mirror of
				https://github.com/asterisk/asterisk.git
				synced 2025-10-31 02:37:10 +00:00 
			
		
		
		
	Kill off red blobs in most of main/*
Everything still compiled after making these changes, so I assume these whitespace-only changes didn't break anything (and shouldn't have). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
		| @@ -226,7 +226,7 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio | ||||
| 	/* Ensure the factory is able to give us the samples we want */ | ||||
| 	if (samples > ast_slinfactory_available(factory)) | ||||
| 		return NULL; | ||||
| 	 | ||||
|  | ||||
| 	/* Read data in from factory */ | ||||
| 	if (!ast_slinfactory_read(factory, buf, samples)) | ||||
| 		return NULL; | ||||
| @@ -356,10 +356,10 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio | ||||
| 		samples_converted = samples * (ast_format_rate(format) / (float) audiohook->hook_internal_samp_rate); | ||||
| 	} | ||||
|  | ||||
| 	if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?  | ||||
| 		audiohook_read_frame_both(audiohook, samples_converted, read_reference, write_reference) :  | ||||
| 		audiohook_read_frame_single(audiohook, samples_converted, direction)))) {  | ||||
| 		return NULL;  | ||||
| 	if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? | ||||
| 		audiohook_read_frame_both(audiohook, samples_converted, read_reference, write_reference) : | ||||
| 		audiohook_read_frame_single(audiohook, samples_converted, direction)))) { | ||||
| 		return NULL; | ||||
| 	} | ||||
|  | ||||
| 	/* If they don't want signed linear back out, we'll have to send it through the translation path */ | ||||
| @@ -536,7 +536,7 @@ int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list) | ||||
| 		if (audiohook_list->out_translate[i].trans_pvt) | ||||
| 			ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt); | ||||
| 	} | ||||
| 	 | ||||
|  | ||||
| 	/* Free ourselves */ | ||||
| 	ast_free(audiohook_list); | ||||
|  | ||||
| @@ -770,7 +770,7 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook | ||||
|  *         because no translation to SLINEAR audio was required. | ||||
|  * Part_3: Translate end_frame's audio back into the format of start frame if necessary.  This | ||||
|  *         is only necessary if manipulation of middle_frame occurred. | ||||
|  *          | ||||
|  * | ||||
|  * \param chan Channel that the list is coming off of | ||||
|  * \param audiohook_list List of audiohooks | ||||
|  * \param direction Direction frame is coming in from | ||||
| @@ -929,9 +929,9 @@ void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook) | ||||
| 	wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000)); | ||||
| 	ts.tv_sec = wait.tv_sec; | ||||
| 	ts.tv_nsec = wait.tv_usec * 1000; | ||||
| 	 | ||||
|  | ||||
| 	ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts); | ||||
| 	 | ||||
|  | ||||
| 	return; | ||||
| } | ||||
|  | ||||
|   | ||||
		Reference in New Issue
	
	Block a user