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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2014-04-11 07:07:36 +00:00
parent 4f30c7e91f
commit 2a4205df2c
1 changed files with 46 additions and 27 deletions
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73
main/audiohook.c
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@@ -227,16 +227,19 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0);
/* Ensure the factory is able to give us the samples we want */
if (samples > ast_slinfactory_available(factory))
if (samples > ast_slinfactory_available(factory)) {
return NULL;
}
/* Read data in from factory */
if (!ast_slinfactory_read(factory, buf, samples))
if (!ast_slinfactory_read(factory, buf, samples)) {
return NULL;
}
/* If a volume adjustment needs to be applied apply it */
if (vol)
if (vol) {
ast_frame_adjust_volume(&frame, vol);
}
return ast_frdup(&frame);
}
@@ -284,10 +287,11 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
int count = 0;
short adjust_value = abs(audiohook->options.read_volume);
for (count = 0; count < samples; count++) {
if (audiohook->options.read_volume > 0)
if (audiohook->options.read_volume > 0) {
ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
else if (audiohook->options.read_volume < 0)
} else if (audiohook->options.read_volume < 0) {
ast_slinear_saturated_divide(&buf1[count], &adjust_value);
}
}
}
}
@@ -304,10 +308,11 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
int count = 0;
short adjust_value = abs(audiohook->options.write_volume);
for (count = 0; count < samples; count++) {
if (audiohook->options.write_volume > 0)
if (audiohook->options.write_volume > 0) {
ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
else if (audiohook->options.write_volume < 0)
} else if (audiohook->options.write_volume < 0) {
ast_slinear_saturated_divide(&buf2[count], &adjust_value);
}
}
}
}
@@ -455,12 +460,13 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
}
/* Drop into respective list */
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
}
audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
@@ -498,13 +504,15 @@ void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audio
*/
int ast_audiohook_detach(struct ast_audiohook *audiohook)
{
if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
return 0;
}
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
ast_audiohook_trigger_wait(audiohook);
}
return 0;
}
@@ -536,10 +544,12 @@ void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
/* Drop translation paths if present */
for (i = 0; i < 2; i++) {
if (audiohook_list->in_translate[i].trans_pvt)
if (audiohook_list->in_translate[i].trans_pvt) {
ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
if (audiohook_list->out_translate[i].trans_pvt)
}
if (audiohook_list->out_translate[i].trans_pvt) {
ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
}
}
/* Free ourselves */
@@ -556,18 +566,21 @@ static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list
struct ast_audiohook *audiohook = NULL;
AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
if (!strcasecmp(audiohook->source, source))
if (!strcasecmp(audiohook->source, source)) {
return audiohook;
}
}
AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
if (!strcasecmp(audiohook->source, source))
if (!strcasecmp(audiohook->source, source)) {
return audiohook;
}
}
AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
if (!strcasecmp(audiohook->source, source))
if (!strcasecmp(audiohook->source, source)) {
return audiohook;
}
}
return NULL;
@@ -618,8 +631,9 @@ int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
ast_channel_unlock(chan);
if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
}
return (audiohook ? 0 : -1);
}
@@ -643,12 +657,13 @@ int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audioho
return -1;
}
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
}
audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
@@ -680,8 +695,9 @@ static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, str
audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
continue;
}
if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) {
audiohook->manipulate_callback(audiohook, chan, frame, direction);
}
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
@@ -829,8 +845,9 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
/* Take audio from this whisper source and combine it into our main buffer */
for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
ast_slinear_saturated_add(data1, data2);
}
}
ast_audiohook_unlock(audiohook);
}
@@ -911,12 +928,13 @@ int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
/* Pass off frame to it's respective list write function */
if (frame->frametype == AST_FRAME_VOICE)
if (frame->frametype == AST_FRAME_VOICE) {
return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
else if (frame->frametype == AST_FRAME_DTMF)
} else if (frame->frametype == AST_FRAME_DTMF) {
return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
else
} else {
return frame;
}
}
/*! \brief Wait for audiohook trigger to be triggered
@@ -942,8 +960,9 @@ int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *
int count = 0;
struct ast_audiohook *ah = NULL;
if (!ast_channel_audiohooks(chan))
if (!ast_channel_audiohooks(chan)) {
return -1;
}
switch (type) {
case AST_AUDIOHOOK_TYPE_SPY: