ARI: Add recording controls

This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changes/78/78/1
David M. Lee 12 years ago
parent b97d318b7b
commit c790848794

@ -1674,7 +1674,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, MVM_OPERATOR))
canceldtmf = "0";
cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
if (record_gain)
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &zero_gain, sizeof(zero_gain), 0);
if (cmd == -1) /* User has hung up, no options to give */

@ -14682,7 +14682,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, VM_OPERATOR))
canceldtmf = "0";
cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
if (strchr(canceldtmf, cmd)) {
/* need this flag here to distinguish between pressing '0' during message recording or after */
canceleddtmf = 1;

@ -343,6 +343,18 @@ static void print_frame(struct ast_frame *frame)
case AST_CONTROL_STREAM_FORWARD:
ast_verbose("SubClass: STREAM_FORWARD\n");
break;
case AST_CONTROL_RECORD_CANCEL:
ast_verbose("SubClass: RECORD_CANCEL\n");
break;
case AST_CONTROL_RECORD_STOP:
ast_verbose("SubClass: RECORD_STOP\n");
break;
case AST_CONTROL_RECORD_SUSPEND:
ast_verbose("SubClass: RECORD_SUSPEND\n");
break;
case AST_CONTROL_RECORD_MUTE:
ast_verbose("SubClass: RECORD_MUTE\n");
break;
}
if (frame->subclass.integer == -1) {

@ -709,11 +709,12 @@ enum ast_record_if_exists {
* skip_confirmation_sound is false.
*
* \param chan the channel being recorded
* \param playfile Filename of sound to play before recording begins
* \param playfile Filename of sound to play before recording begins. A beep is also played when playfile completes, before the recording begins.
* \param recordfile Filename to save the recording
* \param maxtime_sec Longest possible message length in seconds
* \param fmt string containing all formats to be recorded delimited by '|'
* \param duration pointer to integer for storing length of the recording
* \param beep If true, play a beep before recording begins (and doesn't play \a playfile)
* \param sound_duration pointer to integer for storing length of the recording minus all silence
* \param silencethreshold tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
* \param maxsilence_ms Length of time in milliseconds which will trigger a timeout from silence, -1 for default
@ -728,7 +729,7 @@ enum ast_record_if_exists {
* \retval 't' Recording ended from the message exceeding the maximum duration
* \retval dtmfchar Recording ended via the return value's DTMF character for either cancel or accept.
*/
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists);
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence_ms, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists);
/*!
* \brief Record a file based on input from a channel. Use default accept and cancel DTMF.

@ -278,7 +278,11 @@ enum ast_control_frame_type {
AST_CONTROL_STREAM_RESTART = 1002, /*!< Indicate to a channel in playback to restart the stream */
AST_CONTROL_STREAM_REVERSE = 1003, /*!< Indicate to a channel in playback to rewind */
AST_CONTROL_STREAM_FORWARD = 1004, /*!< Indicate to a channel in playback to fast forward */
/* Control frames to manipulate recording on a channel. */
AST_CONTROL_RECORD_CANCEL = 1100, /*!< Indicated to a channel in record to stop recording and discard the file */
AST_CONTROL_RECORD_STOP = 1101, /*!< Indicated to a channel in record to stop recording */
AST_CONTROL_RECORD_SUSPEND = 1102, /*!< Indicated to a channel in record to suspend/unsuspend recording */
AST_CONTROL_RECORD_MUTE = 1103, /*!< Indicated to a channel in record to mute/unmute (i.e. write silence) recording */
};
enum ast_frame_read_action {

@ -44,14 +44,30 @@ enum stasis_app_recording_state {
STASIS_APP_RECORDING_STATE_PAUSED,
/*! The media has stopped recording */
STASIS_APP_RECORDING_STATE_COMPLETE,
/*! The media has stopped playing */
/*! The media has stopped recording, with error */
STASIS_APP_RECORDING_STATE_FAILED,
/*! The media has stopped recording, discard the recording file */
STASIS_APP_RECORDING_STATE_CANCELED,
/*! Sentinel */
STASIS_APP_RECORDING_STATE_MAX,
};
/*! Valid operation for controlling a recording. */
enum stasis_app_recording_media_operation {
/*! Stop the recording operation. */
/*! Stop the recording, deleting the media file(s) */
STASIS_APP_RECORDING_CANCEL,
/*! Stop the recording. */
STASIS_APP_RECORDING_STOP,
/*! Pause the recording */
STASIS_APP_RECORDING_PAUSE,
/*! Unpause the recording */
STASIS_APP_RECORDING_UNPAUSE,
/*! Mute the recording (record silence) */
STASIS_APP_RECORDING_MUTE,
/*! Unmute the recording */
STASIS_APP_RECORDING_UNMUTE,
/*! Sentinel */
STASIS_APP_RECORDING_OPER_MAX,
};
#define STASIS_APP_RECORDING_TERMINATE_INVALID 0

@ -1145,6 +1145,78 @@ int ast_play_and_wait(struct ast_channel *chan, const char *fn)
return d;
}
/*!
* \brief Construct a silence frame of the same duration as \a orig.
*
* The \a orig frame must be \ref AST_FORMAT_SLINEAR.
*
* \param orig Frame as basis for silence to generate.
* \return New frame of silence; free with ast_frfree().
* \return \c NULL on error.
*/
static struct ast_frame *make_silence(const struct ast_frame *orig)
{
struct ast_frame *silence;
size_t size;
size_t datalen;
size_t samples = 0;
struct ast_frame *next;
if (!orig) {
return NULL;
}
if (orig->subclass.format.id != AST_FORMAT_SLINEAR) {
ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
return NULL;
}
for (next = AST_LIST_NEXT(orig, frame_list);
orig;
orig = next, next = orig ? AST_LIST_NEXT(orig, frame_list) : NULL) {
samples += orig->samples;
}
ast_verb(4, "Silencing %zd samples\n", samples);
datalen = sizeof(short) * samples;
size = sizeof(*silence) + datalen;
silence = ast_calloc(1, size);
if (!silence) {
return NULL;
}
silence->mallocd = AST_MALLOCD_HDR;
silence->frametype = AST_FRAME_VOICE;
silence->data.ptr = (void *)(silence + 1);
silence->samples = samples;
silence->datalen = datalen;
ast_format_set(&silence->subclass.format, AST_FORMAT_SLINEAR, 0);
return silence;
}
/*!
* \brief Sets a channel's read format to \ref AST_FORMAT_SLINEAR, recording
* its original format.
*
* \param chan Channel to modify.
* \param[out] orig_format Output variable to store channel's original read
* format.
* \return 0 on success.
* \return -1 on error.
*/
static int set_read_to_slin(struct ast_channel *chan, struct ast_format *orig_format)
{
if (!chan || !orig_format) {
return -1;
}
ast_format_copy(orig_format, ast_channel_readformat(chan));
return ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
}
static int global_silence_threshold = 128;
static int global_maxsilence = 0;
@ -1274,8 +1346,7 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
ast_format_copy(&rfmt, ast_channel_readformat(chan));
res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
ast_dsp_free(sildet);
@ -1293,9 +1364,15 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
}
if (x == fmtcnt) {
/* Loop forever, writing the packets we read to the writer(s), until
we read a digit or get a hangup */
/* Loop, writing the packets we read to the writer(s), until
* we have reason to stop. */
struct ast_frame *f;
int paused = 0;
int muted = 0;
time_t pause_start = 0;
int paused_secs = 0;
int pausedsilence = 0;
for (;;) {
if (!(res = ast_waitfor(chan, 2000))) {
ast_debug(1, "One waitfor failed, trying another\n");
@ -1315,11 +1392,29 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
}
if (f->frametype == AST_FRAME_VOICE) {
/* write each format */
for (x = 0; x < fmtcnt; x++) {
if (prepend && !others[x]) {
break;
if (paused) {
/* It's all good */
res = 0;
} else {
RAII_VAR(struct ast_frame *, silence, NULL, ast_frame_dtor);
struct ast_frame *orig = f;
if (muted) {
silence = make_silence(orig);
if (!silence) {
ast_log(LOG_WARNING,
"Error creating silence\n");
break;
}
f = silence;
}
res = ast_writestream(others[x], f);
for (x = 0; x < fmtcnt; x++) {
if (prepend && !others[x]) {
break;
}
res = ast_writestream(others[x], f);
}
f = orig;
}
/* Silence Detection */
@ -1331,6 +1426,17 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
}
olddspsilence = dspsilence;
if (paused) {
/* record how much silence there was while we are paused */
pausedsilence = dspsilence;
} else if (dspsilence > pausedsilence) {
/* ignore the paused silence */
dspsilence -= pausedsilence;
} else {
/* dspsilence has reset, reset pausedsilence */
pausedsilence = 0;
}
if (dspsilence > maxsilence) {
/* Ended happily with silence */
ast_verb(3, "Recording automatically stopped after a silence of %d seconds\n", dspsilence/1000);
@ -1362,15 +1468,51 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
break;
}
if (strchr(canceldtmf, f->subclass.integer)) {
ast_verb(3, "User cancelled message by pressing %c\n", f->subclass.integer);
ast_verb(3, "User canceled message by pressing %c\n", f->subclass.integer);
res = f->subclass.integer;
outmsg = 0;
break;
}
} else if (f->frametype == AST_FRAME_CONTROL) {
if (f->subclass.integer == AST_CONTROL_RECORD_CANCEL) {
ast_verb(3, "Message canceled by control\n");
outmsg = 0; /* cancels the recording */
res = 0;
break;
} else if (f->subclass.integer == AST_CONTROL_RECORD_STOP) {
ast_verb(3, "Message ended by control\n");
res = 0;
break;
} else if (f->subclass.integer == AST_CONTROL_RECORD_SUSPEND) {
paused = !paused;
ast_verb(3, "Message %spaused by control\n",
paused ? "" : "un");
if (paused) {
pause_start = time(NULL);
} else {
paused_secs += time(NULL) - pause_start;
}
} else if (f->subclass.integer == AST_CONTROL_RECORD_MUTE) {
muted = !muted;
ast_verb(3, "Message %smuted by control\n",
muted ? "" : "un");
/* We can only silence slin frames, so
* set the mode, if we haven't already
* for sildet
*/
if (muted && !rfmt.id) {
ast_verb(3, "Setting read format to linear mode\n");
res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
break;
}
}
}
}
if (maxtime) {
if (maxtime && !paused) {
end = time(NULL);
if (maxtime < (end - start)) {
if (maxtime < (end - start - paused_secs)) {
ast_verb(3, "Took too long, cutting it short...\n");
res = 't';
outmsg = 2;
@ -1493,9 +1635,9 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
static const char default_acceptdtmf[] = "#";
static const char default_canceldtmf[] = "";
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
{
return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
}
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path)

@ -4303,6 +4303,10 @@ static int attribute_const is_visible_indication(enum ast_control_frame_type con
case AST_CONTROL_STREAM_REVERSE:
case AST_CONTROL_STREAM_FORWARD:
case AST_CONTROL_STREAM_RESTART:
case AST_CONTROL_RECORD_CANCEL:
case AST_CONTROL_RECORD_STOP:
case AST_CONTROL_RECORD_SUSPEND:
case AST_CONTROL_RECORD_MUTE:
break;
case AST_CONTROL_INCOMPLETE:
@ -4561,6 +4565,10 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
case AST_CONTROL_STREAM_REVERSE:
case AST_CONTROL_STREAM_FORWARD:
case AST_CONTROL_STREAM_RESTART:
case AST_CONTROL_RECORD_CANCEL:
case AST_CONTROL_RECORD_STOP:
case AST_CONTROL_RECORD_SUSPEND:
case AST_CONTROL_RECORD_MUTE:
/* Nothing left to do for these. */
res = 0;
break;

@ -71,27 +71,81 @@ void ast_ari_get_live_recording(struct ast_variable *headers,
ast_ari_response_ok(response, ast_json_ref(json));
}
void ast_ari_cancel_recording(struct ast_variable *headers, struct ast_cancel_recording_args *args, struct ast_ari_response *response)
static void control_recording(const char *name,
enum stasis_app_recording_media_operation operation,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_cancel_recording\n");
RAII_VAR(struct stasis_app_recording *, recording, NULL, ao2_cleanup);
RAII_VAR(struct ast_json *, json, NULL, ast_json_unref);
enum stasis_app_recording_oper_results res;
recording = stasis_app_recording_find_by_name(name);
if (recording == NULL) {
ast_ari_response_error(response, 404, "Not Found",
"Recording not found");
return;
}
res = stasis_app_recording_operation(recording, operation);
switch (res) {
case STASIS_APP_RECORDING_OPER_OK:
ast_ari_response_no_content(response);
return;
case STASIS_APP_RECORDING_OPER_FAILED:
ast_ari_response_error(response, 500,
"Internal Server Error", "Recording operation failed");
return;
case STASIS_APP_RECORDING_OPER_NOT_RECORDING:
ast_ari_response_error(response, 409,
"Conflict", "Recording not in session");
}
}
void ast_ari_stop_recording(struct ast_variable *headers, struct ast_stop_recording_args *args, struct ast_ari_response *response)
void ast_ari_cancel_recording(struct ast_variable *headers,
struct ast_cancel_recording_args *args,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_stop_recording\n");
control_recording(args->recording_name, STASIS_APP_RECORDING_CANCEL,
response);
}
void ast_ari_pause_recording(struct ast_variable *headers, struct ast_pause_recording_args *args, struct ast_ari_response *response)
void ast_ari_stop_recording(struct ast_variable *headers,
struct ast_stop_recording_args *args,
struct ast_ari_response *response)
{
control_recording(args->recording_name, STASIS_APP_RECORDING_STOP,
response);
}
void ast_ari_pause_recording(struct ast_variable *headers,
struct ast_pause_recording_args *args,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_pause_recording\n");
control_recording(args->recording_name, STASIS_APP_RECORDING_PAUSE,
response);
}
void ast_ari_unpause_recording(struct ast_variable *headers, struct ast_unpause_recording_args *args, struct ast_ari_response *response)
void ast_ari_unpause_recording(struct ast_variable *headers,
struct ast_unpause_recording_args *args,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_unpause_recording\n");
control_recording(args->recording_name, STASIS_APP_RECORDING_UNPAUSE,
response);
}
void ast_ari_mute_recording(struct ast_variable *headers, struct ast_mute_recording_args *args, struct ast_ari_response *response)
void ast_ari_mute_recording(struct ast_variable *headers,
struct ast_mute_recording_args *args,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_mute_recording\n");
control_recording(args->recording_name, STASIS_APP_RECORDING_MUTE,
response);
}
void ast_ari_unmute_recording(struct ast_variable *headers, struct ast_unmute_recording_args *args, struct ast_ari_response *response)
void ast_ari_unmute_recording(struct ast_variable *headers,
struct ast_unmute_recording_args *args,
struct ast_ari_response *response)
{
ast_log(LOG_ERROR, "TODO: ast_ari_unmute_recording\n");
control_recording(args->recording_name, STASIS_APP_RECORDING_UNMUTE,
response);
}

@ -134,7 +134,7 @@ struct ast_pause_recording_args {
/*!
* \brief Pause a live recording.
*
* Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused.
* Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused. Paused time is not included in the accounting for maxDurationSeconds.
*
* \param headers HTTP headers
* \param args Swagger parameters

@ -295,6 +295,7 @@ static void ast_ari_get_live_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
is_valid = 1;
break;
default:
@ -351,6 +352,7 @@ static void ast_ari_cancel_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
is_valid = 1;
break;
default:
@ -407,6 +409,7 @@ static void ast_ari_stop_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
is_valid = 1;
break;
default:
@ -463,6 +466,8 @@ static void ast_ari_pause_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
case 409: /* Recording not in session */
is_valid = 1;
break;
default:
@ -519,6 +524,8 @@ static void ast_ari_unpause_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
case 409: /* Recording not in session */
is_valid = 1;
break;
default:
@ -575,6 +582,8 @@ static void ast_ari_mute_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
case 409: /* Recording not in session */
is_valid = 1;
break;
default:
@ -631,6 +640,8 @@ static void ast_ari_unmute_recording_cb(
break;
case 500: /* Internal Server Error */
case 501: /* Not Implemented */
case 404: /* Recording not found */
case 409: /* Recording not in session */
is_valid = 1;
break;
default:

@ -59,11 +59,13 @@ struct stasis_app_recording {
struct stasis_app_recording_options *options;
/*! Absolute path (minus extension) of the recording */
char *absolute_name;
/*! Control object for the channel we're playing back to */
/*! Control object for the channel we're recording */
struct stasis_app_control *control;
/*! Current state of the recording. */
enum stasis_app_recording_state state;
/*! Indicates whether the recording is currently muted */
int muted:1;
};
static int recording_hash(const void *obj, int flags)
@ -99,6 +101,10 @@ static const char *state_to_string(enum stasis_app_recording_state state)
return "done";
case STASIS_APP_RECORDING_STATE_FAILED:
return "failed";
case STASIS_APP_RECORDING_STATE_CANCELED:
return "canceled";
case STASIS_APP_RECORDING_STATE_MAX:
return "?";
}
return "?";
@ -253,12 +259,13 @@ static void *record_file(struct stasis_app_control *control,
}
ast_play_and_record_full(chan,
recording->options->beep ? "beep" : NULL,
NULL, /* playfile */
recording->absolute_name,
recording->options->max_duration_seconds,
recording->options->format,
&duration,
NULL, /* sound_duration */
recording->options->beep,
-1, /* silencethreshold */
recording->options->max_silence_seconds * 1000,
NULL, /* path */
@ -403,12 +410,127 @@ struct ast_json *stasis_app_recording_to_json(
return ast_json_ref(json);
}
typedef int (*recording_operation_cb)(struct stasis_app_recording *recording);
static int recording_noop(struct stasis_app_recording *recording)
{
return 0;
}
static int recording_disregard(struct stasis_app_recording *recording)
{
recording->state = STASIS_APP_RECORDING_STATE_CANCELED;
return 0;
}
static int recording_cancel(struct stasis_app_recording *recording)
{
int res = 0;
recording->state = STASIS_APP_RECORDING_STATE_CANCELED;
res |= stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_CANCEL);
res |= ast_filedelete(recording->absolute_name, NULL);
return res;
}
static int recording_stop(struct stasis_app_recording *recording)
{
recording->state = STASIS_APP_RECORDING_STATE_COMPLETE;
return stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_STOP);
}
static int recording_pause(struct stasis_app_recording *recording)
{
recording->state = STASIS_APP_RECORDING_STATE_PAUSED;
return stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_SUSPEND);
}
static int recording_unpause(struct stasis_app_recording *recording)
{
recording->state = STASIS_APP_RECORDING_STATE_RECORDING;
return stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_SUSPEND);
}
static int recording_mute(struct stasis_app_recording *recording)
{
if (recording->muted) {
/* already muted */
return 0;
}
recording->muted = 1;
return stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_MUTE);
}
static int recording_unmute(struct stasis_app_recording *recording)
{
if (!recording->muted) {
/* already unmuted */
return 0;
}
return stasis_app_control_queue_control(recording->control,
AST_CONTROL_RECORD_MUTE);
}
recording_operation_cb operations[STASIS_APP_RECORDING_STATE_MAX][STASIS_APP_RECORDING_OPER_MAX] = {
[STASIS_APP_RECORDING_STATE_QUEUED][STASIS_APP_RECORDING_CANCEL] = recording_disregard,
[STASIS_APP_RECORDING_STATE_QUEUED][STASIS_APP_RECORDING_STOP] = recording_disregard,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_CANCEL] = recording_cancel,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_STOP] = recording_stop,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_PAUSE] = recording_pause,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_UNPAUSE] = recording_noop,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_MUTE] = recording_mute,
[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_UNMUTE] = recording_unmute,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_CANCEL] = recording_cancel,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_STOP] = recording_stop,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_PAUSE] = recording_noop,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_UNPAUSE] = recording_unpause,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_MUTE] = recording_mute,
[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_UNMUTE] = recording_unmute,
};
enum stasis_app_recording_oper_results stasis_app_recording_operation(
struct stasis_app_recording *recording,
enum stasis_app_recording_media_operation operation)
{
ast_assert(0); // TODO
return STASIS_APP_RECORDING_OPER_FAILED;
recording_operation_cb cb;
SCOPED_AO2LOCK(lock, recording);
if (recording->state < 0 || recording->state >= STASIS_APP_RECORDING_STATE_MAX) {
ast_log(LOG_WARNING, "Invalid recording state %d\n",
recording->state);
return -1;
}
if (operation < 0 || operation >= STASIS_APP_RECORDING_OPER_MAX) {
ast_log(LOG_WARNING, "Invalid recording operation %d\n",
operation);
return -1;
}
cb = operations[recording->state][operation];
if (!cb) {
if (recording->state != STASIS_APP_RECORDING_STATE_RECORDING) {
/* So we can be specific in our error message. */
return STASIS_APP_RECORDING_OPER_NOT_RECORDING;
} else {
/* And, really, all operations should be valid during
* recording */
ast_log(LOG_ERROR,
"Unhandled operation during recording: %d\n",
operation);
return STASIS_APP_RECORDING_OPER_FAILED;
}
}
return cb(recording) ?
STASIS_APP_RECORDING_OPER_FAILED : STASIS_APP_RECORDING_OPER_OK;
}
static int load_module(void)

@ -87,6 +87,12 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
}
]
},
{
@ -103,6 +109,12 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
}
]
}
]
@ -124,6 +136,12 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
}
]
}
]
@ -134,7 +152,7 @@
{
"httpMethod": "POST",
"summary": "Pause a live recording.",
"notes": "Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused.",
"notes": "Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused. Paused time is not included in the accounting for maxDurationSeconds.",
"nickname": "pauseRecording",
"responseClass": "void",
"parameters": [
@ -146,6 +164,16 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
},
{
"code": 409,
"reason": "Recording not in session"
}
]
}
]
@ -167,6 +195,16 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
},
{
"code": 409,
"reason": "Recording not in session"
}
]
}
]
@ -189,6 +227,16 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
},
{
"code": 409,
"reason": "Recording not in session"
}
]
}
]
@ -210,6 +258,16 @@
"allowMultiple": false,
"dataType": "string"
}
],
"errorResponses": [
{
"code": 404,
"reason": "Recording not found"
},
{
"code": 409,
"reason": "Recording not in session"
}
]
}
]

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