[asterisk-commits] dvossel: branch dvossel/fixtheworld_phase1_step3 r300519 - in /team/dvossel/f...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 4 21:43:07 UTC 2011
Author: dvossel
Date: Tue Jan 4 15:43:03 2011
New Revision: 300519
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=300519
Log:
changes required to get audiohook.c to compile with ast_format conversion
Modified:
team/dvossel/fixtheworld_phase1_step3/include/asterisk/audiohook.h
team/dvossel/fixtheworld_phase1_step3/include/asterisk/callerid.h
team/dvossel/fixtheworld_phase1_step3/include/asterisk/slinfactory.h
team/dvossel/fixtheworld_phase1_step3/main/audiohook.c
Modified: team/dvossel/fixtheworld_phase1_step3/include/asterisk/audiohook.h
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/include/asterisk/audiohook.h?view=diff&rev=300519&r1=300518&r2=300519
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/include/asterisk/audiohook.h (original)
+++ team/dvossel/fixtheworld_phase1_step3/include/asterisk/audiohook.h Tue Jan 4 15:43:03 2011
@@ -30,7 +30,6 @@
/* these two are used in struct ast_audiohook */
#include "asterisk/lock.h"
#include "asterisk/linkedlists.h"
-#include "asterisk/frame_defs.h"
#include "asterisk/slinfactory.h"
enum ast_audiohook_type {
@@ -104,7 +103,7 @@
struct ast_slinfactory write_factory; /*!< Factory where frames written to the channel will go through */
struct timeval read_time; /*!< Last time read factory was fed */
struct timeval write_time; /*!< Last time write factory was fed */
- int format; /*!< Format translation path is setup as */
+ struct ast_format format; /*!< Format translation path is setup as */
struct ast_trans_pvt *trans_pvt; /*!< Translation path for reading frames */
ast_audiohook_manipulate_callback manipulate_callback; /*!< Manipulation callback */
struct ast_audiohook_options options; /*!< Applicable options */
@@ -142,7 +141,7 @@
* \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);
/*! \brief Attach audiohook to channel
* \param chan Channel
Modified: team/dvossel/fixtheworld_phase1_step3/include/asterisk/callerid.h
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/include/asterisk/callerid.h?view=diff&rev=300519&r1=300518&r2=300519
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/include/asterisk/callerid.h (original)
+++ team/dvossel/fixtheworld_phase1_step3/include/asterisk/callerid.h Tue Jan 4 15:43:03 2011
@@ -101,7 +101,7 @@
* \return It returns the size
* (in bytes) of the data (if it returns a size of 0, there is probably an error)
*/
-int callerid_generate(unsigned char *buf, const char *number, const char *name, int flags, int callwaiting, format_t codec);
+int callerid_generate(unsigned char *buf, const char *number, const char *name, int flags, int callwaiting, struct ast_format *codec);
/*! \brief Create a callerID state machine
* \param cid_signalling Type of signalling in use
@@ -124,7 +124,7 @@
* \retval 0 for "needs more samples"
* \retval 1 if the CallerID spill reception is complete.
*/
-int callerid_feed(struct callerid_state *cid, unsigned char *ubuf, int samples, format_t codec);
+int callerid_feed(struct callerid_state *cid, unsigned char *ubuf, int samples, struct ast_format *codec);
/*! \brief Read samples into the state machine.
* \param cid Which state machine to act upon
@@ -138,7 +138,7 @@
* \retval 0 for "needs more samples"
* \retval 1 if the CallerID spill reception is complete.
*/
-int callerid_feed_jp(struct callerid_state *cid, unsigned char *ubuf, int samples, format_t codec);
+int callerid_feed_jp(struct callerid_state *cid, unsigned char *ubuf, int samples, struct ast_format *codec);
/*! \brief Extract info out of callerID state machine. Flags are listed above
* \param cid Callerid state machine to act upon
@@ -177,7 +177,7 @@
* \details
* Acts like callerid_generate except uses an asterisk format callerid string.
*/
-int ast_callerid_generate(unsigned char *buf, const char *name, const char *number, format_t codec);
+int ast_callerid_generate(unsigned char *buf, const char *name, const char *number, struct ast_format *codec);
/*!
* \brief Generate message waiting indicator
@@ -192,13 +192,13 @@
* \see callerid_generate() for more info as it uses the same encoding
* \version 1.6.1 changed mdmf parameter to type, added name, number and flags for caller id message generation
*/
-int ast_callerid_vmwi_generate(unsigned char *buf, int active, int type, format_t codec, const char *name,
+int ast_callerid_vmwi_generate(unsigned char *buf, int active, int type, struct ast_format *codec, const char *name,
const char *number, int flags);
/*! \brief Generate Caller-ID spill but in a format suitable for Call Waiting(tm)'s Caller*ID(tm)
* \see ast_callerid_generate() for other details
*/
-int ast_callerid_callwaiting_generate(unsigned char *buf, const char *name, const char *number, format_t codec);
+int ast_callerid_callwaiting_generate(unsigned char *buf, const char *name, const char *number, struct ast_format *codec);
/*! \brief Destructively parse inbuf into name and location (or number)
* \details
@@ -223,7 +223,7 @@
* \param codec Which codec (AST_FORMAT_ALAW or AST_FORMAT_ULAW)
* \return Returns -1 on error (if len is less than 2400), 0 on success.
*/
-int ast_gen_cas(unsigned char *outbuf, int sas, int len, format_t codec);
+int ast_gen_cas(unsigned char *outbuf, int sas, int len, struct ast_format *codec);
/*!
* \brief Shrink a phone number in place to just digits (more accurately it just removes ()'s, .'s, and -'s...
Modified: team/dvossel/fixtheworld_phase1_step3/include/asterisk/slinfactory.h
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/include/asterisk/slinfactory.h?view=diff&rev=300519&r1=300518&r2=300519
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/include/asterisk/slinfactory.h (original)
+++ team/dvossel/fixtheworld_phase1_step3/include/asterisk/slinfactory.h Tue Jan 4 15:43:03 2011
@@ -37,8 +37,8 @@
short *offset; /*!< Offset into the hold where audio begins */
size_t holdlen; /*!< Number of samples currently in the hold */
unsigned int size; /*!< Number of samples currently in the factory */
- format_t format; /*!< Current format the translation path is converting from */
- format_t output_format; /*!< The output format desired */
+ struct ast_format format; /*!< Current format the translation path is converting from */
+ struct ast_format output_format; /*!< The output format desired */
};
/*!
Modified: team/dvossel/fixtheworld_phase1_step3/main/audiohook.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/main/audiohook.c?view=diff&rev=300519&r1=300518&r2=300519
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/main/audiohook.c (original)
+++ team/dvossel/fixtheworld_phase1_step3/main/audiohook.c Tue Jan 4 15:43:03 2011
@@ -40,7 +40,7 @@
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 @@
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 @@
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 @@
* \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(&audiohook->format, 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 @@
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(&in_translate->format, &frame->subclass.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(&frame->subclass.format, &in_translate->format);
}
if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
return frame;
@@ -707,16 +709,16 @@
/* ---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(&start_frame->subclass.format, &out_translate->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))) {
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