@@ -56,6 +56,21 @@ fi
AM_CONDITIONAL(BUILD_SMPP, test "x$osmo_ac_build_smpp" = "xyes")
AC_SUBST(osmo_ac_build_smpp)
+# Enable/disable transcoding within osmo-bsc_mgcp?
+AC_ARG_ENABLE([mgcp-transcoding], [AS_HELP_STRING([--enable-mgcp-transcoding], [Build the MGCP gateway with internal transcoding enabled.])],
+ [osmo_ac_mgcp_transcoding="$enableval"],[osmo_ac_mgcp_transcoding="no"])
+AC_ARG_WITH([g729], [AS_HELP_STRING([--with-g729], [Enable G.729 encoding/decoding.])], [osmo_ac_with_g729="$withval"],[osmo_ac_with_g729="no"])
+
+if test "$osmo_ac_mgcp_transcoding" = "yes" ; then
+ AC_SEARCH_LIBS(gsm_create, gsm)
+ if test "$osmo_ac_with_g729" = "yes" ; then
+ PKG_CHECK_MODULES(LIBBCG729, libbcg729 >= 0.1, [AC_DEFINE([HAVE_BCG729], [1], [Use bgc729 decoder/encoder])])
+ fi
+ AC_DEFINE(BUILD_MGCP_TRANSCODING, 1, [Define if we want to build the MGCP gateway with transcoding support])
+fi
+AM_CONDITIONAL(BUILD_MGCP_TRANSCODING, test "x$osmo_ac_mgcp_transcoding" = "xyes")
+AC_SUBST(osmo_ac_mgcp_transcoding)
+
found_libgtp=yes
PKG_CHECK_MODULES(LIBGTP, libgtp, , found_libgtp=no)
@@ -1,10 +1,16 @@
AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir)
AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOGSM_CFLAGS) \
- $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS)
+ $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS) \
+ $(LIBBCG729_CFLAGS)
bin_PROGRAMS = osmo-bsc_mgcp
osmo_bsc_mgcp_SOURCES = mgcp_main.c
+if BUILD_MGCP_TRANSCODING
+ osmo_bsc_mgcp_SOURCES += mgcp_transcode.c
+endif
osmo_bsc_mgcp_LDADD = $(top_builddir)/src/libcommon/libcommon.a \
$(top_builddir)/src/libmgcp/libmgcp.a -lrt \
- $(LIBOSMOVTY_LIBS) $(LIBOSMOCORE_LIBS)
+ $(LIBOSMOVTY_LIBS) $(LIBOSMOCORE_LIBS) $(LIBBCG729_LIBS)
+
+noinst_HEADERS = g711common.h mgcp_transcode.h
new file mode 100644
@@ -0,0 +1,187 @@
+/*
+ * PCM - A-Law conversion
+ * Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ * Wrapper for linphone Codec class by Simon Morlat <simon.morlat@linphone.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+static inline int val_seg(int val)
+{
+ int r = 0;
+ val >>= 7; /*7 = 4 + 3*/
+ if (val & 0xf0) {
+ val >>= 4;
+ r += 4;
+ }
+ if (val & 0x0c) {
+ val >>= 2;
+ r += 2;
+ }
+ if (val & 0x02)
+ r += 1;
+ return r;
+}
+
+/*
+ * s16_to_alaw() - Convert a 16-bit linear PCM value to 8-bit A-law
+ *
+ * s16_to_alaw() accepts an 16-bit integer and encodes it as A-law data.
+ *
+ * Linear Input Code Compressed Code
+ * ------------------------ ---------------
+ * 0000000wxyza 000wxyz
+ * 0000001wxyza 001wxyz
+ * 000001wxyzab 010wxyz
+ * 00001wxyzabc 011wxyz
+ * 0001wxyzabcd 100wxyz
+ * 001wxyzabcde 101wxyz
+ * 01wxyzabcdef 110wxyz
+ * 1wxyzabcdefg 111wxyz
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ * G711 is designed for 13 bits input signal, this function add extra shifting to take this into account.
+ */
+
+static inline unsigned char s16_to_alaw(int pcm_val)
+{
+ int mask;
+ int seg;
+ unsigned char aval;
+
+ if (pcm_val >= 0) {
+ mask = 0xD5;
+ } else {
+ mask = 0x55;
+ pcm_val = -pcm_val;
+ if (pcm_val > 0x7fff)
+ pcm_val = 0x7fff;
+ }
+
+ if (pcm_val < 256) /*256 = 32 << 3*/
+ aval = pcm_val >> 4; /*4 = 1 + 3*/
+ else {
+ /* Convert the scaled magnitude to segment number. */
+ seg = val_seg(pcm_val);
+ aval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f);
+ }
+ return aval ^ mask;
+}
+
+/*
+ * alaw_to_s16() - Convert an A-law value to 16-bit linear PCM
+ *
+ */
+static inline int alaw_to_s16(unsigned char a_val)
+{
+ int t;
+ int seg;
+
+ a_val ^= 0x55;
+ t = a_val & 0x7f;
+ if (t < 16)
+ t = (t << 4) + 8;
+ else {
+ seg = (t >> 4) & 0x07;
+ t = ((t & 0x0f) << 4) + 0x108;
+ t <<= seg -1;
+ }
+ return ((a_val & 0x80) ? t : -t);
+}
+/*
+ * s16_to_ulaw() - Convert a linear PCM value to u-law
+ *
+ * In order to simplify the encoding process, the original linear magnitude
+ * is biased by adding 33 which shifts the encoding range from (0 - 8158) to
+ * (33 - 8191). The result can be seen in the following encoding table:
+ *
+ * Biased Linear Input Code Compressed Code
+ * ------------------------ ---------------
+ * 00000001wxyza 000wxyz
+ * 0000001wxyzab 001wxyz
+ * 000001wxyzabc 010wxyz
+ * 00001wxyzabcd 011wxyz
+ * 0001wxyzabcde 100wxyz
+ * 001wxyzabcdef 101wxyz
+ * 01wxyzabcdefg 110wxyz
+ * 1wxyzabcdefgh 111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz. * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+static inline unsigned char s16_to_ulaw(int pcm_val) /* 2's complement (16-bit range) */
+{
+ int mask;
+ int seg;
+ unsigned char uval;
+
+ if (pcm_val < 0) {
+ pcm_val = 0x84 - pcm_val;
+ mask = 0x7f;
+ } else {
+ pcm_val += 0x84;
+ mask = 0xff;
+ }
+ if (pcm_val > 0x7fff)
+ pcm_val = 0x7fff;
+
+ /* Convert the scaled magnitude to segment number. */
+ seg = val_seg(pcm_val);
+
+ /*
+ * Combine the sign, segment, quantization bits;
+ * and complement the code word.
+ */
+ uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f);
+ return uval ^ mask;
+}
+
+/*
+ * ulaw_to_s16() - Convert a u-law value to 16-bit linear PCM
+ *
+ * First, a biased linear code is derived from the code word. An unbiased
+ * output can then be obtained by subtracting 33 from the biased code.
+ *
+ * Note that this function expects to be passed the complement of the
+ * original code word. This is in keeping with ISDN conventions.
+ */
+static inline int ulaw_to_s16(unsigned char u_val)
+{
+ int t;
+
+ /* Complement to obtain normal u-law value. */
+ u_val = ~u_val;
+
+ /*
+ * Extract and bias the quantization bits. Then
+ * shift up by the segment number and subtract out the bias.
+ */
+ t = ((u_val & 0x0f) << 3) + 0x84;
+ t <<= (u_val & 0x70) >> 4;
+
+ return ((u_val & 0x80) ? (0x84 - t) : (t - 0x84));
+}
@@ -31,6 +31,11 @@
#include <sys/socket.h>
+#include "g711common.h"
+#include <gsm.h>
+#include <bcg729/decoder.h>
+#include <bcg729/encoder.h>
+
#include <openbsc/debug.h>
#include <openbsc/gsm_data.h>
#include <openbsc/mgcp.h>
@@ -49,6 +54,10 @@
#include "../../bscconfig.h"
+#ifdef BUILD_MGCP_TRANSCODING
+#include "mgcp_transcode.h"
+#endif
+
/* this is here for the vty... it will never be called */
void subscr_put() { abort(); }
@@ -207,6 +216,12 @@ int main(int argc, char **argv)
if (!cfg)
return -1;
+#ifdef BUILD_MGCP_TRANSCODING
+ cfg->setup_rtp_processing_cb = &mgcp_transcoding_setup;
+ cfg->rtp_processing_cb = &mgcp_transcoding_process_rtp;
+ cfg->get_net_downlink_format_cb = &mgcp_transcoding_net_downlink_format;
+#endif
+
vty_info.copyright = openbsc_copyright;
vty_init(&vty_info);
logging_vty_add_cmds(&log_info);
new file mode 100644
@@ -0,0 +1,459 @@
+/*
+ * (C) 2014 by On-Waves
+ * All Rights Reserved
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Affero General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Affero General Public License for more details.
+ *
+ * You should have received a copy of the GNU Affero General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ *
+ */
+
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+
+
+#include "../../bscconfig.h"
+
+#include "g711common.h"
+#include <gsm.h>
+#ifdef HAVE_BCG729
+#include <bcg729/decoder.h>
+#include <bcg729/encoder.h>
+#endif
+
+#include <openbsc/debug.h>
+#include <openbsc/mgcp.h>
+#include <openbsc/mgcp_internal.h>
+
+#include <osmocom/core/talloc.h>
+
+enum audio_format {
+ AF_INVALID,
+ AF_S16,
+ AF_L16,
+ AF_GSM,
+ AF_G729,
+ AF_PCMA
+};
+
+struct mgcp_process_rtp_state {
+ /* decoding */
+ enum audio_format src_fmt;
+ union {
+ gsm gsm_handle;
+#ifdef HAVE_BCG729
+ bcg729DecoderChannelContextStruct *g729_dec;
+#endif
+ } src;
+ size_t src_frame_size;
+ size_t src_samples_per_frame;
+
+ /* processing */
+
+ /* encoding */
+ enum audio_format dst_fmt;
+ union {
+ gsm gsm_handle;
+#ifdef HAVE_BCG729
+ bcg729EncoderChannelContextStruct *g729_enc;
+#endif
+ } dst;
+ size_t dst_frame_size;
+ size_t dst_samples_per_frame;
+};
+
+static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
+{
+ if (rtp_end->subtype_name) {
+ if (!strcmp("GSM", rtp_end->subtype_name))
+ return AF_GSM;
+ if (!strcmp("PCMA", rtp_end->subtype_name))
+ return AF_PCMA;
+#ifdef HAVE_BCG729
+ if (!strcmp("G729", rtp_end->subtype_name))
+ return AF_G729;
+#endif
+ if (!strcmp("L16", rtp_end->subtype_name))
+ return AF_L16;
+ }
+
+ switch (rtp_end->payload_type) {
+ case 3 /* GSM */:
+ return AF_GSM;
+ case 8 /* PCMA */:
+ return AF_PCMA;
+#ifdef HAVE_BCG729
+ case 18 /* G.729 */:
+ return AF_G729;
+#endif
+ case 11 /* L16 */:
+ return AF_L16;
+ default:
+ return AF_INVALID;
+ }
+}
+
+static void l16_encode(short *sample, unsigned char *buf, size_t n)
+{
+ for (; n > 0; --n, ++sample, buf += 2) {
+ buf[0] = sample[0] >> 8;
+ buf[1] = sample[0] & 0xff;
+ }
+}
+
+static void l16_decode(unsigned char *buf, short *sample, size_t n)
+{
+ for (; n > 0; --n, ++sample, buf += 2)
+ sample[0] = ((short)buf[0] << 8) | buf[1];
+}
+
+static void alaw_encode(short *sample, unsigned char *buf, size_t n)
+{
+ for (; n > 0; --n)
+ *(buf++) = s16_to_alaw(*(sample++));
+}
+
+static void alaw_decode(unsigned char *buf, short *sample, size_t n)
+{
+ for (; n > 0; --n)
+ *(sample++) = alaw_to_s16(*(buf++));
+}
+
+static int processing_state_destructor(struct mgcp_process_rtp_state *state)
+{
+ switch (state->src_fmt) {
+ case AF_GSM:
+ if (state->dst.gsm_handle)
+ gsm_destroy(state->src.gsm_handle);
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ if (state->src.g729_dec)
+ closeBcg729DecoderChannel(state->src.g729_dec);
+ break;
+#endif
+ default:
+ break;
+ }
+ switch (state->dst_fmt) {
+ case AF_GSM:
+ if (state->dst.gsm_handle)
+ gsm_destroy(state->dst.gsm_handle);
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ if (state->dst.g729_enc)
+ closeBcg729EncoderChannel(state->dst.g729_enc);
+ break;
+#endif
+ default:
+ break;
+ }
+ return 0;
+}
+
+int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
+ struct mgcp_rtp_end *dst_end,
+ struct mgcp_rtp_end *src_end)
+{
+ struct mgcp_process_rtp_state *state = dst_end->rtp_process_data;
+ enum audio_format src_fmt, dst_fmt;
+
+ /* cleanup first */
+ if (state) {
+ talloc_free(state);
+ dst_end->rtp_process_data = NULL;
+ }
+
+ if (!src_end)
+ return 0;
+
+ src_fmt = get_audio_format(src_end);
+ dst_fmt = get_audio_format(dst_end);
+
+ LOGP(DMGCP, LOGL_ERROR,
+ "Checking transcoding: %s (%d) -> %s (%d)\n",
+ src_end->subtype_name, src_end->payload_type,
+ dst_end->subtype_name, dst_end->payload_type);
+
+ if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
+ if (!src_end->subtype_name || !dst_end->subtype_name)
+ /* Not enough info, do nothing */
+ return 0;
+
+ if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
+ /* Nothing to do */
+ return 0;
+
+ LOGP(DMGCP, LOGL_ERROR,
+ "Cannot transcode: %s codec not supported (%s -> %s).\n",
+ src_fmt != AF_INVALID ? "destination" : "source",
+ src_end->audio_name, dst_end->audio_name);
+ return -EINVAL;
+ }
+
+ if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Cannot transcode: rate conversion (%d -> %d) not supported.\n",
+ src_end->rate, dst_end->rate);
+ return -EINVAL;
+ }
+
+ state = talloc_zero(endp->tcfg->cfg, struct mgcp_process_rtp_state);
+ talloc_set_destructor(state, processing_state_destructor);
+ dst_end->rtp_process_data = state;
+
+ state->src_fmt = src_fmt;
+
+ switch (state->src_fmt) {
+ case AF_L16:
+ case AF_S16:
+ state->src_frame_size = 80 * sizeof(short);
+ state->src_samples_per_frame = 80;
+ break;
+ case AF_GSM:
+ state->src_frame_size = sizeof(gsm_frame);
+ state->src_samples_per_frame = 160;
+ state->src.gsm_handle = gsm_create();
+ if (!state->src.gsm_handle) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Failed to initialize GSM decoder.\n");
+ return -EINVAL;
+ }
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ state->src_frame_size = 10;
+ state->src_samples_per_frame = 80;
+ state->src.g729_dec = initBcg729DecoderChannel();
+ if (!state->src.g729_dec) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Failed to initialize G.729 decoder.\n");
+ return -EINVAL;
+ }
+ break;
+#endif
+ case AF_PCMA:
+ state->src_frame_size = 80;
+ state->src_samples_per_frame = 80;
+ break;
+ default:
+ break;
+ }
+
+ state->dst_fmt = dst_fmt;
+
+ switch (state->dst_fmt) {
+ case AF_L16:
+ case AF_S16:
+ state->dst_frame_size = 80*sizeof(short);
+ state->dst_samples_per_frame = 80;
+ break;
+ case AF_GSM:
+ state->dst_frame_size = sizeof(gsm_frame);
+ state->dst_samples_per_frame = 160;
+ state->dst.gsm_handle = gsm_create();
+ if (!state->dst.gsm_handle) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Failed to initialize GSM encoder.\n");
+ return -EINVAL;
+ }
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ state->dst_frame_size = 10;
+ state->dst_samples_per_frame = 80;
+ state->dst.g729_enc = initBcg729EncoderChannel();
+ if (!state->dst.g729_enc) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Failed to initialize G.729 decoder.\n");
+ return -EINVAL;
+ }
+ break;
+#endif
+ case AF_PCMA:
+ state->dst_frame_size = 80;
+ state->dst_samples_per_frame = 80;
+ break;
+ default:
+ break;
+ }
+
+ LOGP(DMGCP, LOGL_INFO,
+ "Initialized RTP processing on: 0x%x "
+ "conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n",
+ ENDPOINT_NUMBER(endp),
+ src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra,
+ dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra);
+
+ return 0;
+}
+
+void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
+ int *payload_type,
+ const char**audio_name,
+ const char**fmtp_extra)
+{
+ struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data;
+ if (!state || endp->net_end.payload_type < 0) {
+ *payload_type = endp->bts_end.payload_type;
+ *audio_name = endp->bts_end.audio_name;
+ *fmtp_extra = endp->bts_end.fmtp_extra;
+ return;
+ }
+
+ *payload_type = endp->net_end.payload_type;
+ *fmtp_extra = endp->net_end.fmtp_extra;
+ *audio_name = endp->net_end.audio_name;
+}
+
+
+int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
+ struct mgcp_rtp_end *dst_end,
+ char *data, int *len, int buf_size)
+{
+ struct mgcp_process_rtp_state *state = dst_end->rtp_process_data;
+ size_t rtp_hdr_size = 12;
+ char *payload_data = data + rtp_hdr_size;
+ int payload_len = *len - rtp_hdr_size;
+ size_t sample_cnt = 0;
+ size_t sample_idx;
+ int16_t samples[10*160];
+ uint8_t *src = (uint8_t *)payload_data;
+ uint8_t *dst = (uint8_t *)payload_data;
+ size_t nbytes = payload_len;
+ size_t frame_remainder;
+
+ if (!state)
+ return 0;
+
+ if (state->src_fmt == state->dst_fmt)
+ return 0;
+
+ /* TODO: check payload type (-> G.711 comfort noise) */
+
+ /* Decode src into samples */
+ while (nbytes >= state->src_frame_size) {
+ if (sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(samples)) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Sample buffer too small: %d > %d.\n",
+ sample_cnt + state->src_samples_per_frame,
+ ARRAY_SIZE(samples));
+ return -ENOSPC;
+ }
+ switch (state->src_fmt) {
+ case AF_GSM:
+ if (gsm_decode(state->src.gsm_handle,
+ (gsm_byte *)src, samples + sample_cnt) < 0) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Failed to decode GSM.\n");
+ return -EINVAL;
+ }
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ bcg729Decoder(state->src.g729_dec, src, 0, samples + sample_cnt);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_decode(src, samples + sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(samples + sample_cnt, src,
+ state->src_frame_size);
+ break;
+ case AF_L16:
+ l16_decode(src, samples + sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ src += state->src_frame_size;
+ nbytes -= state->src_frame_size;
+ sample_cnt += state->src_samples_per_frame;
+ }
+
+ /* Add silence if necessary */
+ frame_remainder = sample_cnt % state->dst_samples_per_frame;
+ if (frame_remainder) {
+ size_t silence = state->dst_samples_per_frame - frame_remainder;
+ if (sample_cnt + silence > ARRAY_SIZE(samples)) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Sample buffer too small for silence: %d > %d.\n",
+ sample_cnt + silence,
+ ARRAY_SIZE(samples));
+ return -ENOSPC;
+ }
+
+ while (silence > 0) {
+ samples[sample_cnt] = 0;
+ sample_cnt += 1;
+ silence -= 1;
+ }
+ }
+
+ /* Encode samples into dst */
+ sample_idx = 0;
+ nbytes = 0;
+ while (sample_idx + state->dst_samples_per_frame <= sample_cnt) {
+ if (nbytes + state->dst_frame_size > buf_size) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Encoding (RTP) buffer too small: %d > %d.\n",
+ nbytes + state->dst_frame_size, buf_size);
+ return -ENOSPC;
+ }
+ switch (state->dst_fmt) {
+ case AF_GSM:
+ gsm_encode(state->dst.gsm_handle,
+ samples + sample_idx, dst);
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ bcg729Encoder(state->dst.g729_enc,
+ samples + sample_idx, dst);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_encode(samples + sample_idx, dst,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(dst, samples + sample_idx, state->dst_frame_size);
+ break;
+ case AF_L16:
+ l16_encode(samples + sample_idx, dst,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ dst += state->dst_frame_size;
+ nbytes += state->dst_frame_size;
+ sample_idx += state->dst_samples_per_frame;
+ }
+
+ *len = rtp_hdr_size + nbytes;
+ /* Patch payload type */
+ data[1] = (data[1] & 0x80) | (dst_end->payload_type & 0x7f);
+
+ /* TODO: remove me
+ fprintf(stderr, "sample_cnt = %d, sample_idx = %d, plen = %d -> %d, "
+ "hdr_size = %d, len = %d, pt = %d\n",
+ sample_cnt, sample_idx, payload_len, nbytes, rtp_hdr_size, *len,
+ data[1]);
+ */
+
+ return 0;
+}
new file mode 100644
@@ -0,0 +1,34 @@
+/*
+ * (C) 2014 by On-Waves
+ * All Rights Reserved
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Affero General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Affero General Public License for more details.
+ *
+ * You should have received a copy of the GNU Affero General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ *
+ */
+#ifndef OPENBSC_MGCP_TRANSCODE_H
+#define OPENBSC_MGCP_TRANSCODE_H
+
+int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
+ struct mgcp_rtp_end *dst_end,
+ struct mgcp_rtp_end *src_end);
+
+void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
+ int *payload_type,
+ const char**audio_name,
+ const char**fmtp_extra);
+
+int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
+ struct mgcp_rtp_end *dst_end,
+ char *data, int *len, int buf_size);
+#endif /* OPENBSC_MGCP_TRANSCODE_H */