Merged revisions 46154 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines

add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 5cc7cddb0294776dcf261a43394ad677a6e4c998..ffb86b0d752897916391fee1e4315214182bbe73 100644 (file)
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -3122,6 +3122,8 @@ static char *convertcap(int cap)
                return "ULAW";
        case AST_FORMAT_ALAW:
                return "ALAW";
+       case AST_FORMAT_G722:
+               return "G.722";
        case AST_FORMAT_ADPCM:
                return "G.728";
        case AST_FORMAT_G729A:

diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index dfefcea9c993194c6ff364042b868daffdf6d190..108846d6b7cd2d3745bf23bee837e31d7af5b74b 100644 (file)
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -183,7 +183,8 @@ int (*iax2_regfunk)(const char *username, int onoff) = NULL;
 #define IAX_CAPABILITY_MEDBANDWIDTH    (IAX_CAPABILITY_FULLBANDWIDTH &         \
                                         ~AST_FORMAT_SLINEAR &                  \
                                         ~AST_FORMAT_ULAW &                     \
-                                        ~AST_FORMAT_ALAW) 
+                                        ~AST_FORMAT_ALAW &                     \
+                                        ~AST_FORMAT_G722) 
 /* A modem */
 #define IAX_CAPABILITY_LOWBANDWIDTH    (IAX_CAPABILITY_MEDBANDWIDTH &          \
                                         ~AST_FORMAT_G726 &                     \

diff --git a/formats/format_pcm.c b/formats/format_pcm.c
index 9ae1398e6e44dc6d703131819f31825753120937..a3a40b7d49e0c2fc3b33528a5b3abd4514f8deab 100644 (file)
--- a/formats/format_pcm.c
+++ b/formats/format_pcm.c
@@ -443,6 +443,18 @@ static const struct ast_format pcm_f = {
        .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
 };
 
+static const struct ast_format g722_f = {
+       .name = "g722",
+       .exts = "g722",
+       .format = AST_FORMAT_G722,
+       .write = pcm_write,
+       .seek = pcm_seek,
+       .trunc = pcm_trunc,
+       .tell = pcm_tell,
+       .read = pcm_read,
+       .buf_size = (BUF_SIZE * 2) + AST_FRIENDLY_OFFSET,
+};
+
 static const struct ast_format au_f = {
        .name = "au",
        .exts = "au",
@@ -467,14 +479,18 @@ static int load_module(void)
        for (index = 0; index < (sizeof(alaw_silence) / sizeof(alaw_silence[0])); index++)
                alaw_silence[index] = AST_LIN2A(0);
 
-       return ast_format_register(&pcm_f) || ast_format_register(&alaw_f)
-               || ast_format_register(&au_f);
+       return ast_format_register(&pcm_f)
+               || ast_format_register(&alaw_f)
+               || ast_format_register(&au_f)
+               || ast_format_register(&g722_f);
 }
 
 static int unload_module(void)
 {
-       return ast_format_unregister(pcm_f.name) || ast_format_unregister(alaw_f.name)
-               || ast_format_unregister(au_f.name);
+       return ast_format_unregister(pcm_f.name)
+               || ast_format_unregister(alaw_f.name)
+               || ast_format_unregister(au_f.name)
+               || ast_format_unregister(g722_f.name);
 }      
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw/Sun uLaw/ALaw 8khz Audio support (PCM,PCMA,AU)");
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw/Sun uLaw/ALaw 8KHz Audio support (PCM,PCMA,AU) and G.722 16Khz Audio Support");

diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index bcc533a95aa0669f68640e58275177688d53b371..fa1c1289128b926d767419ae1233498576664403 100644 (file)
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -240,6 +240,8 @@ extern struct ast_frame ast_null_frame;
 #define AST_FORMAT_ILBC                (1 << 10)
 /*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
 #define AST_FORMAT_G726                (1 << 11)
+/*! G.722 */
+#define AST_FORMAT_G722                (1 << 12)
 /*! Maximum audio format */
 #define AST_FORMAT_MAX_AUDIO   (1 << 15)
 /*! Maximum audio mask */

diff --git a/main/frame.c b/main/frame.c
index f797f55fad48b55b2a22f112e8ae8cf9f2049540..1b18a611da5d0ee8fc817a0f88f8a545042fb8d6 100644 (file)
--- a/main/frame.c
+++ b/main/frame.c
@@ -106,14 +106,15 @@ static struct ast_format_list AST_FORMAT_LIST[] = {                                       /*!< Bit number: comment
        { 1, AST_FORMAT_GSM, "gsm" , "GSM", 33, 20, 300, 20, 20 },              /*!<  2: codec_gsm.c */
        { 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law", 80, 10, 150, 10, 20 },     /*!<  3: codec_ulaw.c */
        { 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law", 80, 10, 150, 10, 20 },     /*!<  4: codec_alaw.c */
-       { 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 300, 10, 20 },/*!<  5: codec_g726.c */
+       { 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 300, 10, 20 },   /*!<  5: codec_g726.c */
        { 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM", 40, 10, 300, 10, 20 },        /*!<  6: codec_adpcm.c */
        { 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE },       /*!< 7 */
-       { 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },   /*!<  8: codec_lpc10.c */ 
+       { 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },           /*!<  8: codec_lpc10.c */ 
        { 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 }, /*!<  9: Binary commercial distribution */
-       { 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },  /*!< 10: codec_speex.c */
+       { 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },          /*!< 10: codec_speex.c */
        { 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },             /*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
        { 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10, 20 },     /*!<  12: codec_g726.c */
+       { 1, AST_FORMAT_G722, "g722", "G722"},                                  /*!< 13 */
        { 0, 0, "nothing", "undefined" },
        { 0, 0, "nothing", "undefined" },
        { 0, 0, "nothing", "undefined" },
@@ -1356,6 +1357,7 @@ int ast_codec_get_samples(struct ast_frame *f)
                break;
        case AST_FORMAT_ULAW:
        case AST_FORMAT_ALAW:
+       case AST_FORMAT_G722:
                samples = f->datalen;
                break;
        case AST_FORMAT_ADPCM:

diff --git a/main/rtp.c b/main/rtp.c
index 589584deeb846834d443f50a44ff1d926eaac979..c34f56b117493e5e16e0c46a44dffc26b401053e 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1330,6 +1330,7 @@ static struct {
        {{1, AST_FORMAT_G729A}, "audio", "G729"},
        {{1, AST_FORMAT_SPEEX}, "audio", "speex"},
        {{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
+       {{1, AST_FORMAT_G722}, "audio", "G722"},
        {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"},
        {{0, AST_RTP_DTMF}, "audio", "telephone-event"},
        {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
@@ -1356,6 +1357,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
        [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
        [7] = {1, AST_FORMAT_LPC10},
        [8] = {1, AST_FORMAT_ALAW},
+       [9] = {1, AST_FORMAT_G722},
        [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
        [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
        [13] = {0, AST_RTP_CN},

diff --git a/main/translate.c b/main/translate.c
index 7845a24f16cc45561bbf2828391f986ce47219e1..d130a38dea659c319ffcd5626595964765b37507 100644 (file)
--- a/main/translate.c
+++ b/main/translate.c
@@ -476,7 +476,7 @@ static void rebuild_matrix(int samples)
 
 static int show_translation(int fd, int argc, char *argv[])
 {
-#define SHOW_TRANS 12
+#define SHOW_TRANS 13
        int x, y, z;
        int curlen = 0, longest = 0;