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Expand codec bitfield from 32 bits to 64 bits.

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Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2009-11-04 14:05:12 +00:00
parent 6a50e7a031
commit d8e0c58437
124 changed files with 1777 additions and 1366 deletions
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24
main/audiohook.c
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@@ -40,7 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
int format;
format_t format;
};
struct ast_audiohook_list {
@@ -173,7 +173,7 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
short buf[samples];
struct ast_frame frame = {
.frametype = AST_FRAME_VOICE,
.subclass = AST_FORMAT_SLINEAR,
.subclass.codec = AST_FORMAT_SLINEAR,
.data.ptr = buf,
.datalen = sizeof(buf),
.samples = samples,
@@ -200,7 +200,7 @@ 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 = AST_FORMAT_SLINEAR,
.subclass.codec = AST_FORMAT_SLINEAR,
.data.ptr = NULL,
.datalen = sizeof(buf1),
.samples = samples,
@@ -292,7 +292,7 @@ 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, int format)
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 *read_frame = NULL, *final_frame = NULL;
@@ -576,13 +576,13 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
int samples = frame->samples;
/* If the frame coming in is not signed linear we have to send it through the in_translate path */
if (frame->subclass != AST_FORMAT_SLINEAR) {
if (in_translate->format != frame->subclass) {
if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
if (in_translate->format != frame->subclass.codec) {
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)))
if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass.codec)))
return frame;
in_translate->format = frame->subclass;
in_translate->format = frame->subclass.codec;
}
if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
return frame;
@@ -660,16 +660,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
/* Now we figure out what to do with our 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 != start_frame->subclass) {
if (out_translate->format != start_frame->subclass) {
if (end_frame->subclass.codec != start_frame->subclass.codec) {
if (out_translate->format != start_frame->subclass.codec) {
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, AST_FORMAT_SLINEAR))) {
if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass.codec, AST_FORMAT_SLINEAR))) {
/* 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;
out_translate->format = start_frame->subclass.codec;
}
/* 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))) {