media formats: re-architect handling of media for performance improvements

In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
 1. Asterisk was limited in how many formats it could handle.
 2. Formats, being a bit field, could not include any attribute information.
    A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.

Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.

Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.

Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.

Generally, the new philosophy for handling formats is as follows:
 * The ast_format structure is reference counted. This removed a large amount
   of the memory allocations and copying that was done in prior versions.
 * In order to prevent race conditions while keeping things performant, the
   ast_format structure is immutable by convention and lock-free. Violate this
   tenet at your peril!
 * Because formats are reference counted, codecs are also reference counted.
   The Asterisk core generally provides built-in codecs and caches the
   ast_format structures created to represent them. Generally, to prevent
   inordinate amounts of module reference bumping, codecs and formats can be
   added at run-time but cannot be removed.
 * All compatibility with the bit field representation of codecs/formats has
   been moved to a compatibility API. The primary user of this representation
   is chan_iax2, which must continue to maintain its bit-field usage of formats
   for interoperability concerns.
 * When a format is negotiated with attributes, or when a format cannot be
   represented by one of the cached formats, a new format object is created or
   cloned from an existing format. That format may have the same codec
   underlying it, but is a different format than a version of the format with
   different attributes or without attributes.
 * While formats are reference counted objects, the reference count maintained
   on the format should be manipulated with care. Formats are generally cached
   and will persist for the lifetime of Asterisk and do not explicitly need
   to have their lifetime modified. An exception to this is when the user of a
   format does not know where the format came from *and* the user may outlive
   the provider of the format. This occurs, for example, when a format is read
   from a channel: the channel may have a format with attributes (hence,
   non-cached) and the user of the format may last longer than the channel (if
   the reference to the channel is released prior to the format's reference).

For more information on this work, see the API design notes:
  https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite

Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.

There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).

Reviews:
 https://reviewboard.asterisk.org/r/3814
 https://reviewboard.asterisk.org/r/3808
 https://reviewboard.asterisk.org/r/3805
 https://reviewboard.asterisk.org/r/3803
 https://reviewboard.asterisk.org/r/3801
 https://reviewboard.asterisk.org/r/3798
 https://reviewboard.asterisk.org/r/3800
 https://reviewboard.asterisk.org/r/3794
 https://reviewboard.asterisk.org/r/3793
 https://reviewboard.asterisk.org/r/3792
 https://reviewboard.asterisk.org/r/3791
 https://reviewboard.asterisk.org/r/3790
 https://reviewboard.asterisk.org/r/3789
 https://reviewboard.asterisk.org/r/3788
 https://reviewboard.asterisk.org/r/3787
 https://reviewboard.asterisk.org/r/3786
 https://reviewboard.asterisk.org/r/3784
 https://reviewboard.asterisk.org/r/3783
 https://reviewboard.asterisk.org/r/3778
 https://reviewboard.asterisk.org/r/3774
 https://reviewboard.asterisk.org/r/3775
 https://reviewboard.asterisk.org/r/3772
 https://reviewboard.asterisk.org/r/3761
 https://reviewboard.asterisk.org/r/3754
 https://reviewboard.asterisk.org/r/3753
 https://reviewboard.asterisk.org/r/3751
 https://reviewboard.asterisk.org/r/3750
 https://reviewboard.asterisk.org/r/3748
 https://reviewboard.asterisk.org/r/3747
 https://reviewboard.asterisk.org/r/3746
 https://reviewboard.asterisk.org/r/3742
 https://reviewboard.asterisk.org/r/3740
 https://reviewboard.asterisk.org/r/3739
 https://reviewboard.asterisk.org/r/3738
 https://reviewboard.asterisk.org/r/3737
 https://reviewboard.asterisk.org/r/3736
 https://reviewboard.asterisk.org/r/3734
 https://reviewboard.asterisk.org/r/3722
 https://reviewboard.asterisk.org/r/3713
 https://reviewboard.asterisk.org/r/3703
 https://reviewboard.asterisk.org/r/3689
 https://reviewboard.asterisk.org/r/3687
 https://reviewboard.asterisk.org/r/3674
 https://reviewboard.asterisk.org/r/3671
 https://reviewboard.asterisk.org/r/3667
 https://reviewboard.asterisk.org/r/3665
 https://reviewboard.asterisk.org/r/3625
 https://reviewboard.asterisk.org/r/3602
 https://reviewboard.asterisk.org/r/3519
 https://reviewboard.asterisk.org/r/3518
 https://reviewboard.asterisk.org/r/3516
 https://reviewboard.asterisk.org/r/3515
 https://reviewboard.asterisk.org/r/3512
 https://reviewboard.asterisk.org/r/3506
 https://reviewboard.asterisk.org/r/3413
 https://reviewboard.asterisk.org/r/3410
 https://reviewboard.asterisk.org/r/3387
 https://reviewboard.asterisk.org/r/3388
 https://reviewboard.asterisk.org/r/3389
 https://reviewboard.asterisk.org/r/3390
 https://reviewboard.asterisk.org/r/3321
 https://reviewboard.asterisk.org/r/3320
 https://reviewboard.asterisk.org/r/3319
 https://reviewboard.asterisk.org/r/3318
 https://reviewboard.asterisk.org/r/3266
 https://reviewboard.asterisk.org/r/3265
 https://reviewboard.asterisk.org/r/3234
 https://reviewboard.asterisk.org/r/3178

ASTERISK-23114 #close
Reported by: mjordan
  media_formats_translation_core.diff uploaded by kharwell (License 6464)
  rb3506.diff uploaded by mjordan (License 6283)
  media_format_app_file.diff uploaded by kharwell (License 6464) 
  misc-2.diff uploaded by file (License 5000)
  chan_mild-3.diff uploaded by file (License 5000) 
  chan_obscure.diff uploaded by file (License 5000) 
  jingle.diff uploaded by file (License 5000) 
  funcs.diff uploaded by file (License 5000) 
  formats.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  bridges.diff uploaded by file (License 5000) 
  mf-codecs-2.diff uploaded by file (License 5000) 
  mf-app_fax.diff uploaded by file (License 5000) 
  mf-apps-3.diff uploaded by file (License 5000) 
  media-formats-3.diff uploaded by file (License 5000) 

ASTERISK-23715
  rb3713.patch uploaded by coreyfarrell (License 5909)
  rb3689.patch uploaded by mjordan (License 6283)
  
ASTERISK-23957
  rb3722.patch uploaded by mjordan (License 6283) 
  mf-attributes-3.diff uploaded by file (License 5000) 

ASTERISK-23958
Tested by: jrose
  rb3822.patch uploaded by coreyfarrell (License 5909) 
  rb3800.patch uploaded by jrose (License 6182)
  chan_sip.diff uploaded by mjordan (License 6283) 
  rb3747.patch uploaded by jrose (License 6182)

ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
  sip_cleanup.diff uploaded by opticron (License 6273)
  chan_sip_caps.diff uploaded by mjordan (License 6283) 
  rb3751.patch uploaded by coreyfarrell (License 5909) 
  chan_sip-3.diff uploaded by file (License 5000) 

ASTERISK-23960 #close
Tested by: opticron
  direct_media.diff uploaded by opticron (License 6273) 
  pjsip-direct-media.diff uploaded by file (License 5000) 
  format_cap_remove.diff uploaded by opticron (License 6273) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  chan_pjsip-2.diff uploaded by file (License 5000) 

ASTERISK-23966 #close
Tested by: rmudgett
  rb3803.patch uploaded by rmudgetti (License 5621)
  chan_dahdi.diff uploaded by file (License 5000) 
  
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
  rb3814.patch uploaded by rmudgett (License 5621) 
  moh_cleanup.diff uploaded by opticron (License 6273) 
  bridge_leak.diff uploaded by opticron (License 6273) 
  translate.diff uploaded by file (License 5000) 
  rb3795.patch uploaded by rmudgett (License 5621) 
  tls_fix.diff uploaded by mjordan (License 6283) 
  fax-mf-fix-2.diff uploaded by file (License 5000) 
  rtp_transfer_stuff uploaded by mjordan (License 6283) 
  rb3787.patch uploaded by rmudgett (License 5621) 
  media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) 
  format_cache_case_fix.diff uploaded by opticron (License 6273) 
  rb3774.patch uploaded by rmudgett (License 5621) 
  rb3775.patch uploaded by rmudgett (License 5621) 
  rtp_engine_fix.diff uploaded by opticron (License 6273) 
  rtp_crash_fix.diff uploaded by opticron (License 6273) 
  rb3753.patch uploaded by mjordan (License 6283) 
  rb3750.patch uploaded by mjordan (License 6283) 
  rb3748.patch uploaded by rmudgett (License 5621) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  rb3740.patch uploaded by mjordan (License 6283) 
  rb3739.patch uploaded by mjordan (License 6283) 
  rb3734.patch uploaded by mjordan (License 6283) 
  rb3689.patch uploaded by mjordan (License 6283) 
  rb3674.patch uploaded by coreyfarrell (License 5909) 
  rb3671.patch uploaded by coreyfarrell (License 5909) 
  rb3667.patch uploaded by coreyfarrell (License 5909) 
  rb3665.patch uploaded by mjordan (License 6283) 
  rb3625.patch uploaded by coreyfarrell (License 5909) 
  rb3602.patch uploaded by coreyfarrell (License 5909) 
  format_compatibility-2.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3

21 files changed, 803 insertions, 285 deletions

diff --git a/codecs/codec_a_mu.c b/codecs/codec_a_mu.c
index 470f363e7..c21c706f5 100644
--- a/codecs/codec_a_mu.c
+++ b/codecs/codec_a_mu.c
@@ -80,6 +80,17 @@ static int ulawtoalaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 
 static struct ast_translator alawtoulaw = {
 	.name = "alawtoulaw",
+	.src_codec = {
+		.name = "alaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "ulaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "ulaw",
 	.framein = alawtoulaw_framein,
 	.sample = alaw_sample,
 	.buffer_samples = BUFFER_SAMPLES,
@@ -88,6 +99,17 @@ static struct ast_translator alawtoulaw = {
 
 static struct ast_translator ulawtoalaw = {
 	.name = "ulawtoalaw",
+	.src_codec = {
+		.name = "ulaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "alaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "alaw",
 	.framein = ulawtoalaw_framein,
 	.sample = ulaw_sample,
 	.buffer_samples = BUFFER_SAMPLES,
@@ -111,23 +133,19 @@ static int load_module(void)
 	int res;
 	int x;
 
-	ast_format_set(&alawtoulaw.src_format, AST_FORMAT_ALAW, 0);
-	ast_format_set(&alawtoulaw.dst_format, AST_FORMAT_ULAW, 0);
-
-	ast_format_set(&ulawtoalaw.src_format, AST_FORMAT_ULAW, 0);
-	ast_format_set(&ulawtoalaw.dst_format, AST_FORMAT_ALAW, 0);
-
 	for (x=0;x<256;x++) {
 		mu2a[x] = AST_LIN2A(AST_MULAW(x));
 		a2mu[x] = AST_LIN2MU(AST_ALAW(x));
 	}
+
 	res = ast_register_translator(&alawtoulaw);
-	if (!res)
-		res = ast_register_translator(&ulawtoalaw);
-	else
-		ast_unregister_translator(&alawtoulaw);
-	if (res)
+	res |= ast_register_translator(&ulawtoalaw);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
diff --git a/codecs/codec_adpcm.c b/codecs/codec_adpcm.c
index c48eb1ab2..03d2a4ab3 100644
--- a/codecs/codec_adpcm.c
+++ b/codecs/codec_adpcm.c
@@ -290,6 +290,17 @@ static struct ast_frame *lintoadpcm_frameout(struct ast_trans_pvt *pvt)
 
 static struct ast_translator adpcmtolin = {
 	.name = "adpcmtolin",
+	.src_codec = {
+		.name = "adpcm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.framein = adpcmtolin_framein,
 	.sample = adpcm_sample,
 	.desc_size = sizeof(struct adpcm_decoder_pvt),
@@ -299,6 +310,17 @@ static struct ast_translator adpcmtolin = {
 
 static struct ast_translator lintoadpcm = {
 	.name = "lintoadpcm",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "adpcm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "adpcm",
 	.framein = lintoadpcm_framein,
 	.frameout = lintoadpcm_frameout,
 	.sample = slin8_sample,
@@ -307,12 +329,6 @@ static struct ast_translator lintoadpcm = {
 	.buf_size = BUFFER_SAMPLES/ 2,	/* 2 samples per byte */
 };
 
-/*! \brief standard module glue */
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res;
@@ -325,26 +341,20 @@ static int unload_module(void)
 
 static int load_module(void)
 {
-	int res;
-
-	ast_format_set(&adpcmtolin.src_format, AST_FORMAT_ADPCM, 0);
-	ast_format_set(&adpcmtolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintoadpcm.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintoadpcm.dst_format, AST_FORMAT_ADPCM, 0);
+	int res = 0;
 
 	res = ast_register_translator(&adpcmtolin);
-	if (!res)
-		res = ast_register_translator(&lintoadpcm);
-	else
-		ast_unregister_translator(&adpcmtolin);
-	if (res)
+	res |= ast_register_translator(&lintoadpcm);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Adaptive Differential PCM Coder/Decoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_alaw.c b/codecs/codec_alaw.c
index 34a71388e..2ca577d7f 100644
--- a/codecs/codec_alaw.c
+++ b/codecs/codec_alaw.c
@@ -77,6 +77,17 @@ static int lintoalaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 
 static struct ast_translator alawtolin = {
 	.name = "alawtolin",
+	.src_codec = {
+		.name = "alaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.framein = alawtolin_framein,
 	.sample = alaw_sample,
 	.buffer_samples = BUFFER_SAMPLES,
@@ -84,20 +95,24 @@ static struct ast_translator alawtolin = {
 };
 
 static struct ast_translator lintoalaw = {
-	"lintoalaw",
+	.name = "lintoalaw",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "alaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "alaw",
 	.framein = lintoalaw_framein,
 	.sample = slin8_sample,
 	.buffer_samples = BUFFER_SAMPLES,
 	.buf_size = BUFFER_SAMPLES,
 };
 
-/*! \brief standard module stuff */
-
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res;
@@ -112,24 +127,18 @@ static int load_module(void)
 {
 	int res;
 
-	ast_format_set(&lintoalaw.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintoalaw.dst_format, AST_FORMAT_ALAW, 0);
-
-	ast_format_set(&alawtolin.src_format, AST_FORMAT_ALAW, 0);
-	ast_format_set(&alawtolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
 	res = ast_register_translator(&alawtolin);
-	if (!res)
-		res = ast_register_translator(&lintoalaw);
-	else
-		ast_unregister_translator(&alawtolin);
-	if (res)
+	res |= ast_register_translator(&lintoalaw);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "A-law Coder/Decoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_dahdi.c b/codecs/codec_dahdi.c
index caaf99cdf..64f220eef 100644
--- a/codecs/codec_dahdi.c
+++ b/codecs/codec_dahdi.c
@@ -51,6 +51,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/utils.h"
 #include "asterisk/linkedlists.h"
 #include "asterisk/ulaw.h"
+#include "asterisk/format_compatibility.h"
 
 #define BUFFER_SIZE 8000
 
@@ -58,17 +59,29 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define G729_SAMPLES 160
 #define ULAW_SAMPLES 160
 
+/* Defines from DAHDI. */
 #ifndef DAHDI_FORMAT_MAX_AUDIO
+/*! G.723.1 compression */
 #define DAHDI_FORMAT_G723_1    (1 << 0)
+/*! GSM compression */
 #define DAHDI_FORMAT_GSM       (1 << 1)
+/*! Raw mu-law data (G.711) */
 #define DAHDI_FORMAT_ULAW      (1 << 2)
+/*! Raw A-law data (G.711) */
 #define DAHDI_FORMAT_ALAW      (1 << 3)
+/*! ADPCM (G.726, 32kbps) */
 #define DAHDI_FORMAT_G726      (1 << 4)
+/*! ADPCM (IMA) */
 #define DAHDI_FORMAT_ADPCM     (1 << 5)
+/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
 #define DAHDI_FORMAT_SLINEAR   (1 << 6)
+/*! LPC10, 180 samples/frame */
 #define DAHDI_FORMAT_LPC10     (1 << 7)
+/*! G.729A audio */
 #define DAHDI_FORMAT_G729A     (1 << 8)
+/*! SpeeX Free Compression */
 #define DAHDI_FORMAT_SPEEX     (1 << 9)
+/*! iLBC Free Compression */
 #define DAHDI_FORMAT_ILBC      (1 << 10)
 #endif
 
@@ -78,6 +91,233 @@ static struct channel_usage {
 	int decoders;
 } channels;
 
+#if defined(NOT_NEEDED)
+/*!
+ * \internal
+ * \brief Convert DAHDI format bitfield to old Asterisk format bitfield.
+ * \since 13.0.0
+ *
+ * \param dahdi Bitfield from DAHDI to convert.
+ *
+ * \note They should be the same values but they don't have to be.
+ *
+ * \return Old Asterisk bitfield equivalent.
+ */
+static uint64_t bitfield_dahdi2ast(unsigned dahdi)
+{
+	uint64_t ast;
+
+	switch (dahdi) {
+	case DAHDI_FORMAT_G723_1:
+		ast = AST_FORMAT_G723;
+		break;
+	case DAHDI_FORMAT_GSM:
+		ast = AST_FORMAT_GSM;
+		break;
+	case DAHDI_FORMAT_ULAW:
+		ast = AST_FORMAT_ULAW;
+		break;
+	case DAHDI_FORMAT_ALAW:
+		ast = AST_FORMAT_ALAW;
+		break;
+	case DAHDI_FORMAT_G726:
+		ast = AST_FORMAT_G726_AAL2;
+		break;
+	case DAHDI_FORMAT_ADPCM:
+		ast = AST_FORMAT_ADPCM;
+		break;
+	case DAHDI_FORMAT_SLINEAR:
+		ast = AST_FORMAT_SLIN;
+		break;
+	case DAHDI_FORMAT_LPC10:
+		ast = AST_FORMAT_LPC10;
+		break;
+	case DAHDI_FORMAT_G729A:
+		ast = AST_FORMAT_G729;
+		break;
+	case DAHDI_FORMAT_SPEEX:
+		ast = AST_FORMAT_SPEEX;
+		break;
+	case DAHDI_FORMAT_ILBC:
+		ast = AST_FORMAT_ILBC;
+		break;
+	default:
+		ast = 0;
+		break;
+	}
+
+	return ast;
+}
+#endif	/* defined(NOT_NEEDED) */
+
+/*!
+ * \internal
+ * \brief Convert old Asterisk format bitfield to DAHDI format bitfield.
+ * \since 13.0.0
+ *
+ * \param ast Old Asterisk bitfield to convert.
+ *
+ * \note They should be the same values but they don't have to be.
+ *
+ * \return DAHDI bitfield equivalent.
+ */
+static unsigned bitfield_ast2dahdi(uint64_t ast)
+{
+	unsigned dahdi;
+
+	switch (ast) {
+	case AST_FORMAT_G723:
+		dahdi = DAHDI_FORMAT_G723_1;
+		break;
+	case AST_FORMAT_GSM:
+		dahdi = DAHDI_FORMAT_GSM;
+		break;
+	case AST_FORMAT_ULAW:
+		dahdi = DAHDI_FORMAT_ULAW;
+		break;
+	case AST_FORMAT_ALAW:
+		dahdi = DAHDI_FORMAT_ALAW;
+		break;
+	case AST_FORMAT_G726_AAL2:
+		dahdi = DAHDI_FORMAT_G726;
+		break;
+	case AST_FORMAT_ADPCM:
+		dahdi = DAHDI_FORMAT_ADPCM;
+		break;
+	case AST_FORMAT_SLIN:
+		dahdi = DAHDI_FORMAT_SLINEAR;
+		break;
+	case AST_FORMAT_LPC10:
+		dahdi = DAHDI_FORMAT_LPC10;
+		break;
+	case AST_FORMAT_G729:
+		dahdi = DAHDI_FORMAT_G729A;
+		break;
+	case AST_FORMAT_SPEEX:
+		dahdi = DAHDI_FORMAT_SPEEX;
+		break;
+	case AST_FORMAT_ILBC:
+		dahdi = DAHDI_FORMAT_ILBC;
+		break;
+	default:
+		dahdi = 0;
+		break;
+	}
+
+	return dahdi;
+}
+
+/*!
+ * \internal
+ * \brief Get the DAHDI codec by index.
+ * \since 13.0.0
+ *
+ * \param idx Codex index (0-31).
+ *
+ * \return Specified codec if exists otherwise NULL.
+ */
+static const struct ast_codec *get_dahdi_codec(unsigned idx)
+{
+	const struct ast_codec *codec;
+
+	static const struct ast_codec dahdi_g723_1 = {
+		.name = "g723",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_gsm = {
+		.name = "gsm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_ulaw = {
+		.name = "ulaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_alaw = {
+		.name = "alaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_g726 = {
+		.name = "g726",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_adpcm = {
+		.name = "adpcm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_slinear = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_lpc10 = {
+		.name = "lpc10",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_g729a = {
+		.name = "g729",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_speex = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+	static const struct ast_codec dahdi_ilbc = {
+		.name = "ilbc",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	};
+
+	switch (1UL << idx) {
+	case DAHDI_FORMAT_G723_1:
+		codec = &dahdi_g723_1;
+		break;
+	case DAHDI_FORMAT_GSM:
+		codec = &dahdi_gsm;
+		break;
+	case DAHDI_FORMAT_ULAW:
+		codec = &dahdi_ulaw;
+		break;
+	case DAHDI_FORMAT_ALAW:
+		codec = &dahdi_alaw;
+		break;
+	case DAHDI_FORMAT_G726:
+		codec = &dahdi_g726;
+		break;
+	case DAHDI_FORMAT_ADPCM:
+		codec = &dahdi_adpcm;
+		break;
+	case DAHDI_FORMAT_SLINEAR:
+		codec = &dahdi_slinear;
+		break;
+	case DAHDI_FORMAT_LPC10:
+		codec = &dahdi_lpc10;
+		break;
+	case DAHDI_FORMAT_G729A:
+		codec = &dahdi_g729a;
+		break;
+	case DAHDI_FORMAT_SPEEX:
+		codec = &dahdi_speex;
+		break;
+	case DAHDI_FORMAT_ILBC:
+		codec = &dahdi_ilbc;
+		break;
+	default:
+		codec = NULL;
+		break;
+	}
+
+	return codec;
+}
+
 static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 
 static struct ast_cli_entry cli[] = {
@@ -190,7 +430,7 @@ static int dahdi_encoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 {
 	struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
-	if (!f->subclass.format.id) {
+	if (!f->subclass.format) {
 		/* We're just faking a return for calculation purposes. */
 		dahdip->fake = 2;
 		pvt->samples = f->samples;
@@ -245,18 +485,16 @@ static struct ast_frame *dahdi_encoder_frameout(struct ast_trans_pvt *pvt)
 	int res;
 
 	if (2 == dahdip->fake) {
+		struct ast_frame frm = {
+			.frametype = AST_FRAME_VOICE,
+			.samples = dahdip->required_samples,
+			.src = pvt->t->name,
+		};
+
 		dahdip->fake = 1;
-		pvt->f.frametype = AST_FRAME_VOICE;
-		ast_format_clear(&pvt->f.subclass.format);
-		pvt->f.samples = dahdip->required_samples;
-		pvt->f.data.ptr = NULL;
-		pvt->f.offset = 0;
-		pvt->f.datalen = 0;
-		pvt->f.mallocd = 0;
 		pvt->samples = 0;
 
-		return ast_frisolate(&pvt->f);
-
+		return ast_frisolate(&frm);
 	} else if (1 == dahdip->fake) {
 		dahdip->fake = 0;
 		return NULL;
@@ -277,13 +515,7 @@ static struct ast_frame *dahdi_encoder_frameout(struct ast_trans_pvt *pvt)
 		}
 	} else {
 		pvt->f.datalen = res;
-		pvt->f.frametype = AST_FRAME_VOICE;
-		ast_format_copy(&pvt->f.subclass.format, &pvt->t->dst_format);
-		pvt->f.mallocd = 0;
-		pvt->f.offset = AST_FRIENDLY_OFFSET;
-		pvt->f.src = pvt->t->name;
-		pvt->f.data.ptr = pvt->outbuf.c;
-		pvt->f.samples = ast_codec_get_samples(&pvt->f);
+		pvt->f.samples = ast_codec_samples_count(&pvt->f);
 
 		dahdip->samples_written_to_hardware =
 		  (dahdip->samples_written_to_hardware >= pvt->f.samples) ?
@@ -302,7 +534,7 @@ static int dahdi_decoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 {
 	struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
-	if (!f->subclass.format.id) {
+	if (!f->subclass.format) {
 		/* We're just faking a return for calculation purposes. */
 		dahdip->fake = 2;
 		pvt->samples = f->samples;
@@ -327,16 +559,16 @@ static struct ast_frame *dahdi_decoder_frameout(struct ast_trans_pvt *pvt)
 	struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
 	if (2 == dahdip->fake) {
+		struct ast_frame frm = {
+			.frametype = AST_FRAME_VOICE,
+			.samples = dahdip->required_samples,
+			.src = pvt->t->name,
+		};
+
 		dahdip->fake = 1;
-		pvt->f.frametype = AST_FRAME_VOICE;
-		ast_format_clear(&pvt->f.subclass.format);
-		pvt->f.samples = dahdip->required_samples;
-		pvt->f.data.ptr = NULL;
-		pvt->f.offset = 0;
-		pvt->f.datalen = 0;
-		pvt->f.mallocd = 0;
 		pvt->samples = 0;
-		return ast_frisolate(&pvt->f);
+
+		return ast_frisolate(&frm);
 	} else if (1 == dahdip->fake) {
 		pvt->samples = 0;
 		dahdip->fake = 0;
@@ -370,12 +602,6 @@ static struct ast_frame *dahdi_decoder_frameout(struct ast_trans_pvt *pvt)
 			pvt->f.datalen = res;
 		}
 		pvt->datalen = 0;
-		pvt->f.frametype = AST_FRAME_VOICE;
-		ast_format_copy(&pvt->f.subclass.format, &pvt->t->dst_format);
-		pvt->f.mallocd = 0;
-		pvt->f.offset = AST_FRIENDLY_OFFSET;
-		pvt->f.src = pvt->t->name;
-		pvt->f.data.ptr = pvt->outbuf.c;
 		pvt->f.samples = res;
 		pvt->samples = 0;
 		dahdip->samples_written_to_hardware =
@@ -394,9 +620,9 @@ static void dahdi_destroy(struct ast_trans_pvt *pvt)
 {
 	struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
-	switch (ast_format_id_from_old_bitfield(dahdip->fmts.dstfmt)) {
-	case AST_FORMAT_G729A:
-	case AST_FORMAT_G723_1:
+	switch (dahdip->fmts.dstfmt) {
+	case DAHDI_FORMAT_G729A:
+	case DAHDI_FORMAT_G723_1:
 		ast_atomic_fetchadd_int(&channels.encoders, -1);
 		break;
 	default:
@@ -407,7 +633,39 @@ static void dahdi_destroy(struct ast_trans_pvt *pvt)
 	close(dahdip->fd);
 }
 
-static int dahdi_translate(struct ast_trans_pvt *pvt, struct ast_format *dst_format, struct ast_format *src_format)
+static struct ast_format *dahdi_format_to_cached(int format)
+{
+	switch (format) {
+	case DAHDI_FORMAT_G723_1:
+		return ast_format_g723;
+	case DAHDI_FORMAT_GSM:
+		return ast_format_gsm;
+	case DAHDI_FORMAT_ULAW:
+		return ast_format_ulaw;
+	case DAHDI_FORMAT_ALAW:
+		return ast_format_alaw;
+	case DAHDI_FORMAT_G726:
+		return ast_format_g726;
+	case DAHDI_FORMAT_ADPCM:
+		return ast_format_adpcm;
+	case DAHDI_FORMAT_SLINEAR:
+		return ast_format_slin;
+	case DAHDI_FORMAT_LPC10:
+		return ast_format_lpc10;
+	case DAHDI_FORMAT_G729A:
+		return ast_format_g729;
+	case DAHDI_FORMAT_SPEEX:
+		return ast_format_speex;
+	case DAHDI_FORMAT_ILBC:
+		return ast_format_ilbc;
+	}
+
+	/* This will never be reached */
+	ast_assert(0);
+	return NULL;
+}
+
+static int dahdi_translate(struct ast_trans_pvt *pvt, struct ast_codec *dst_codec, struct ast_codec *src_codec)
 {
 	/* Request translation through zap if possible */
 	int fd;
@@ -421,10 +679,13 @@ static int dahdi_translate(struct ast_trans_pvt *pvt, struct ast_format *dst_for
 		return -1;
 	}
 
-	dahdip->fmts.srcfmt = ast_format_to_old_bitfield(src_format);
-	dahdip->fmts.dstfmt = ast_format_to_old_bitfield(dst_format);
+	dahdip->fmts.srcfmt = bitfield_ast2dahdi(ast_format_compatibility_codec2bitfield(src_codec));
+	dahdip->fmts.dstfmt = bitfield_ast2dahdi(ast_format_compatibility_codec2bitfield(dst_codec));
+
+	ast_assert(pvt->f.subclass.format == NULL);
+	pvt->f.subclass.format = ao2_bump(dahdi_format_to_cached(dahdip->fmts.dstfmt));
 
-	ast_debug(1, "Opening transcoder channel from %s to %s.\n", ast_getformatname(src_format), ast_getformatname(dst_format));
+	ast_debug(1, "Opening transcoder channel from %s to %s.\n", src_codec->name, dst_codec->name);
 
 retry:
 	if (ioctl(fd, DAHDI_TC_ALLOCATE, &dahdip->fmts)) {
@@ -437,14 +698,14 @@ retry:
 			 * support for ULAW instead of signed linear and then
 			 * we'll just convert from ulaw to signed linear in
 			 * software. */
-			if (AST_FORMAT_SLINEAR == ast_format_id_from_old_bitfield(dahdip->fmts.srcfmt)) {
+			if (dahdip->fmts.srcfmt == DAHDI_FORMAT_SLINEAR) {
 				ast_debug(1, "Using soft_slin support on source\n");
 				dahdip->softslin = 1;
-				dahdip->fmts.srcfmt = ast_format_id_to_old_bitfield(AST_FORMAT_ULAW);
-			} else if (AST_FORMAT_SLINEAR == ast_format_id_from_old_bitfield(dahdip->fmts.dstfmt)) {
+				dahdip->fmts.srcfmt = DAHDI_FORMAT_ULAW;
+			} else if (dahdip->fmts.dstfmt == DAHDI_FORMAT_SLINEAR) {
 				ast_debug(1, "Using soft_slin support on destination\n");
 				dahdip->softslin = 1;
-				dahdip->fmts.dstfmt = ast_format_id_to_old_bitfield(AST_FORMAT_ULAW);
+				dahdip->fmts.dstfmt = DAHDI_FORMAT_ULAW;
 			}
 			tried_once = 1;
 			goto retry;
@@ -463,13 +724,13 @@ retry:
 
 	dahdip->fd = fd;
 
-	dahdip->required_samples = ((dahdip->fmts.dstfmt|dahdip->fmts.srcfmt) & (ast_format_id_to_old_bitfield(AST_FORMAT_G723_1))) ? G723_SAMPLES : G729_SAMPLES;
+	dahdip->required_samples = ((dahdip->fmts.dstfmt|dahdip->fmts.srcfmt) & (DAHDI_FORMAT_G723_1)) ? G723_SAMPLES : G729_SAMPLES;
 
-	switch (ast_format_id_from_old_bitfield(dahdip->fmts.dstfmt)) {
-	case AST_FORMAT_G729A:
+	switch (dahdip->fmts.dstfmt) {
+	case DAHDI_FORMAT_G729A:
 		ast_atomic_fetchadd_int(&channels.encoders, +1);
 		break;
-	case AST_FORMAT_G723_1:
+	case DAHDI_FORMAT_G723_1:
 		ast_atomic_fetchadd_int(&channels.encoders, +1);
 		break;
 	default:
@@ -483,8 +744,8 @@ retry:
 static int dahdi_new(struct ast_trans_pvt *pvt)
 {
 	return dahdi_translate(pvt,
-		&pvt->t->dst_format,
-		&pvt->t->src_format);
+		pvt->t->core_dst_codec,
+		pvt->t->core_src_codec);
 }
 
 static struct ast_frame *fakesrc_sample(void)
@@ -501,33 +762,37 @@ static struct ast_frame *fakesrc_sample(void)
 
 static int is_encoder(struct translator *zt)
 {
-	if ((zt->t.src_format.id == AST_FORMAT_ULAW) ||
-		(zt->t.src_format.id == AST_FORMAT_ALAW) ||
-		(zt->t.src_format.id == AST_FORMAT_SLINEAR)) {
+	if ((zt->t.core_src_codec->id == ast_format_get_codec_id(ast_format_ulaw)) ||
+		(zt->t.core_src_codec->id == ast_format_get_codec_id(ast_format_alaw)) ||
+		(zt->t.core_src_codec->id == ast_format_get_codec_id(ast_format_slin))) {
 		return 1;
 	} else {
 		return 0;
 	}
 }
 
-static int register_translator(int dst, int src)
+static int register_translator(unsigned dst, unsigned src)
 {
+	const struct ast_codec *dst_codec;
+	const struct ast_codec *src_codec;
 	struct translator *zt;
 	int res;
-	struct ast_format dst_format;
-	struct ast_format src_format;
 
-	ast_format_from_old_bitfield(&dst_format, (1 << dst));
-	ast_format_from_old_bitfield(&src_format, (1 << src));
+	dst_codec = get_dahdi_codec(dst);
+	src_codec = get_dahdi_codec(src);
+	if (!dst_codec || !src_codec) {
+		return -1;
+	}
 
 	if (!(zt = ast_calloc(1, sizeof(*zt)))) {
 		return -1;
 	}
 
-	snprintf((char *) (zt->t.name), sizeof(zt->t.name), "zap%sto%s",
-		 ast_getformatname(&src_format), ast_getformatname(&dst_format));
-	ast_format_copy(&zt->t.src_format, &src_format);
-	ast_format_copy(&zt->t.dst_format, &dst_format);
+	snprintf(zt->t.name, sizeof(zt->t.name), "dahdi_%s_to_%s",
+		src_codec->name, dst_codec->name);
+
+	memcpy(&zt->t.src_codec, src_codec, sizeof(*src_codec));
+	memcpy(&zt->t.dst_codec, dst_codec, sizeof(*dst_codec));
 	zt->t.buf_size = BUFFER_SIZE;
 	if (is_encoder(zt)) {
 		zt->t.framein = dahdi_encoder_framein;
@@ -557,17 +822,20 @@ static int register_translator(int dst, int src)
 	return res;
 }
 
-static void drop_translator(int dst, int src)
+static void drop_translator(unsigned dst, unsigned src)
 {
 	struct translator *cur;
 
 	AST_LIST_LOCK(&translators);
 	AST_LIST_TRAVERSE_SAFE_BEGIN(&translators, cur, entry) {
-		if (cur->t.src_format.id != ast_format_id_from_old_bitfield((1 << src)))
+		if (bitfield_ast2dahdi(ast_format_compatibility_codec2bitfield(cur->t.core_src_codec))
+			!= (1U << src)) {
 			continue;
-
-		if (cur->t.dst_format.id != ast_format_id_from_old_bitfield((1 << dst)))
+		}
+		if (bitfield_ast2dahdi(ast_format_compatibility_codec2bitfield(cur->t.core_dst_codec))
+			!= (1U << dst)) {
 			continue;
+		}
 
 		AST_LIST_REMOVE_CURRENT(entry);
 		ast_unregister_translator(&cur->t);
@@ -678,7 +946,6 @@ static int unload_module(void)
 
 static int load_module(void)
 {
-	ast_ulaw_init();
 	find_transcoders();
 	ast_cli_register_multiple(cli, ARRAY_LEN(cli));
 	return AST_MODULE_LOAD_SUCCESS;
diff --git a/codecs/codec_g722.c b/codecs/codec_g722.c
index 89641f175..1eba8067f 100644
--- a/codecs/codec_g722.c
+++ b/codecs/codec_g722.c
@@ -138,6 +138,17 @@ static int lintog722_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 
 static struct ast_translator g722tolin = {
 	.name = "g722tolin",
+	.src_codec = {
+		.name = "g722",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = g722tolin_new,	/* same for both directions */
 	.framein = g722tolin_framein,
 	.sample = g722_sample,
@@ -148,6 +159,17 @@ static struct ast_translator g722tolin = {
 
 static struct ast_translator lintog722 = {
 	.name = "lintog722",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "g722",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "g722",
 	.newpvt = lintog722_new,	/* same for both directions */
 	.framein = lintog722_framein,
 	.sample = slin8_sample,
@@ -158,6 +180,17 @@ static struct ast_translator lintog722 = {
 
 static struct ast_translator g722tolin16 = {
 	.name = "g722tolin16",
+	.src_codec = {
+		.name = "g722",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "slin16",
 	.newpvt = g722tolin16_new,	/* same for both directions */
 	.framein = g722tolin_framein,
 	.sample = g722_sample,
@@ -168,6 +201,17 @@ static struct ast_translator g722tolin16 = {
 
 static struct ast_translator lin16tog722 = {
 	.name = "lin16tog722",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "g722",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "g722",
 	.newpvt = lin16tog722_new,	/* same for both directions */
 	.framein = lintog722_framein,
 	.sample = slin16_sample,
@@ -176,11 +220,6 @@ static struct ast_translator lin16tog722 = {
 	.buf_size = BUFFER_SAMPLES,
 };
 
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res = 0;
@@ -197,18 +236,6 @@ static int load_module(void)
 {
 	int res = 0;
 
-	ast_format_set(&g722tolin.src_format, AST_FORMAT_G722, 0);
-	ast_format_set(&g722tolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintog722.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintog722.dst_format, AST_FORMAT_G722, 0);
-
-	ast_format_set(&g722tolin16.src_format, AST_FORMAT_G722, 0);
-	ast_format_set(&g722tolin16.dst_format, AST_FORMAT_SLINEAR16, 0);
-
-	ast_format_set(&lin16tog722.src_format, AST_FORMAT_SLINEAR16, 0);
-	ast_format_set(&lin16tog722.dst_format, AST_FORMAT_G722, 0);
-
 	res |= ast_register_translator(&g722tolin);
 	res |= ast_register_translator(&lintog722);
 	res |= ast_register_translator(&g722tolin16);
@@ -225,5 +252,4 @@ static int load_module(void)
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "ITU G.722-64kbps G722 Transcoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_g726.c b/codecs/codec_g726.c
index 72e77f955..a41039d53 100644
--- a/codecs/codec_g726.c
+++ b/codecs/codec_g726.c
@@ -785,6 +785,17 @@ static int lintog726_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 
 static struct ast_translator g726tolin = {
 	.name = "g726tolin",
+	.src_codec = {
+		.name = "g726",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = lintog726_new,	/* same for both directions */
 	.framein = g726tolin_framein,
 	.sample = g726_sample,
@@ -795,6 +806,17 @@ static struct ast_translator g726tolin = {
 
 static struct ast_translator lintog726 = {
 	.name = "lintog726",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "g726",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "g726",
 	.newpvt = lintog726_new,	/* same for both directions */
 	.framein = lintog726_framein,
 	.sample = slin8_sample,
@@ -805,6 +827,17 @@ static struct ast_translator lintog726 = {
 
 static struct ast_translator g726aal2tolin = {
 	.name = "g726aal2tolin",
+	.src_codec = {
+		.name = "g726aal2",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = lintog726_new,	/* same for both directions */
 	.framein = g726aal2tolin_framein,
 	.sample = g726_sample,
@@ -815,6 +848,17 @@ static struct ast_translator g726aal2tolin = {
 
 static struct ast_translator lintog726aal2 = {
 	.name = "lintog726aal2",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "g726aal2",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "g726aal2",
 	.newpvt = lintog726_new,	/* same for both directions */
 	.framein = lintog726aal2_framein,
 	.sample = slin8_sample,
@@ -823,11 +867,6 @@ static struct ast_translator lintog726aal2 = {
 	.buf_size = BUFFER_SAMPLES / 2,
 };
 
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res = 0;
@@ -845,18 +884,6 @@ static int load_module(void)
 {
 	int res = 0;
 
-	ast_format_set(&g726tolin.src_format, AST_FORMAT_G726, 0);
-	ast_format_set(&g726tolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintog726.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintog726.dst_format, AST_FORMAT_G726, 0);
-
-	ast_format_set(&g726aal2tolin.src_format, AST_FORMAT_G726_AAL2, 0);
-	ast_format_set(&g726aal2tolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintog726aal2.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintog726aal2.dst_format, AST_FORMAT_G726_AAL2, 0);
-
 	res |= ast_register_translator(&g726tolin);
 	res |= ast_register_translator(&lintog726);
 
@@ -874,5 +901,4 @@ static int load_module(void)
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "ITU G.726-32kbps G726 Transcoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c
index f42a5f1bd..49f672adb 100644
--- a/codecs/codec_gsm.c
+++ b/codecs/codec_gsm.c
@@ -168,7 +168,18 @@ static void gsm_destroy_stuff(struct ast_trans_pvt *pvt)
 }
 
 static struct ast_translator gsmtolin = {
-	.name = "gsmtolin", 
+	.name = "gsmtolin",
+	.src_codec = {
+		.name = "gsm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = gsm_new,
 	.framein = gsmtolin_framein,
 	.destroy = gsm_destroy_stuff,
@@ -179,7 +190,18 @@ static struct ast_translator gsmtolin = {
 };
 
 static struct ast_translator lintogsm = {
-	.name = "lintogsm", 
+	.name = "lintogsm",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "gsm",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "gsm",
 	.newpvt = gsm_new,
 	.framein = lintogsm_framein,
 	.frameout = lintogsm_frameout,
@@ -189,19 +211,12 @@ static struct ast_translator lintogsm = {
 	.buf_size = (BUFFER_SAMPLES * GSM_FRAME_LEN + GSM_SAMPLES - 1)/GSM_SAMPLES,
 };
 
-/*! \brief standard module glue */
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_translator(&lintogsm);
-	if (!res)
-		res = ast_unregister_translator(&gsmtolin);
+	res |= ast_unregister_translator(&gsmtolin);
 
 	return res;
 }
@@ -210,24 +225,18 @@ static int load_module(void)
 {
 	int res;
 
-	ast_format_set(&gsmtolin.src_format, AST_FORMAT_GSM, 0);
-	ast_format_set(&gsmtolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintogsm.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintogsm.dst_format, AST_FORMAT_GSM, 0);
-
 	res = ast_register_translator(&gsmtolin);
-	if (!res) 
-		res=ast_register_translator(&lintogsm);
-	else
-		ast_unregister_translator(&gsmtolin);
-	if (res) 
+	res |= ast_register_translator(&lintogsm);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "GSM Coder/Decoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_ilbc.c b/codecs/codec_ilbc.c
index 632169589..af23b906d 100644
--- a/codecs/codec_ilbc.c
+++ b/codecs/codec_ilbc.c
@@ -178,7 +178,18 @@ static struct ast_frame *lintoilbc_frameout(struct ast_trans_pvt *pvt)
 }
 
 static struct ast_translator ilbctolin = {
-	.name = "ilbctolin", 
+	.name = "ilbctolin",
+	.src_codec = {
+		.name = "ilbc",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = ilbctolin_new,
 	.framein = ilbctolin_framein,
 	.sample = ilbc_sample,
@@ -188,7 +199,18 @@ static struct ast_translator ilbctolin = {
 };
 
 static struct ast_translator lintoilbc = {
-	.name = "lintoilbc", 
+	.name = "lintoilbc",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "ilbc",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "ilbc",
 	.newpvt = lintoilbc_new,
 	.framein = lintoilbc_framein,
 	.frameout = lintoilbc_frameout,
@@ -211,20 +233,14 @@ static int load_module(void)
 {
 	int res;
 
-	ast_format_set(&ilbctolin.src_format, AST_FORMAT_ILBC, 0);
-	ast_format_set(&ilbctolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintoilbc.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintoilbc.dst_format, AST_FORMAT_ILBC, 0);
-
-
 	res = ast_register_translator(&ilbctolin);
-	if (!res) 
-		res=ast_register_translator(&lintoilbc);
-	else
-		ast_unregister_translator(&ilbctolin);
-	if (res)
+	res |= ast_register_translator(&lintoilbc);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
diff --git a/codecs/codec_lpc10.c b/codecs/codec_lpc10.c
index 5f2047a3f..0ef8856ff 100644
--- a/codecs/codec_lpc10.c
+++ b/codecs/codec_lpc10.c
@@ -196,7 +196,18 @@ static void lpc10_destroy(struct ast_trans_pvt *arg)
 }
 
 static struct ast_translator lpc10tolin = {
-	.name = "lpc10tolin", 
+	.name = "lpc10tolin",
+	.src_codec = {
+		.name = "lpc10",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = lpc10_dec_new,
 	.framein = lpc10tolin_framein,
 	.destroy = lpc10_destroy,
@@ -207,7 +218,18 @@ static struct ast_translator lpc10tolin = {
 };
 
 static struct ast_translator lintolpc10 = {
-	.name = "lintolpc10", 
+	.name = "lintolpc10",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "lpc10",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "lpc10",
 	.newpvt = lpc10_enc_new,
 	.framein = lintolpc10_framein,
 	.frameout = lintolpc10_frameout,
@@ -218,12 +240,6 @@ static struct ast_translator lintolpc10 = {
 	.buf_size = LPC10_BYTES_IN_COMPRESSED_FRAME * (1 + BUFFER_SAMPLES / LPC10_SAMPLES_PER_FRAME),
 };
 
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
-
 static int unload_module(void)
 {
 	int res;
@@ -238,24 +254,18 @@ static int load_module(void)
 {
 	int res;
 
-	ast_format_set(&lpc10tolin.src_format, AST_FORMAT_LPC10, 0);
-	ast_format_set(&lpc10tolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintolpc10.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintolpc10.dst_format, AST_FORMAT_LPC10, 0);
-
 	res = ast_register_translator(&lpc10tolin);
-	if (!res) 
-		res = ast_register_translator(&lintolpc10);
-	else
-		ast_unregister_translator(&lpc10tolin);
-	if (res)
+	res |= ast_register_translator(&lintolpc10);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "LPC10 2.4kbps Coder/Decoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/codec_resample.c b/codecs/codec_resample.c
index 29d95a798..26b1f0e08 100644
--- a/codecs/codec_resample.c
+++ b/codecs/codec_resample.c
@@ -42,32 +42,72 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 static struct ast_translator *translators;
 static int trans_size;
-static int id_list[] = {
-	AST_FORMAT_SLINEAR,
-	AST_FORMAT_SLINEAR12,
-	AST_FORMAT_SLINEAR16,
-	AST_FORMAT_SLINEAR24,
-	AST_FORMAT_SLINEAR32,
-	AST_FORMAT_SLINEAR44,
-	AST_FORMAT_SLINEAR48,
-	AST_FORMAT_SLINEAR96,
-	AST_FORMAT_SLINEAR192,
+static struct ast_codec codec_list[] = {
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 12000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 24000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 32000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 44100,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 48000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 96000,
+	},
+	{
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 192000,
+	},
 };
 
 static int resamp_new(struct ast_trans_pvt *pvt)
 {
 	int err;
 
-	if (!(pvt->pvt = speex_resampler_init(1, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
+	if (!(pvt->pvt = speex_resampler_init(1, pvt->t->src_codec.sample_rate, pvt->t->dst_codec.sample_rate, 5, &err))) {
 		return -1;
 	}
 
+	ast_assert(pvt->f.subclass.format == NULL);
+	pvt->f.subclass.format = ao2_bump(ast_format_cache_get_slin_by_rate(pvt->t->dst_codec.sample_rate));
+
 	return 0;
 }
 
 static void resamp_destroy(struct ast_trans_pvt *pvt)
 {
 	SpeexResamplerState *resamp_pvt = pvt->pvt;
+
 	speex_resampler_destroy(resamp_pvt);
 }
 
@@ -113,13 +153,13 @@ static int load_module(void)
 	int res = 0;
 	int x, y, idx = 0;
 
-	trans_size = ARRAY_LEN(id_list) * (ARRAY_LEN(id_list) - 1);
+	trans_size = ARRAY_LEN(codec_list) * (ARRAY_LEN(codec_list) - 1);
 	if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
 		return AST_MODULE_LOAD_FAILURE;
 	}
 
-	for (x = 0; x < ARRAY_LEN(id_list); x++) {
-		for (y = 0; y < ARRAY_LEN(id_list); y++) {
+	for (x = 0; x < ARRAY_LEN(codec_list); x++) {
+		for (y = 0; y < ARRAY_LEN(codec_list); y++) {
 			if (x == y) {
 				continue;
 			}
@@ -129,10 +169,10 @@ static int load_module(void)
 			translators[idx].desc_size = 0;
 			translators[idx].buffer_samples = (OUTBUF_SIZE / sizeof(int16_t));
 			translators[idx].buf_size = OUTBUF_SIZE;
-			ast_format_set(&translators[idx].src_format, id_list[x], 0);
-			ast_format_set(&translators[idx].dst_format, id_list[y], 0);
-			snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %dkhz -> %dkhz",
-				ast_format_rate(&translators[idx].src_format), ast_format_rate(&translators[idx].dst_format));
+			memcpy(&translators[idx].src_codec, &codec_list[x], sizeof(struct ast_codec));
+			memcpy(&translators[idx].dst_codec, &codec_list[y], sizeof(struct ast_codec));
+			snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %ukhz -> %ukhz",
+				translators[idx].src_codec.sample_rate, translators[idx].dst_codec.sample_rate);
 			res |= ast_register_translator(&translators[idx]);
 			idx++;
 		}
diff --git a/codecs/codec_speex.c b/codecs/codec_speex.c
index 72e948057..8c2c3f395 100644
--- a/codecs/codec_speex.c
+++ b/codecs/codec_speex.c
@@ -308,10 +308,14 @@ static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt)
 		} else {
 			tmp->silent_state = 1;
 			speex_bits_reset(&tmp->bits);
+
+/* BUGBUG need to setup a new static frame to prevent destroying the translators normal static frame. */
+			ao2_cleanup(pvt->f.subclass.format);
 			memset(&pvt->f, 0, sizeof(pvt->f));
 			pvt->f.frametype = AST_FRAME_CNG;
 			pvt->f.samples = samples;
 			/* XXX what now ? format etc... */
+/* BUGBUG should return ast_frisolate(setup local static frame) here */
 		}
 	}
 
@@ -341,7 +345,18 @@ static void lintospeex_destroy(struct ast_trans_pvt *arg)
 }
 
 static struct ast_translator speextolin = {
-	.name = "speextolin", 
+	.name = "speextolin",
+	.src_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.newpvt = speextolin_new,
 	.framein = speextolin_framein,
 	.destroy = speextolin_destroy,
@@ -354,6 +369,17 @@ static struct ast_translator speextolin = {
 
 static struct ast_translator lintospeex = {
 	.name = "lintospeex", 
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "speex",
 	.newpvt = lintospeex_new,
 	.framein = lintospeex_framein,
 	.frameout = lintospeex_frameout,
@@ -365,7 +391,18 @@ static struct ast_translator lintospeex = {
 };
 
 static struct ast_translator speexwbtolin16 = {
-	.name = "speexwbtolin16", 
+	.name = "speexwbtolin16",
+	.src_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	}, 
+	.format = "slin16",
 	.newpvt = speexwbtolin16_new,
 	.framein = speextolin_framein,
 	.destroy = speextolin_destroy,
@@ -377,7 +414,18 @@ static struct ast_translator speexwbtolin16 = {
 };
 
 static struct ast_translator lin16tospeexwb = {
-	.name = "lin16tospeexwb", 
+	.name = "lin16tospeexwb",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.dst_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 16000,
+	},
+	.format = "speex16",
 	.newpvt = lin16tospeexwb_new,
 	.framein = lintospeex_framein,
 	.frameout = lintospeex_frameout,
@@ -389,7 +437,18 @@ static struct ast_translator lin16tospeexwb = {
 };
 
 static struct ast_translator speexuwbtolin32 = {
-	.name = "speexuwbtolin32", 
+	.name = "speexuwbtolin32",
+	.src_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 32000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 32000,
+	},
+	.format = "slin32",
 	.newpvt = speexuwbtolin32_new,
 	.framein = speextolin_framein,
 	.destroy = speextolin_destroy,
@@ -400,7 +459,18 @@ static struct ast_translator speexuwbtolin32 = {
 };
 
 static struct ast_translator lin32tospeexuwb = {
-	.name = "lin32tospeexuwb", 
+	.name = "lin32tospeexuwb",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 32000,
+	},
+	.dst_codec = {
+		.name = "speex",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 32000,
+	},
+	.format = "speex32",
 	.newpvt = lin32tospeexuwb_new,
 	.framein = lintospeex_framein,
 	.frameout = lintospeex_frameout,
@@ -534,25 +604,6 @@ static int load_module(void)
 	if (parse_config(0))
 		return AST_MODULE_LOAD_DECLINE;
 
-
-	ast_format_set(&speextolin.src_format, AST_FORMAT_SPEEX, 0);
-	ast_format_set(&speextolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&lintospeex.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintospeex.dst_format, AST_FORMAT_SPEEX, 0);
-
-	ast_format_set(&speexwbtolin16.src_format, AST_FORMAT_SPEEX16, 0);
-	ast_format_set(&speexwbtolin16.dst_format, AST_FORMAT_SLINEAR16, 0);
-
-	ast_format_set(&lin16tospeexwb.src_format, AST_FORMAT_SLINEAR16, 0);
-	ast_format_set(&lin16tospeexwb.dst_format, AST_FORMAT_SPEEX16, 0);
-
-	ast_format_set(&speexuwbtolin32.src_format, AST_FORMAT_SPEEX32, 0);
-	ast_format_set(&speexuwbtolin32.dst_format, AST_FORMAT_SLINEAR32, 0);
-
-	ast_format_set(&lin32tospeexuwb.src_format, AST_FORMAT_SLINEAR32, 0);
-	ast_format_set(&lin32tospeexuwb.dst_format, AST_FORMAT_SPEEX32, 0);
-
 	res |= ast_register_translator(&speextolin);
 	res |= ast_register_translator(&lintospeex);
 	res |= ast_register_translator(&speexwbtolin16);
@@ -560,6 +611,10 @@ static int load_module(void)
 	res |= ast_register_translator(&speexuwbtolin32);
 	res |= ast_register_translator(&lin32tospeexuwb);
 
+	if (res) {
+		unload_module();
+		return res;
+	}
 
 	return res;
 }
diff --git a/codecs/codec_ulaw.c b/codecs/codec_ulaw.c
index 86a0706c4..9d0aa3c14 100644
--- a/codecs/codec_ulaw.c
+++ b/codecs/codec_ulaw.c
@@ -82,6 +82,17 @@ static int lintoulaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
 
 static struct ast_translator ulawtolin = {
 	.name = "ulawtolin",
+	.src_codec = {
+		.name = "ulaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.framein = ulawtolin_framein,
 	.sample = ulaw_sample,
 	.buffer_samples = BUFFER_SAMPLES,
@@ -90,6 +101,17 @@ static struct ast_translator ulawtolin = {
 
 static struct ast_translator testlawtolin = {
 	.name = "testlawtolin",
+	.src_codec = {
+		.name = "testlaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "slin",
 	.framein = ulawtolin_framein,
 	.sample = ulaw_sample,
 	.buffer_samples = BUFFER_SAMPLES,
@@ -102,6 +124,17 @@ static struct ast_translator testlawtolin = {
 
 static struct ast_translator lintoulaw = {
 	.name = "lintoulaw",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "ulaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "ulaw",
 	.framein = lintoulaw_framein,
 	.sample = slin8_sample,
 	.buf_size = BUFFER_SAMPLES,
@@ -110,17 +143,23 @@ static struct ast_translator lintoulaw = {
 
 static struct ast_translator lintotestlaw = {
 	.name = "lintotestlaw",
+	.src_codec = {
+		.name = "slin",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.dst_codec = {
+		.name = "testlaw",
+		.type = AST_MEDIA_TYPE_AUDIO,
+		.sample_rate = 8000,
+	},
+	.format = "testlaw",
 	.framein = lintoulaw_framein,
 	.sample = slin8_sample,
 	.buf_size = BUFFER_SAMPLES,
 	.buffer_samples = BUFFER_SAMPLES,
 };
 
-static int reload(void)
-{
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
 static int unload_module(void)
 {
 	int res;
@@ -137,32 +176,20 @@ static int load_module(void)
 {
 	int res;
 
-	ast_format_set(&lintoulaw.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintoulaw.dst_format, AST_FORMAT_ULAW, 0);
-
-	ast_format_set(&lintotestlaw.src_format, AST_FORMAT_SLINEAR, 0);
-	ast_format_set(&lintotestlaw.dst_format, AST_FORMAT_TESTLAW, 0);
-
-	ast_format_set(&ulawtolin.src_format, AST_FORMAT_ULAW, 0);
-	ast_format_set(&ulawtolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
-	ast_format_set(&testlawtolin.src_format, AST_FORMAT_TESTLAW, 0);
-	ast_format_set(&testlawtolin.dst_format, AST_FORMAT_SLINEAR, 0);
-
 	res = ast_register_translator(&ulawtolin);
-	if (!res) {
-		res = ast_register_translator(&lintoulaw);
-		res |= ast_register_translator(&lintotestlaw);
-		res |= ast_register_translator(&testlawtolin);
-	} else
-		ast_unregister_translator(&ulawtolin);
-	if (res)
+	res |= ast_register_translator(&lintoulaw);
+	res |= ast_register_translator(&lintotestlaw);
+	res |= ast_register_translator(&testlawtolin);
+
+	if (res) {
+		unload_module();
 		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	return AST_MODULE_LOAD_SUCCESS;
 }
 
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "mu-Law Coder/Decoder",
 		.load = load_module,
 		.unload = unload_module,
-		.reload = reload,
 	       );
diff --git a/codecs/ex_adpcm.h b/codecs/ex_adpcm.h
index 96b7c51f8..360239789 100644
--- a/codecs/ex_adpcm.h
+++ b/codecs/ex_adpcm.h
@@ -26,7 +26,8 @@ static struct ast_frame *adpcm_sample(void)
 		.src = __PRETTY_FUNCTION__,
 		.data.ptr = ex_adpcm,
 	};
-	ast_format_set(&f.subclass.format, AST_FORMAT_ADPCM, 0);
+
+	f.subclass.format = ast_format_adpcm;
 
 	return &f;
 }
diff --git a/codecs/ex_alaw.h b/codecs/ex_alaw.h
index bbf8ad982..e8629be5e 100644
--- a/codecs/ex_alaw.h
+++ b/codecs/ex_alaw.h
@@ -31,6 +31,6 @@ static struct ast_frame *alaw_sample(void)
 		.src = __PRETTY_FUNCTION__,
 		.data.ptr = ex_alaw,
 	};
-	ast_format_set(&f.subclass.format, AST_FORMAT_ALAW, 0);
+	f.subclass.format = ast_format_alaw;
 	return &f;
 }
diff --git a/codecs/ex_g722.h b/codecs/ex_g722.h
index 0e9f22686..390cc7b5c 100644
--- a/codecs/ex_g722.h
+++ b/codecs/ex_g722.h
@@ -42,7 +42,7 @@ static struct ast_frame *g722_sample(void)
 		.data.ptr = ex_g722,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_G722, 0);
+	f.subclass.format = ast_format_slin;
 
 	return &f;
 }
diff --git a/codecs/ex_g726.h b/codecs/ex_g726.h
index d5438c91a..f125253ee 100644
--- a/codecs/ex_g726.h
+++ b/codecs/ex_g726.h
@@ -27,7 +27,7 @@ static struct ast_frame *g726_sample(void)
 		.data.ptr = ex_g726,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_G726, 0);
+	f.subclass.format = ast_format_g726;
 
 	return &f;
 }
diff --git a/codecs/ex_gsm.h b/codecs/ex_gsm.h
index 8f8b4f9fe..006706038 100644
--- a/codecs/ex_gsm.h
+++ b/codecs/ex_gsm.h
@@ -27,6 +27,7 @@ static struct ast_frame *gsm_sample(void)
 		.data.ptr = ex_gsm,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_GSM, 0);
+	f.subclass.format = ast_format_gsm;
+
 	return &f;
 }
diff --git a/codecs/ex_ilbc.h b/codecs/ex_ilbc.h
index 93cf5eacf..3a79b0918 100644
--- a/codecs/ex_ilbc.h
+++ b/codecs/ex_ilbc.h
@@ -28,6 +28,7 @@ static struct ast_frame *ilbc_sample(void)
 		.data.ptr = ex_ilbc,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_ILBC, 0);
+	f.subclass.format = ast_format_ilbc;
+
 	return &f;
 }
diff --git a/codecs/ex_lpc10.h b/codecs/ex_lpc10.h
index a36e06add..2e271c005 100644
--- a/codecs/ex_lpc10.h
+++ b/codecs/ex_lpc10.h
@@ -25,7 +25,7 @@ static struct ast_frame *lpc10_sample(void)
 		.data.ptr = ex_lpc10,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_LPC10, 0);
+	f.subclass.format = ast_format_lpc10;
 
 	return &f;
 }
diff --git a/codecs/ex_speex.h b/codecs/ex_speex.h
index e9411e5d7..76e5925b8 100644
--- a/codecs/ex_speex.h
+++ b/codecs/ex_speex.h
@@ -27,7 +27,7 @@ static struct ast_frame *speex_sample(void)
 		.data.ptr = ex_speex,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_SPEEX, 0);
+	f.subclass.format = ast_format_speex;
 
 	return &f;
 }
@@ -58,7 +58,8 @@ static struct ast_frame *speex16_sample(void)
 		.src = __PRETTY_FUNCTION__,
 		.data.ptr = ex_speex16,
 	};
-	ast_format_set(&f.subclass.format, AST_FORMAT_SPEEX16, 0);
+
+	f.subclass.format = ast_format_speex16;
 
 	return &f;
 }
diff --git a/codecs/ex_ulaw.h b/codecs/ex_ulaw.h
index 2ab9222fb..d18a08e9c 100644
--- a/codecs/ex_ulaw.h
+++ b/codecs/ex_ulaw.h
@@ -32,6 +32,7 @@ static struct ast_frame *ulaw_sample(void)
 		.data.ptr = ex_ulaw,
 	};
 
-	ast_format_set(&f.subclass.format, AST_FORMAT_ULAW, 0);
+	f.subclass.format = ast_format_ulaw;
+
 	return &f;
 }