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-rw-r--r--codecs/codec_a_mu.c40
-rw-r--r--codecs/codec_adpcm.c48
-rw-r--r--codecs/codec_alaw.c49
-rw-r--r--codecs/codec_dahdi.c399
-rw-r--r--codecs/codec_g722.c62
-rw-r--r--codecs/codec_g726.c62
-rw-r--r--codecs/codec_gsm.c53
-rw-r--r--codecs/codec_ilbc.c44
-rw-r--r--codecs/codec_lpc10.c50
-rw-r--r--codecs/codec_resample.c76
-rw-r--r--codecs/codec_speex.c103
-rw-r--r--codecs/codec_ulaw.c77
-rw-r--r--codecs/ex_adpcm.h3
-rw-r--r--codecs/ex_alaw.h2
-rw-r--r--codecs/ex_g722.h2
-rw-r--r--codecs/ex_g726.h2
-rw-r--r--codecs/ex_gsm.h3
-rw-r--r--codecs/ex_ilbc.h3
-rw-r--r--codecs/ex_lpc10.h2
-rw-r--r--codecs/ex_speex.h5
-rw-r--r--codecs/ex_ulaw.h3
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;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-01 18:51:33 +0000