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Merged revisions 97976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines Fix various timing calculations that made assumptions that the audio being processed was at a sample rate of 8 kHz. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -284,6 +284,14 @@ struct ast_trans_pvt *ast_translator_build_path(int dest, int source)
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return head;
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}
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static inline int is16kHz(int format)
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{
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if (format == AST_FORMAT_G722 || format == AST_FORMAT_SLINEAR16)
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return 1;
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return 0;
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}
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/*! \brief do the actual translation */
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struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, int consume)
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{
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@@ -302,6 +310,11 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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/* XXX hmmm... check this below */
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if (!ast_tvzero(f->delivery)) {
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int in_rate = 8000;
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if (is16kHz(f->subclass))
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in_rate = 16000;
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if (!ast_tvzero(path->nextin)) {
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/* Make sure this is in line with what we were expecting */
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if (!ast_tveq(path->nextin, f->delivery)) {
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@@ -320,7 +333,7 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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path->nextout = f->delivery;
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}
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/* Predict next incoming sample */
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path->nextin = ast_tvadd(path->nextin, ast_samp2tv(f->samples, 8000));
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path->nextin = ast_tvadd(path->nextin, ast_samp2tv(f->samples, in_rate));
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}
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delivery = f->delivery;
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for ( ; out && p ; p = p->next) {
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@@ -333,6 +346,11 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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return NULL;
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/* we have a frame, play with times */
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if (!ast_tvzero(delivery)) {
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int out_rate = 8000;
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if (is16kHz(out->subclass))
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out_rate = 16000;
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/* Regenerate prediction after a discontinuity */
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if (ast_tvzero(path->nextout))
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path->nextout = ast_tvnow();
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@@ -342,7 +360,7 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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/* Predict next outgoing timestamp from samples in this
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frame. */
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path->nextout = ast_tvadd(path->nextout, ast_samp2tv( out->samples, 8000));
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path->nextout = ast_tvadd(path->nextout, ast_samp2tv(out->samples, out_rate));
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} else {
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out->delivery = ast_tv(0, 0);
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out->has_timing_info = has_timing_info;
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@@ -361,11 +379,15 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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/*! \brief compute the cost of a single translation step */
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static void calc_cost(struct ast_translator *t, int seconds)
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{
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int sofar=0;
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int num_samples = 0;
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struct ast_trans_pvt *pvt;
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struct rusage start;
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struct rusage end;
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int cost;
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int out_rate = 8000;
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if (is16kHz(t->dstfmt))
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out_rate = 16000;
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if (!seconds)
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seconds = 1;
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@@ -376,15 +398,18 @@ static void calc_cost(struct ast_translator *t, int seconds)
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t->cost = 999999;
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return;
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}
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pvt = newpvt(t);
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if (!pvt) {
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ast_log(LOG_WARNING, "Translator '%s' appears to be broken and will probably fail.\n", t->name);
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t->cost = 999999;
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return;
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}
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getrusage(RUSAGE_SELF, &start);
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/* Call the encoder until we've processed the required number of samples */
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while (sofar < seconds * 8000) {
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while (num_samples < seconds * out_rate) {
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struct ast_frame *f = t->sample();
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if (!f) {
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ast_log(LOG_WARNING, "Translator '%s' failed to produce a sample frame.\n", t->name);
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@@ -395,15 +420,20 @@ static void calc_cost(struct ast_translator *t, int seconds)
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framein(pvt, f);
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ast_frfree(f);
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while ((f = t->frameout(pvt))) {
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sofar += f->samples;
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num_samples += f->samples;
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ast_frfree(f);
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}
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}
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getrusage(RUSAGE_SELF, &end);
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cost = ((end.ru_utime.tv_sec - start.ru_utime.tv_sec)*1000000) + end.ru_utime.tv_usec - start.ru_utime.tv_usec;
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cost += ((end.ru_stime.tv_sec - start.ru_stime.tv_sec)*1000000) + end.ru_stime.tv_usec - start.ru_stime.tv_usec;
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cost = ((end.ru_utime.tv_sec - start.ru_utime.tv_sec) * 1000000) + end.ru_utime.tv_usec - start.ru_utime.tv_usec;
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cost += ((end.ru_stime.tv_sec - start.ru_stime.tv_sec) * 1000000) + end.ru_stime.tv_usec - start.ru_stime.tv_usec;
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destroy(pvt);
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t->cost = cost / seconds;
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if (!t->cost)
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t->cost = 1;
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}
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