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Merged revisions 260049 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines
  
  Fixes crash in audiohook_write_list
  
  The middle_frame in the audiohook_write_list function was
  being freed if a audiohook manipulator returned a failure.
  This is incorrect logic.  This patch resolves this and
  adds detailed descriptions of how this function should work
  and why manipulator failures must be ignored.
  
  (closes issue #17052)
  Reported by: dvossel
  Tested by: dvossel

  (closes issue #16196)
  Reported by: atis
  
  Review: https://reviewboard.asterisk.org/r/623/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2010-04-29 15:33:27 +00:00
parent 6adcbd6be5
commit d4358a46a9
2 changed files with 44 additions and 14 deletions
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13
include/asterisk/audiohook.h
View File

@@ -75,9 +75,16 @@ struct ast_audiohook;
* \param chan Channel
* \param frame Frame of audio to manipulate
* \param direction Direction frame came from
* \return Returns 0 on success, -1 on failure
* \note An audiohook does not have any reference to a private data structure for manipulate types. It is up to the manipulate callback to store this data
* via it's own method. An example would be datastores.
* \return Returns 0 on success, -1 on failure.
* \note An audiohook does not have any reference to a private data structure for manipulate
* types. It is up to the manipulate callback to store this data via it's own method.
* An example would be datastores.
* \note The input frame should never be freed or corrupted during a manipulate callback.
* If the callback has the potential to corrupt the frame's data during manipulation,
* local data should be used for the manipulation and only copied to the frame on
* success.
* \note A failure return value indicates that the frame was not manipulated and that
* is being returned in its original state.
*/
typedef int (*ast_audiohook_manipulate_callback)(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction);

45
main/audiohook.c
View File

@@ -584,7 +584,29 @@ static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, str
return frame;
}
/*! \brief Pass an AUDIO frame off to be handled by the audiohook core
/*!
* \brief Pass an AUDIO frame off to be handled by the audiohook core
*
* \details
* This function has 3 ast_frames and 3 parts to handle each. At the beginning of this
* function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
* input frame.
*
* Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
* format. The result of this part is middle_frame is guaranteed to be in
* SLINEAR format for Part_2.
* Part_2: Send middle_frame off to spies and manipulators. At this point middle_frame is
* either a new frame as result of the translation, or points directly to the start_frame
* because no translation to SLINEAR audio was required. The result of this part
* is end_frame will be updated to point to middle_frame if any audiohook manipulation
* took place.
* Part_3: Translate end_frame's audio back into the format of start frame if necessary.
* At this point if middle_frame != end_frame, we are guaranteed that no manipulation
* took place and middle_frame can be freed as it was translated... If middle_frame was
* not translated and still pointed to start_frame, it would be equal to end_frame as well
* regardless if manipulation took place which would not result in this free. The result
* of this part is end_frame is guaranteed to be the format of start_frame for the return.
*
* \param chan Channel that the list is coming off of
* \param audiohook_list List of audiohooks
* \param direction Direction frame is coming in from
@@ -599,6 +621,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
struct ast_audiohook *audiohook = NULL;
int samples = frame->samples;
/* ---Part_1. translate start_frame to SLINEAR if necessary. */
/* If the frame coming in is not signed linear we have to send it through the in_translate path */
if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
if (in_translate->format != frame->subclass.codec) {
@@ -613,6 +636,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
samples = middle_frame->samples;
}
/* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
/* Queue up signed linear frame to each spy */
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
ast_audiohook_lock(audiohook);
@@ -666,20 +690,21 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
audiohook->manipulate_callback(audiohook, chan, NULL, direction);
continue;
}
/* Feed in frame to manipulation */
/* Feed in frame to manipulation. */
if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
ast_frfree(middle_frame);
middle_frame = NULL;
/* XXX IGNORE FAILURE */
/* If the manipulation fails then the frame will be returned in its original state.
* Since there are potentially more manipulator callbacks in the list, no action should
* be taken here to exit early. */
}
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
if (middle_frame) {
end_frame = middle_frame;
}
end_frame = middle_frame;
}
/* Now we figure out what to do with our end frame (whether to transcode or not) */
/* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
if (middle_frame == end_frame) {
/* Middle frame was modified and became the end frame... let's see if we need to transcode */
if (end_frame->subclass.codec != start_frame->subclass.codec) {
@@ -704,9 +729,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
} else {
/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
if (middle_frame) {
ast_frfree(middle_frame);
}
ast_frfree(middle_frame);
}
return end_frame;