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translate.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <[email protected]>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Translate via the use of pseudo channels
*
* \author Mark Spencer <[email protected]>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <sys/time.h>
#include <sys/resource.h>
#include <math.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/translate.h"
#include "asterisk/module.h"
#include "asterisk/frame.h"
#include "asterisk/sched.h"
#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.
* We build this on the fly, by taking an SLIN frame and using
* the existing converter to play with it.
*/
/*! max sample recalc */
#define MAX_RECALC 1000
/*! \brief the list of translators */
static AST_RWLIST_HEAD_STATIC(translators, ast_translator);
struct translator_path {
struct ast_translator *step; /*!< Next step translator */
uint32_t table_cost; /*!< Complete table cost to destination */
uint8_t multistep; /*!< Multiple conversions required for this translation */
};
/*!
* \brief a matrix that, for any pair of supported formats,
* indicates the total cost of translation and the first step.
* The full path can be reconstructed iterating on the matrix
* until step->dstfmt == desired_format.
*
* Array indexes are 'src' and 'dest', in that order.
*
* Note: the lock in the 'translators' list is also used to protect
* this structure.
*/
static struct translator_path **__matrix;
/*!
* \brief table for converting index to format values.
*
* \note this table is protected by the table_lock.
*/
static unsigned int *__indextable;
/*! protects the __indextable for resizing */
static ast_rwlock_t tablelock;
/* index size starts at this*/
#define INIT_INDEX 32
/* index size grows by this as necessary */
#define GROW_INDEX 16
/*! the current largest index used by the __matrix and __indextable arrays*/
static int cur_max_index;
/*! the largest index that can be used in either the __indextable or __matrix before resize must occur */
static int index_size;
static void matrix_rebuild(int samples);
/*!
* \internal
* \brief converts codec id to index value.
*/
static int codec_to_index(unsigned int id)
{
int x;
ast_rwlock_rdlock(&tablelock);
for (x = 0; x < cur_max_index; x++) {
if (__indextable[x] == id) {
/* format already exists in index2format table */
ast_rwlock_unlock(&tablelock);
return x;
}
}
ast_rwlock_unlock(&tablelock);
return -1; /* not found */
}
/*!
* \internal
* \brief converts codec to index value.
*/
static int codec2index(struct ast_codec *codec)
{
return codec_to_index(codec->id);
}
/*!
* \internal
* \brief converts format to codec index value.
*/
static int format2index(struct ast_format *format)
{
return codec_to_index(ast_format_get_codec_id(format));
}
/*!
* \internal
* \brief add a new codec to the matrix and index table structures.
*
* \note it is perfectly safe to call this on codecs already indexed.
*
* \retval 0 success
* \retval -1 matrix and index table need to be resized
*/
static int add_codec2index(struct ast_codec *codec)
{
if (codec2index(codec) != -1) {
/* format is already already indexed */
return 0;
}
ast_rwlock_wrlock(&tablelock);
if (cur_max_index == (index_size)) {
ast_rwlock_unlock(&tablelock);
return -1; /* hit max length */
}
__indextable[cur_max_index] = codec->id;
cur_max_index++;
ast_rwlock_unlock(&tablelock);
return 0;
}
/*!
* \internal
* \brief converts index value back to codec
*/
static struct ast_codec *index2codec(int index)
{
struct ast_codec *codec;
if (index >= cur_max_index) {
return 0;
}
ast_rwlock_rdlock(&tablelock);
codec = ast_codec_get_by_id(__indextable[index]);
ast_rwlock_unlock(&tablelock);
return codec;
}
/*!
* \internal
* \brief resize both the matrix and index table so they can represent
* more translators
*
* \note _NO_ locks can be held prior to calling this function
*
* \retval 0 success
* \retval -1 failure. Old matrix and index table can still be used though
*/
static int matrix_resize(int init)
{
struct translator_path **tmp_matrix = NULL;
unsigned int *tmp_table = NULL;
int old_index;
int x;
AST_RWLIST_WRLOCK(&translators);
ast_rwlock_wrlock(&tablelock);
old_index = index_size;
if (init) {
index_size += INIT_INDEX;
} else {
index_size += GROW_INDEX;
}
/* make new 2d array of translator_path structures */
if (!(tmp_matrix = ast_calloc(1, sizeof(struct translator_path *) * (index_size)))) {
goto resize_cleanup;
}
for (x = 0; x < index_size; x++) {
if (!(tmp_matrix[x] = ast_calloc(1, sizeof(struct translator_path) * (index_size)))) {
goto resize_cleanup;
}
}
/* make new index table */
if (!(tmp_table = ast_calloc(1, sizeof(unsigned int) * index_size))) {
goto resize_cleanup;
}
/* if everything went well this far, free the old and use the new */
if (!init) {
for (x = 0; x < old_index; x++) {
ast_free(__matrix[x]);
}
ast_free(__matrix);
memcpy(tmp_table, __indextable, sizeof(unsigned int) * old_index);
ast_free(__indextable);
}
/* now copy them over */
__matrix = tmp_matrix;
__indextable = tmp_table;
matrix_rebuild(0);
ast_rwlock_unlock(&tablelock);
AST_RWLIST_UNLOCK(&translators);
return 0;
resize_cleanup:
ast_rwlock_unlock(&tablelock);
AST_RWLIST_UNLOCK(&translators);
if (tmp_matrix) {
for (x = 0; x < index_size; x++) {
ast_free(tmp_matrix[x]);
}
ast_free(tmp_matrix);
}
ast_free(tmp_table);
return -1;
}
/*!
* \internal
* \brief reinitialize the __matrix during matrix rebuild
*
* \note must be protected by the translators list lock
*/
static void matrix_clear(void)
{
int x;
for (x = 0; x < index_size; x++) {
memset(__matrix[x], '\0', sizeof(struct translator_path) * (index_size));
}
}
/*!
* \internal
* \brief get a matrix entry
*
* \note This function must be protected by the translators list lock
*/
static struct translator_path *matrix_get(unsigned int x, unsigned int y)
{
return __matrix[x] + y;
}
/*
* wrappers around the translator routines.
*/
static void destroy(struct ast_trans_pvt *pvt)
{
struct ast_translator *t = pvt->t;
if (t->destroy) {
t->destroy(pvt);
}
ao2_cleanup(pvt->f.subclass.format);
if (pvt->explicit_dst) {
ao2_ref(pvt->explicit_dst, -1);
pvt->explicit_dst = NULL;
}
ast_free(pvt);
ast_module_unref(t->module);
}
/*!
* \brief Allocate the descriptor, required outbuf space,
* and possibly desc.
*/
static struct ast_trans_pvt *newpvt(struct ast_translator *t, struct ast_format *explicit_dst)
{
struct ast_trans_pvt *pvt;
int len;
char *ofs;
/*
* compute the required size adding private descriptor,
* buffer, AST_FRIENDLY_OFFSET.
*/
len = sizeof(*pvt) + t->desc_size;
if (t->buf_size)
len += AST_FRIENDLY_OFFSET + t->buf_size;
pvt = ast_calloc(1, len);
if (!pvt) {
return NULL;
}
pvt->t = t;
ofs = (char *)(pvt + 1); /* pointer to data space */
if (t->desc_size) { /* first comes the descriptor */
pvt->pvt = ofs;
ofs += t->desc_size;
}
if (t->buf_size) {/* finally buffer and header */
pvt->outbuf.c = ofs + AST_FRIENDLY_OFFSET;
}
/*
* If the format has an attribute module, explicit_dst includes the (joined)
* result of the SDP negotiation. For example with the Opus Codec, the format
* knows whether both parties want to do forward-error correction (FEC).
*/
pvt->explicit_dst = ao2_bump(explicit_dst);
ast_module_ref(t->module);
/* call local init routine, if present */
if (t->newpvt && t->newpvt(pvt)) {
ast_free(pvt);
ast_module_unref(t->module);
return NULL;
}
/* Setup normal static translation frame. */
pvt->f.frametype = AST_FRAME_VOICE;
pvt->f.mallocd = 0;
pvt->f.offset = AST_FRIENDLY_OFFSET;
pvt->f.src = pvt->t->name;
pvt->f.data.ptr = pvt->outbuf.c;
/*
* If the translator has not provided a format
* A) use the joined one,
* B) use the cached one, or
* C) create one.
*/
if (!pvt->f.subclass.format) {
pvt->f.subclass.format = ao2_bump(pvt->explicit_dst);
if (!pvt->f.subclass.format && !ast_strlen_zero(pvt->t->format)) {
pvt->f.subclass.format = ast_format_cache_get(pvt->t->format);
}
if (!pvt->f.subclass.format) {
struct ast_codec *codec = ast_codec_get(t->dst_codec.name,
t->dst_codec.type, t->dst_codec.sample_rate);
if (!codec) {
ast_log(LOG_ERROR, "Unable to get destination codec\n");
destroy(pvt);
return NULL;
}
pvt->f.subclass.format = ast_format_create(codec);
ao2_ref(codec, -1);
}
if (!pvt->f.subclass.format) {
ast_log(LOG_ERROR, "Unable to create format\n");
destroy(pvt);
return NULL;
}
}
return pvt;
}
/*! \brief framein wrapper, deals with bound checks. */
static int framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
/* Copy the last in jb timing info to the pvt */
ast_copy_flags(&pvt->f, f, AST_FRFLAG_HAS_TIMING_INFO);
pvt->f.ts = f->ts;
pvt->f.len = f->len;
pvt->f.seqno = f->seqno;
if (f->samples == 0) {
/* Do not log empty audio frame */
if (!f->src || strcasecmp(f->src, "ast_prod")) {
ast_log(LOG_WARNING, "no samples for %s\n", pvt->t->name);
}
}
if (pvt->t->buffer_samples) { /* do not pass empty frames to callback */
int src_srate = pvt->t->src_codec.sample_rate;
int dst_srate = pvt->t->dst_codec.sample_rate;
ast_assert(src_srate > 0);
if (f->datalen == 0) { /* perform native PLC if available */
/* If the codec has native PLC, then do that */
if (!pvt->t->native_plc)
return 0;
}
if (pvt->samples + (f->samples * dst_srate / src_srate) > pvt->t->buffer_samples) {
ast_log(LOG_WARNING, "Out of buffer space\n");
return -1;
}
}
/* we require a framein routine, wouldn't know how to do
* it otherwise.
*/
return pvt->t->framein(pvt, f);
}
/*! \brief generic frameout routine.
* If samples and datalen are 0, take whatever is in pvt
* and reset them, otherwise take the values in the caller and
* leave alone the pvt values.
*/
struct ast_frame *ast_trans_frameout(struct ast_trans_pvt *pvt,
int datalen, int samples)
{
struct ast_frame *f = &pvt->f;
if (samples) {
f->samples = samples;
} else {
if (pvt->samples == 0) {
return NULL;
}
f->samples = pvt->samples;
pvt->samples = 0;
}
if (datalen) {
f->datalen = datalen;
f->data.ptr = pvt->outbuf.c;
} else {
f->datalen = pvt->datalen;
if (!f->datalen) {
f->data.ptr = NULL;
} else {
f->data.ptr = pvt->outbuf.c;
}
pvt->datalen = 0;
}
return ast_frisolate(f);
}
static struct ast_frame *default_frameout(struct ast_trans_pvt *pvt)
{
return ast_trans_frameout(pvt, 0, 0);
}
/* end of callback wrappers and helpers */
void ast_translator_free_path(struct ast_trans_pvt *p)
{
struct ast_trans_pvt *pn = p;
while ( (p = pn) ) {
pn = p->next;
destroy(p);
}
}
/*! \brief Build a chain of translators based upon the given source and dest formats */
struct ast_trans_pvt *ast_translator_build_path(struct ast_format *dst, struct ast_format *src)
{
struct ast_trans_pvt *head = NULL, *tail = NULL;
int src_index, dst_index;
src_index = format2index(src);
dst_index = format2index(dst);
if (src_index < 0 || dst_index < 0) {
ast_log(LOG_WARNING, "No translator path: (%s codec is not valid)\n", src_index < 0 ? "starting" : "ending");
return NULL;
}
AST_RWLIST_RDLOCK(&translators);
while (src_index != dst_index) {
struct ast_trans_pvt *cur;
struct ast_format *explicit_dst = NULL;
struct ast_translator *t = matrix_get(src_index, dst_index)->step;
if (!t) {
ast_log(LOG_WARNING, "No translator path from %s to %s\n",
ast_format_get_name(src), ast_format_get_name(dst));
AST_RWLIST_UNLOCK(&translators);
ast_translator_free_path(head);
return NULL;
}
if ((t->dst_codec.sample_rate == ast_format_get_sample_rate(dst)) && (t->dst_codec.type == ast_format_get_type(dst))) {
explicit_dst = dst;
}
if (!(cur = newpvt(t, explicit_dst))) {
ast_log(LOG_WARNING, "Failed to build translator step from %s to %s\n",
ast_format_get_name(src), ast_format_get_name(dst));
ast_translator_free_path(head);
AST_RWLIST_UNLOCK(&translators);
return NULL;
}
if (!head) {
head = cur;
} else {
tail->next = cur;
}
tail = cur;
cur->nextin = cur->nextout = ast_tv(0, 0);
/* Keep going if this isn't the final destination */
src_index = cur->t->dst_fmt_index;
}
AST_RWLIST_UNLOCK(&translators);
return head;
}
static struct ast_frame *generate_interpolated_slin(struct ast_trans_pvt *p, struct ast_frame *f)
{
struct ast_frame res = { AST_FRAME_VOICE };
/*
* If we've gotten here then we should have an interpolated frame that was not handled
* by the translation codec. So create an interpolated frame in the appropriate format
* that was going to be written. This frame might be handled later by other resources.
* For instance, generic plc.
*
* Note, generic plc is currently only available for the format type 'slin' (8KHz only -
* The generic plc code appears to have been based around that). Generic plc is filled
* in later on frame write.
*/
if (!ast_opt_generic_plc || f->datalen != 0 ||
ast_format_cmp(p->explicit_dst, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
return NULL;
}
res.subclass.format = ast_format_cache_get_slin_by_rate(8000); /* ref bumped on dup */
res.samples = f->samples;
res.datalen = 0;
res.data.ptr = NULL;
res.offset = AST_FRIENDLY_OFFSET;
return ast_frdup(&res);
}
/*! \brief do the actual translation */
struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, int consume)
{
struct ast_trans_pvt *p = path;
struct ast_frame *out;
struct timeval delivery;
int has_timing_info;
long ts;
long len;
int seqno;
if (f->frametype == AST_FRAME_RTCP) {
/* Just pass the feedback to the right callback, if it exists.
* This "translation" does nothing so return a null frame. */
struct ast_trans_pvt *tp;
for (tp = p; tp; tp = tp->next) {
if (tp->t->feedback)
tp->t->feedback(tp, f);
}
return &ast_null_frame;
}
has_timing_info = ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO);
ts = f->ts;
len = f->len;
seqno = f->seqno;
if (!ast_tvzero(f->delivery)) {
if (!ast_tvzero(path->nextin)) {
/* Make sure this is in line with what we were expecting */
if (!ast_tveq(path->nextin, f->delivery)) {
/* The time has changed between what we expected and this
most recent time on the new packet. If we have a
valid prediction adjust our output time appropriately */
if (!ast_tvzero(path->nextout)) {
path->nextout = ast_tvadd(path->nextout,
ast_tvsub(f->delivery, path->nextin));
}
path->nextin = f->delivery;
}
} else {
/* This is our first pass. Make sure the timing looks good */
path->nextin = f->delivery;
path->nextout = f->delivery;
}
/* Predict next incoming sample */
path->nextin = ast_tvadd(path->nextin, ast_samp2tv(
f->samples, ast_format_get_sample_rate(f->subclass.format)));
}
delivery = f->delivery;
for (out = f; out && p ; p = p->next) {
struct ast_frame *current = out;
do {
framein(p, current);
current = AST_LIST_NEXT(current, frame_list);
} while (current);
if (out != f) {
ast_frfree(out);
}
out = p->t->frameout(p);
}
if (!out) {
out = generate_interpolated_slin(path, f);
}
if (out) {
/* we have a frame, play with times */
if (!ast_tvzero(delivery)) {
struct ast_frame *current = out;
do {
/* Regenerate prediction after a discontinuity */
if (ast_tvzero(path->nextout)) {
path->nextout = ast_tvnow();
}
/* 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);
if (has_timing_info) {
out->ts = ts;
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);
}
}
}
if (consume) {
ast_frfree(f);
}
return out;
}
/*!
* \internal
* \brief Compute the computational cost of a single translation step.
*
* \note This function is only used to decide which translation path to
* use between two translators with identical src and dst formats. Computational
* cost acts only as a tie breaker. This is done so hardware translators
* can naturally have precedence over software translators.
*/
static void generate_computational_cost(struct ast_translator *t, int seconds)
{
int num_samples = 0;
struct ast_trans_pvt *pvt;
struct rusage start;
struct rusage end;
int cost;
int out_rate = t->dst_codec.sample_rate;
if (!seconds) {
seconds = 1;
}
/* If they don't make samples, give them a terrible score */
if (!t->sample) {
ast_debug(3, "Translator '%s' does not produce sample frames.\n", t->name);
t->comp_cost = 999999;
return;
}
pvt = newpvt(t, NULL);
if (!pvt) {
ast_log(LOG_WARNING, "Translator '%s' appears to be broken and will probably fail.\n", t->name);
t->comp_cost = 999999;
return;
}
getrusage(RUSAGE_SELF, &start);
/* Call the encoder until we've processed the required number of samples */
while (num_samples < seconds * out_rate) {
struct ast_frame *f = t->sample();
if (!f) {
ast_log(LOG_WARNING, "Translator '%s' failed to produce a sample frame.\n", t->name);
destroy(pvt);
t->comp_cost = 999999;
return;
}
framein(pvt, f);
ast_frfree(f);
while ((f = t->frameout(pvt))) {
num_samples += f->samples;
ast_frfree(f);
}
}
getrusage(RUSAGE_SELF, &end);
cost = ((end.ru_utime.tv_sec - start.ru_utime.tv_sec) * 1000000) + end.ru_utime.tv_usec - start.ru_utime.tv_usec;
cost += ((end.ru_stime.tv_sec - start.ru_stime.tv_sec) * 1000000) + end.ru_stime.tv_usec - start.ru_stime.tv_usec;
destroy(pvt);
t->comp_cost = cost / seconds;
if (!t->comp_cost) {
t->comp_cost = 1;
}
}
/*!
* \internal
*
* \brief If no table cost value was pre set by the translator. An attempt is made to
* automatically generate that cost value from the cost table based on our src and
* dst formats.
*
* \note This function allows older translators built before the translation cost
* changed away from using onely computational time to continue to be registered
* correctly. It is expected that translators built after the introduction of this
* function will manually assign their own table cost value.
*
* \note This function is safe to use on any audio formats that used to be defined in the
* first 64 bits of the old bit field codec representation.
*
* \return Table Cost value greater than 0.
* \retval 0 on error.
*/
static int generate_table_cost(struct ast_codec *src, struct ast_codec *dst)
{
int src_rate = src->sample_rate;
int src_ll = 0;
int dst_rate = dst->sample_rate;
int dst_ll = 0;
if ((src->type != AST_MEDIA_TYPE_AUDIO) ||
(dst->type != AST_MEDIA_TYPE_AUDIO)) {
/* This method of generating table cost is limited to audio.
* Translators for media other than audio must manually set their
* table cost. */
return 0;
}
src_ll = !strcmp(src->name, "slin");
dst_ll = !strcmp(dst->name, "slin");
if (src_ll) {
if (dst_ll && (src_rate == dst_rate)) {
return AST_TRANS_COST_LL_LL_ORIGSAMP;
} else if (!dst_ll && (src_rate == dst_rate)) {
return AST_TRANS_COST_LL_LY_ORIGSAMP;
} else if (dst_ll && (src_rate < dst_rate)) {
return AST_TRANS_COST_LL_LL_UPSAMP;
} else if (!dst_ll && (src_rate < dst_rate)) {
return AST_TRANS_COST_LL_LY_UPSAMP;
} else if (dst_ll && (src_rate > dst_rate)) {
return AST_TRANS_COST_LL_LL_DOWNSAMP;
} else if (!dst_ll && (src_rate > dst_rate)) {
return AST_TRANS_COST_LL_LY_DOWNSAMP;
} else {
return AST_TRANS_COST_LL_UNKNOWN;
}
} else {
if (dst_ll && (src_rate == dst_rate)) {
return AST_TRANS_COST_LY_LL_ORIGSAMP;
} else if (!dst_ll && (src_rate == dst_rate)) {
return AST_TRANS_COST_LY_LY_ORIGSAMP;
} else if (dst_ll && (src_rate < dst_rate)) {
return AST_TRANS_COST_LY_LL_UPSAMP;
} else if (!dst_ll && (src_rate < dst_rate)) {
return AST_TRANS_COST_LY_LY_UPSAMP;
} else if (dst_ll && (src_rate > dst_rate)) {
return AST_TRANS_COST_LY_LL_DOWNSAMP;
} else if (!dst_ll && (src_rate > dst_rate)) {
return AST_TRANS_COST_LY_LY_DOWNSAMP;
} else {
return AST_TRANS_COST_LY_UNKNOWN;
}
}
}
/*!
* \brief rebuild a translation matrix.
* \note This function expects the list of translators to be locked
*/
static void matrix_rebuild(int samples)
{
struct ast_translator *t;
int newtablecost;
int x; /* source format index */
int y; /* intermediate format index */
int z; /* destination format index */
ast_debug(1, "Resetting translation matrix\n");
matrix_clear();
/* first, compute all direct costs */
AST_RWLIST_TRAVERSE(&translators, t, list) {
if (!t->active) {
continue;
}
x = t->src_fmt_index;
z = t->dst_fmt_index;
if (samples) {
generate_computational_cost(t, samples);
}
/* This new translator is the best choice if any of the below are true.
* 1. no translation path is set between x and z yet.
* 2. the new table cost is less.
* 3. the new computational cost is less. Computational cost is only used
* to break a tie between two identical translation paths.
*/
if (!matrix_get(x, z)->step ||
(t->table_cost < matrix_get(x, z)->step->table_cost) ||
(t->comp_cost < matrix_get(x, z)->step->comp_cost)) {
matrix_get(x, z)->step = t;
matrix_get(x, z)->table_cost = t->table_cost;
}
}
/*
* For each triple x, y, z of distinct formats, check if there is
* a path from x to z through y which is cheaper than what is
* currently known, and in case, update the matrix.
* Repeat until the matrix is stable.
*/
for (;;) {
int changed = 0;
for (x = 0; x < cur_max_index; x++) { /* source format */
for (y = 0; y < cur_max_index; y++) { /* intermediate format */
if (x == y) { /* skip ourselves */
continue;
}
for (z = 0; z < cur_max_index; z++) { /* dst format */
if ((z == x || z == y) || /* skip null conversions */
!matrix_get(x, y)->step || /* no path from x to y */
!matrix_get(y, z)->step) { /* no path from y to z */
continue;
}
/* calculate table cost from x->y->z */
newtablecost = matrix_get(x, y)->table_cost + matrix_get(y, z)->table_cost;
/* if no step already exists between x and z OR the new cost of using the intermediate
* step is cheaper, use this step. */
if (!matrix_get(x, z)->step || (newtablecost < matrix_get(x, z)->table_cost)) {
matrix_get(x, z)->step = matrix_get(x, y)->step;
matrix_get(x, z)->table_cost = newtablecost;
matrix_get(x, z)->multistep = 1;
changed++;
if (DEBUG_ATLEAST(10)) {
struct ast_codec *x_codec = index2codec(x);
struct ast_codec *y_codec = index2codec(y);
struct ast_codec *z_codec = index2codec(z);
ast_log(LOG_DEBUG,
"Discovered %u cost path from %s to %s, via %s\n",
matrix_get(x, z)->table_cost, x_codec->name,
y_codec->name, z_codec->name);
ao2_ref(x_codec, -1);
ao2_ref(y_codec, -1);
ao2_ref(z_codec, -1);
}
}
}
}
}
if (!changed) {
break;
}
}
}
static void codec_append_name(const struct ast_codec *codec, struct ast_str **buf)
{
if (codec) {
ast_str_append(buf, 0, "(%s@%u)", codec->name, codec->sample_rate);
} else {
ast_str_append(buf, 0, "(nothing)");
}
}
const char *ast_translate_path_to_str(struct ast_trans_pvt *p, struct ast_str **str)
{
if (!p || !p->t) {
return "";
}
ast_str_reset(*str);
codec_append_name(&p->t->src_codec, str);
while (p) {
ast_str_append(str, 0, "->");
codec_append_name(&p->t->dst_codec, str);
p = p->next;
}
return ast_str_buffer(*str);
}
static char *complete_trans_path_choice(const char *word)
{
int i = 1;
int wordlen = strlen(word);
struct ast_codec *codec;
while ((codec = ast_codec_get_by_id(i))) {
++i;
if (codec->type != AST_MEDIA_TYPE_AUDIO) {
ao2_ref(codec, -1);
continue;
}
if (!strncasecmp(word, codec->name, wordlen)) {
if (ast_cli_completion_add(ast_strdup(codec->name))) {
ao2_ref(codec, -1);
break;
}
}
ao2_ref(codec, -1);
}
return NULL;
}
static void handle_cli_recalc(struct ast_cli_args *a)
{
int time = a->argv[4] ? atoi(a->argv[4]) : 1;
if (time <= 0) {
ast_cli(a->fd, " Recalc must be greater than 0. Defaulting to 1.\n");
time = 1;
}
if (time > MAX_RECALC) {
ast_cli(a->fd, " Maximum limit of recalc exceeded by %d, truncating value to %d\n", time - MAX_RECALC, MAX_RECALC);
time = MAX_RECALC;
}
ast_cli(a->fd, " Recalculating Codec Translation (number of sample seconds: %d)\n\n", time);
AST_RWLIST_WRLOCK(&translators);
matrix_rebuild(time);
AST_RWLIST_UNLOCK(&translators);
}
static char *handle_show_translation_table(struct ast_cli_args *a)
{
int x, y, i, k;
int longest = 7; /* slin192 */
int max_codec_index = 0, curlen = 0;
struct ast_str *out = ast_str_create(1024);
struct ast_codec *codec;
/* Get the length of the longest (usable?) codec name,
so we know how wide the left side should be */
for (i = 1; (codec = ast_codec_get_by_id(i)); ao2_ref(codec, -1), ++i) {
++max_codec_index;
if (codec->type != AST_MEDIA_TYPE_AUDIO) {