
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;
- /* 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 = NULL;
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);
+ tmpf[i] = tmp->buf[i];
+ iLBC_encode( (ilbc_bytes*)pvt->outbuf.BUF_TYPE, tmpf, &tmp->enc);
- datalen += ILBC_FRAME_LEN;
- samples += ILBC_SAMPLES;
+ /* Move the data at the end of the buffer to the front */
pvt->samples -= ILBC_SAMPLES;
+ if (pvt->samples) {
+ memmove(tmp->buf, tmp->buf + ILBC_SAMPLES, pvt->samples * 2);
+ }
+
+ current = ast_trans_frameout(pvt, ILBC_FRAME_LEN, ILBC_SAMPLES);
+ 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)
- 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 ca2eb8e..ed0945f 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"
@@ -161,30 +162,39 @@
{
struct lpc10_coder_pvt *tmp = pvt->pvt;
int x;
- int datalen = 0; /* output frame */
- int samples = 0; /* output samples */
+ struct ast_frame *result = NULL;
+ struct ast_frame *last = NULL;
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) {
+
+ while (pvt->samples >= LPC10_SAMPLES_PER_FRAME) {
+ struct ast_frame *current = NULL;
+
/* Encode a frame of data */
for (x=0;x<LPC10_SAMPLES_PER_FRAME;x++)
- tmpbuf[x] = (float)tmp->buf[x + samples] / 32768.0;
+ tmpbuf[x] = (float)tmp->buf[x] / 32768.0;
lpc10_encode(tmpbuf, bits, tmp->lpc10.enc);
- build_bits(pvt->outbuf.uc + datalen, bits);
- datalen += LPC10_BYTES_IN_COMPRESSED_FRAME;
- samples += LPC10_SAMPLES_PER_FRAME;
+ build_bits(pvt->outbuf.uc, bits);
+
+ /* Move the data at the end of the buffer to the front */
pvt->samples -= LPC10_SAMPLES_PER_FRAME;
+ if (pvt->samples) {
+ memmove(tmp->buf, tmp->buf + LPC10_SAMPLES_PER_FRAME, pvt->samples * 2);
+ }
+
/* 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 (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)
- 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 c91070d..4ef191d 100644
--- a/codecs/codec_speex.c
+++ b/codecs/codec_speex.c
@@ -54,6 +54,7 @@
#include "asterisk/module.h"
#include "asterisk/config.h"
#include "asterisk/utils.h"
+#include "asterisk/linkedlists.h"
/* codec variables */
static int quality = 3;
@@ -259,23 +260,24 @@
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;
- /* 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) {
+ int is_speech=1;
+ int datalen = 0; /* output bytes */
+ struct ast_frame *current = NULL;
+
+ speex_bits_reset(&tmp->bits);
+
#ifdef _SPEEX_TYPES_H
/* Preprocess audio */
if (preproc)
- is_speech = speex_preprocess(tmp->pp, tmp->buf + samples, NULL);
+ is_speech = speex_preprocess(tmp->pp, tmp->buf, NULL);
/* Encode a frame of data */
if (is_speech) {
/* If DTX enabled speex_encode returns 0 during silence */
- is_speech = speex_encode_int(tmp->speex, tmp->buf + samples, &tmp->bits) || !dtx;
+ is_speech = speex_encode_int(tmp->speex, tmp->buf, &tmp->bits) || !dtx;
} else {
/* 5 zeros interpreted by Speex as silence (submode 0) */
speex_bits_pack(&tmp->bits, 0, 5);
@@ -286,26 +288,25 @@
int x;
/* Convert to floating point */
for (x = 0; x < tmp->framesize; x++)
- fbuf[x] = tmp->buf[samples + x];
+ fbuf[x] = tmp->buf[x];
/* Encode a frame of data */
is_speech = speex_encode(tmp->speex, fbuf, &tmp->bits) || !dtx;
}
#endif
- samples += tmp->framesize;
+ /* Move the data at the end of the buffer to the front */
pvt->samples -= tmp->framesize;
- }
+ if (pvt->samples) {
+ memmove(tmp->buf, tmp->buf + tmp->framesize, pvt->samples * 2);
+ }
- /* 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;
- } else {
+ /* Use AST_FRAME_CNG to signify the start of any silence period */
+ if (is_speech) {
+ 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) {
struct ast_frame frm = {
.frametype = AST_FRAME_CNG,
.src = pvt->t->name,
@@ -320,14 +321,17 @@
tmp->silent_state = 1;
/* XXX what now ? format etc... */
- return ast_frisolate(&frm);
+ current = ast_frisolate(&frm);
}
- }
- /* 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);
+ if (last) {
+ AST_LIST_NEXT(last, frame_list) = current;
+ } else {
+ result = current;
+ }
+ last = current;
+ }
+ return result;
}
static void speextolin_destroy(struct ast_trans_pvt *arg)
diff --git a/main/translate.c b/main/translate.c
index 2b11d7b..128cc73 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.
@@ -556,21 +557,26 @@
if (out) {
/* we have a frame, play with times */
if (!ast_tvzero(delivery)) {
+ struct ast_frame *current = out;
+
/* Regenerate prediction after a discontinuity */
if (ast_tvzero(path->nextout)) {
path->nextout = ast_tvnow();
}
- /* Use next predicted outgoing timestamp */
- out->delivery = path->nextout;
+ while (current) {
+ /* Use next predicted outgoing timestamp */
+ current->delivery = path->nextout;
- /* 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);
+ /* Predict next outgoing timestamp from samples in this
+ frame. */
+ 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);
}
} else {
out->delivery = ast_tv(0, 0);
--
To view, visit https://gerrit.asterisk.org/1155
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I2e229569d73191d66a4e43fef35432db24000212
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
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