Asterisk media architecture conversion - no more format bitfields

This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 17 insertions, 15 deletions

diff --git a/main/audiohook.c b/main/audiohook.c
index ad977ecbc..23650a79f 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -40,7 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 struct ast_audiohook_translate {
 	struct ast_trans_pvt *trans_pvt;
-	format_t format;
+	struct ast_format format;
 };
 
 struct ast_audiohook_list {
@@ -185,11 +185,11 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
 	short buf[samples];
 	struct ast_frame frame = {
 		.frametype = AST_FRAME_VOICE,
-		.subclass.codec = AST_FORMAT_SLINEAR,
 		.data.ptr = buf,
 		.datalen = sizeof(buf),
 		.samples = samples,
 	};
+	ast_format_set(&frame.subclass.format, AST_FORMAT_SLINEAR, 0);
 
 	/* Ensure the factory is able to give us the samples we want */
 	if (samples > ast_slinfactory_available(factory))
@@ -212,11 +212,11 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
 	short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
 	struct ast_frame frame = {
 		.frametype = AST_FRAME_VOICE,
-		.subclass.codec = AST_FORMAT_SLINEAR,
 		.data.ptr = NULL,
 		.datalen = sizeof(buf1),
 		.samples = samples,
 	};
+	ast_format_set(&frame.subclass.format, AST_FORMAT_SLINEAR, 0);
 
 	/* Make sure both factories have the required samples */
 	usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
@@ -304,23 +304,24 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
  * \param format Format of frame remote side wants back
  * \return Returns frame on success, NULL on failure
  */
-struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, format_t format)
+struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
 {
 	struct ast_frame *read_frame = NULL, *final_frame = NULL;
+	struct ast_format tmp_fmt;
 
 	if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
 		return NULL;
 
 	/* If they don't want signed linear back out, we'll have to send it through the translation path */
-	if (format != AST_FORMAT_SLINEAR) {
+	if (format->id != AST_FORMAT_SLINEAR) {
 		/* Rebuild translation path if different format then previously */
-		if (audiohook->format != format) {
+		if (ast_format_cmp(format, &audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
 			if (audiohook->trans_pvt) {
 				ast_translator_free_path(audiohook->trans_pvt);
 				audiohook->trans_pvt = NULL;
 			}
 			/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
-			if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
+			if (!(audiohook->trans_pvt = ast_translator_build_path(format, ast_format_set(&tmp_fmt, AST_FORMAT_SLINEAR, 0)))) {
 				ast_frfree(read_frame);
 				return NULL;
 			}
@@ -619,17 +620,18 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 	struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
 	struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
 	struct ast_audiohook *audiohook = NULL;
+	struct ast_format tmp_fmt;
 	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) {
+	if (frame->subclass.format.id != AST_FORMAT_SLINEAR) {
+		if (ast_format_cmp(&frame->subclass.format, &in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
 			if (in_translate->trans_pvt)
 				ast_translator_free_path(in_translate->trans_pvt);
-			if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass.codec)))
+			if (!(in_translate->trans_pvt = ast_translator_build_path(ast_format_set(&tmp_fmt, AST_FORMAT_SLINEAR, 0), &frame->subclass.format)))
 				return frame;
-			in_translate->format = frame->subclass.codec;
+			ast_format_copy(&in_translate->format, &frame->subclass.format);
 		}
 		if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
 			return frame;
@@ -707,16 +709,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 	/* ---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) {
-			if (out_translate->format != start_frame->subclass.codec) {
+		if (ast_format_cmp(&end_frame->subclass.format, &start_frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
+			if (ast_format_cmp(&out_translate->format, &start_frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
 				if (out_translate->trans_pvt)
 					ast_translator_free_path(out_translate->trans_pvt);
-				if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass.codec, AST_FORMAT_SLINEAR))) {
+				if (!(out_translate->trans_pvt = ast_translator_build_path(&start_frame->subclass.format, ast_format_set(&tmp_fmt, AST_FORMAT_SLINEAR, 0)))) {
 					/* We can't transcode this... drop our middle frame and return the original */
 					ast_frfree(middle_frame);
 					return start_frame;
 				}
-				out_translate->format = start_frame->subclass.codec;
+				ast_format_copy(&out_translate->format, &start_frame->subclass.format);
 			}
 			/* Transcode from our middle (signed linear) frame to new format of the frame that came in */
 			if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {