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Merged revisions 224855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -142,25 +142,24 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
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struct ast_datastore *datastore = NULL;
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struct speex_direction_info *sdi = NULL;
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struct speex_info *si = NULL;
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char source[80];
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/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
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return 0;
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return -1;
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}
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ast_channel_lock(chan);
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/* We are called with chan already locked */
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if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
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ast_channel_unlock(chan);
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return 0;
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return -1;
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}
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ast_channel_unlock(chan);
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si = datastore->data;
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sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
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if (!sdi) {
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return 0;
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return -1;
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}
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if (sdi->samples != frame->samples) {
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@@ -171,7 +170,7 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
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if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
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return -1;
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}
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speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
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if (sdi->agc) {
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@@ -182,6 +181,12 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
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}
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speex_preprocess(sdi->state, frame->data.ptr, NULL);
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snprintf(source, sizeof(source), "%s/speex", frame->src);
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if (frame->mallocd & AST_MALLOCD_SRC) {
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ast_free((char *) frame->src);
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}
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frame->src = ast_strdup(source);
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frame->mallocd |= AST_MALLOCD_SRC;
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return 0;
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}
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@@ -574,7 +574,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
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struct ast_audiohook *audiohook = NULL;
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int samples = frame->samples;
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/* If the frame coming in is not signed linear we have to send it through the in_translate path */
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if (frame->subclass != AST_FORMAT_SLINEAR) {
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if (in_translate->format != frame->subclass) {
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@@ -645,11 +645,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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continue;
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}
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/* Feed in frame to manipulation */
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audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
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if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
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ast_frfree(middle_frame);
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middle_frame = NULL;
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}
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ast_audiohook_unlock(audiohook);
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}
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AST_LIST_TRAVERSE_SAFE_END;
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end_frame = middle_frame;
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if (middle_frame) {
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end_frame = middle_frame;
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}
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}
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/* Now we figure out what to do with our end frame (whether to transcode or not) */
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@@ -677,7 +682,9 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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}
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} else {
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/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
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ast_frfree(middle_frame);
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if (middle_frame) {
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ast_frfree(middle_frame);
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}
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}
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return end_frame;
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