@@ -1,5 +1,5 @@
-OBJS = testconv_main.o mgcp_transcode.o
+OBJS = testconv_main.o
CC = gcc
CFLAGS = -O0 -ggdb -Wall
@@ -11,7 +11,6 @@ testconv: $(OBJS)
$(CC) -o $@ $^ $(LDFLAGS) $(LIBS)
testconv_main.o: testconv_main.c
-mgcp_transcode.o: ../../src/osmo-bsc_mgcp/mgcp_transcode.c
$(OBJS):
$(CC) $(CFLAGS) $(CPPFLAGS) -c -o $@ $<
@@ -17,7 +17,7 @@
#error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)"
#endif
-#include "src/osmo-bsc_mgcp/mgcp_transcode.h"
+#include "openbsc/mgcp_transcode.h"
static int audio_name_to_type(const char *name)
{
@@ -13,7 +13,8 @@ noinst_HEADERS = abis_nm.h abis_rsl.h db.h gsm_04_08.h gsm_data.h \
osmo_bsc_rf.h osmo_bsc.h network_listen.h bsc_nat_sccp.h \
osmo_msc_data.h osmo_bsc_grace.h sms_queue.h abis_om2000.h \
bss.h gsm_data_shared.h control_cmd.h ipaccess.h mncc_int.h \
- arfcn_range_encode.h nat_rewrite_trie.h bsc_nat_callstats.h
+ arfcn_range_encode.h nat_rewrite_trie.h bsc_nat_callstats.h \
+ mgcp_transcode.h
openbsc_HEADERS = gsm_04_08.h meas_rep.h bsc_api.h
openbscdir = $(includedir)/openbsc
new file mode 100644
@@ -0,0 +1,36 @@
+/*
+ * (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);
+
+int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst);
+#endif /* OPENBSC_MGCP_TRANSCODE_H */
@@ -1,7 +1,15 @@
AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir)
-AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) $(COVERAGE_CFLAGS)
-AM_LDFLAGS = $(LIBOSMOCORE_LIBS) $(LIBOSMOGSM_LIBS) $(COVERAGE_LDFLAGS)
+AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) $(COVERAGE_CFLAGS) \
+ $(LIBBCG729_CFLAGS)
+AM_LDFLAGS = $(LIBOSMOCORE_LIBS) $(LIBOSMOGSM_LIBS) $(COVERAGE_LDFLAGS) \
+ $(LIBBCG729_LIBS)
noinst_LIBRARIES = libmgcp.a
+noinst_HEADERS = g711common.h
+
libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c
+
+if BUILD_MGCP_TRANSCODING
+ libmgcp_a_SOURCES += mgcp_transcode.c
+endif
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));
+}
new file mode 100644
@@ -0,0 +1,553 @@
+/*
+ * (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;
+ int dst_packet_duration;
+
+ int is_running;
+ uint16_t next_seq;
+ uint32_t next_time;
+ int16_t samples[10*160];
+ size_t sample_cnt;
+ size_t sample_offs;
+};
+
+int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
+{
+ struct mgcp_process_rtp_state *state = state_;
+ if (dst)
+ return (nsamples >= 0 ?
+ nsamples / state->dst_samples_per_frame :
+ 1) * state->dst_frame_size;
+ else
+ return (nsamples >= 0 ?
+ nsamples / state->src_samples_per_frame :
+ 1) * state->src_frame_size;
+}
+
+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;
+ }
+
+ if (dst_end->force_output_ptime)
+ state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end);
+
+ 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;
+}
+
+static int decode_audio(struct mgcp_process_rtp_state *state,
+ uint8_t **src, size_t *nbytes)
+{
+ while (*nbytes >= state->src_frame_size) {
+ if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Sample buffer too small: %d > %d.\n",
+ state->sample_cnt + state->src_samples_per_frame,
+ ARRAY_SIZE(state->samples));
+ return -ENOSPC;
+ }
+ switch (state->src_fmt) {
+ case AF_GSM:
+ if (gsm_decode(state->src.gsm_handle,
+ (gsm_byte *)*src, state->samples + state->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, state->samples + state->sample_cnt);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_decode(*src, state->samples + state->sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(state->samples + state->sample_cnt, *src,
+ state->src_frame_size);
+ break;
+ case AF_L16:
+ l16_decode(*src, state->samples + state->sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ *src += state->src_frame_size;
+ *nbytes -= state->src_frame_size;
+ state->sample_cnt += state->src_samples_per_frame;
+ }
+ return 0;
+}
+
+static int encode_audio(struct mgcp_process_rtp_state *state,
+ uint8_t *dst, size_t buf_size, size_t max_samples)
+{
+ int nbytes = 0;
+ size_t nsamples = 0;
+ /* Encode samples into dst */
+ while (nsamples + state->dst_samples_per_frame <= max_samples) {
+ if (nbytes + state->dst_frame_size > buf_size) {
+ if (nbytes > 0)
+ break;
+
+ /* Not even one frame fits into the buffer */
+ LOGP(DMGCP, LOGL_INFO,
+ "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,
+ state->samples + state->sample_offs, dst);
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ bcg729Encoder(state->dst.g729_enc,
+ state->samples + state->sample_offs, dst);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_encode(state->samples + state->sample_offs, dst,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(dst, state->samples + state->sample_offs,
+ state->dst_frame_size);
+ break;
+ case AF_L16:
+ l16_encode(state->samples + state->sample_offs, dst,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ dst += state->dst_frame_size;
+ nbytes += state->dst_frame_size;
+ state->sample_offs += state->dst_samples_per_frame;
+ nsamples += state->dst_samples_per_frame;
+ }
+ state->sample_cnt -= nsamples;
+ return nbytes;
+}
+
+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_idx;
+ uint8_t *src = (uint8_t *)payload_data;
+ uint8_t *dst = (uint8_t *)payload_data;
+ size_t nbytes = payload_len;
+ // size_t frame_remainder;
+ size_t nsamples;
+ size_t max_samples;
+ uint32_t ts_no;
+ int rc;
+
+ if (!state)
+ return 0;
+
+ if (state->src_fmt == state->dst_fmt) {
+ if (!state->dst_packet_duration)
+ return 0;
+
+ /* TODO: repackage without transcoding */
+ }
+
+ /* If the remaining samples do not fit into a fixed ptime,
+ * a) discard them, if the next packet is much later
+ * b) add silence and * send it, if the current packet is not
+ * yet too late
+ * c) append the sample data, if the timestamp matches exactly
+ */
+
+ /* TODO: check payload type (-> G.711 comfort noise) */
+
+ if (payload_len > 0) {
+ ts_no = ntohl(*(uint32_t*)(data+4));
+ if (!state->is_running)
+ state->next_seq = ntohs(*(uint32_t*)(data+4));
+
+ state->is_running = 1;
+
+ if (state->sample_cnt > 0) {
+ int32_t delta = ts_no - state->next_time;
+ /* TODO: check sequence? reordering? packet loss? */
+
+ if (delta > state->sample_cnt)
+ /* There is a time gap between the last packet
+ * and the current one. Just discard the
+ * partial data that is left in the buffer.
+ * TODO: This can be improved by adding silence
+ * instead if the delta is small enough.
+ */
+ state->sample_cnt = 0;
+ else if (delta < 0) {
+ LOGP(DMGCP, LOGL_NOTICE,
+ "RTP time jumps backwards, delta = %d, "
+ "discarding buffered samples\n",
+ delta);
+ state->sample_cnt = 0;
+ state->sample_offs = 0;
+ return -EAGAIN;
+ }
+
+ /* Make sure the samples start without offset */
+ fprintf(stderr, "Moving %d samples to buffer start (offset %d)\n", state->sample_cnt, state->sample_offs);
+ if (state->sample_offs && state->sample_cnt)
+ memmove(&state->samples[0],
+ &state->samples[state->sample_offs],
+ state->sample_cnt * sizeof(state->samples[0]));
+ }
+
+ state->sample_offs = 0;
+
+ /* Append decoded audio to samples */
+ decode_audio(state, &src, &nbytes);
+
+ if (nbytes > 0)
+ LOGP(DMGCP, LOGL_NOTICE,
+ "Skipped audio frame in RTP packet: %d octets\n",
+ nbytes);
+ } else
+ ts_no = state->next_time;
+
+ if (state->sample_cnt < state->dst_packet_duration)
+ return -EAGAIN;
+
+ max_samples =
+ state->dst_packet_duration ?
+ state->dst_packet_duration : state->sample_cnt;
+
+ nsamples = state->sample_cnt;
+
+ rc = encode_audio(state, dst, buf_size, max_samples);
+ if (rc <= 0)
+ return rc;
+
+ nsamples -= state->sample_cnt;
+ fprintf(stderr, "Wrote %d samples to buffer (offset %d)\n", nsamples, state->sample_offs);
+
+ *len = rtp_hdr_size + rc;
+ *(uint16_t*)(data+2) = htonl(state->next_seq);
+ *(uint32_t*)(data+4) = htonl(ts_no);
+
+ state->next_seq += 1;
+ state->next_time = ts_no + nsamples;
+
+ return nsamples ? rtp_hdr_size : 0;
+}
@@ -1,16 +1,11 @@
AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir)
AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOGSM_CFLAGS) \
- $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS) \
- $(LIBBCG729_CFLAGS)
+ $(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_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) $(LIBBCG729_LIBS)
-
-noinst_HEADERS = g711common.h mgcp_transcode.h
deleted file mode 100644
@@ -1,187 +0,0 @@
-/*
- * 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));
-}
@@ -31,11 +31,6 @@
#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>
@@ -55,7 +50,7 @@
#include "../../bscconfig.h"
#ifdef BUILD_MGCP_TRANSCODING
-#include "mgcp_transcode.h"
+#include "openbsc/mgcp_transcode.h"
#endif
/* this is here for the vty... it will never be called */
deleted file mode 100644
@@ -1,553 +0,0 @@
-/*
- * (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;
- int dst_packet_duration;
-
- int is_running;
- uint16_t next_seq;
- uint32_t next_time;
- int16_t samples[10*160];
- size_t sample_cnt;
- size_t sample_offs;
-};
-
-int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
-{
- struct mgcp_process_rtp_state *state = state_;
- if (dst)
- return (nsamples >= 0 ?
- nsamples / state->dst_samples_per_frame :
- 1) * state->dst_frame_size;
- else
- return (nsamples >= 0 ?
- nsamples / state->src_samples_per_frame :
- 1) * state->src_frame_size;
-}
-
-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;
- }
-
- if (dst_end->force_output_ptime)
- state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end);
-
- 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;
-}
-
-static int decode_audio(struct mgcp_process_rtp_state *state,
- uint8_t **src, size_t *nbytes)
-{
- while (*nbytes >= state->src_frame_size) {
- if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) {
- LOGP(DMGCP, LOGL_ERROR,
- "Sample buffer too small: %d > %d.\n",
- state->sample_cnt + state->src_samples_per_frame,
- ARRAY_SIZE(state->samples));
- return -ENOSPC;
- }
- switch (state->src_fmt) {
- case AF_GSM:
- if (gsm_decode(state->src.gsm_handle,
- (gsm_byte *)*src, state->samples + state->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, state->samples + state->sample_cnt);
- break;
-#endif
- case AF_PCMA:
- alaw_decode(*src, state->samples + state->sample_cnt,
- state->src_samples_per_frame);
- break;
- case AF_S16:
- memmove(state->samples + state->sample_cnt, *src,
- state->src_frame_size);
- break;
- case AF_L16:
- l16_decode(*src, state->samples + state->sample_cnt,
- state->src_samples_per_frame);
- break;
- default:
- break;
- }
- *src += state->src_frame_size;
- *nbytes -= state->src_frame_size;
- state->sample_cnt += state->src_samples_per_frame;
- }
- return 0;
-}
-
-static int encode_audio(struct mgcp_process_rtp_state *state,
- uint8_t *dst, size_t buf_size, size_t max_samples)
-{
- int nbytes = 0;
- size_t nsamples = 0;
- /* Encode samples into dst */
- while (nsamples + state->dst_samples_per_frame <= max_samples) {
- if (nbytes + state->dst_frame_size > buf_size) {
- if (nbytes > 0)
- break;
-
- /* Not even one frame fits into the buffer */
- LOGP(DMGCP, LOGL_INFO,
- "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,
- state->samples + state->sample_offs, dst);
- break;
-#ifdef HAVE_BCG729
- case AF_G729:
- bcg729Encoder(state->dst.g729_enc,
- state->samples + state->sample_offs, dst);
- break;
-#endif
- case AF_PCMA:
- alaw_encode(state->samples + state->sample_offs, dst,
- state->src_samples_per_frame);
- break;
- case AF_S16:
- memmove(dst, state->samples + state->sample_offs,
- state->dst_frame_size);
- break;
- case AF_L16:
- l16_encode(state->samples + state->sample_offs, dst,
- state->src_samples_per_frame);
- break;
- default:
- break;
- }
- dst += state->dst_frame_size;
- nbytes += state->dst_frame_size;
- state->sample_offs += state->dst_samples_per_frame;
- nsamples += state->dst_samples_per_frame;
- }
- state->sample_cnt -= nsamples;
- return nbytes;
-}
-
-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_idx;
- uint8_t *src = (uint8_t *)payload_data;
- uint8_t *dst = (uint8_t *)payload_data;
- size_t nbytes = payload_len;
- // size_t frame_remainder;
- size_t nsamples;
- size_t max_samples;
- uint32_t ts_no;
- int rc;
-
- if (!state)
- return 0;
-
- if (state->src_fmt == state->dst_fmt) {
- if (!state->dst_packet_duration)
- return 0;
-
- /* TODO: repackage without transcoding */
- }
-
- /* If the remaining samples do not fit into a fixed ptime,
- * a) discard them, if the next packet is much later
- * b) add silence and * send it, if the current packet is not
- * yet too late
- * c) append the sample data, if the timestamp matches exactly
- */
-
- /* TODO: check payload type (-> G.711 comfort noise) */
-
- if (payload_len > 0) {
- ts_no = ntohl(*(uint32_t*)(data+4));
- if (!state->is_running)
- state->next_seq = ntohs(*(uint32_t*)(data+4));
-
- state->is_running = 1;
-
- if (state->sample_cnt > 0) {
- int32_t delta = ts_no - state->next_time;
- /* TODO: check sequence? reordering? packet loss? */
-
- if (delta > state->sample_cnt)
- /* There is a time gap between the last packet
- * and the current one. Just discard the
- * partial data that is left in the buffer.
- * TODO: This can be improved by adding silence
- * instead if the delta is small enough.
- */
- state->sample_cnt = 0;
- else if (delta < 0) {
- LOGP(DMGCP, LOGL_NOTICE,
- "RTP time jumps backwards, delta = %d, "
- "discarding buffered samples\n",
- delta);
- state->sample_cnt = 0;
- state->sample_offs = 0;
- return -EAGAIN;
- }
-
- /* Make sure the samples start without offset */
- fprintf(stderr, "Moving %d samples to buffer start (offset %d)\n", state->sample_cnt, state->sample_offs);
- if (state->sample_offs && state->sample_cnt)
- memmove(&state->samples[0],
- &state->samples[state->sample_offs],
- state->sample_cnt * sizeof(state->samples[0]));
- }
-
- state->sample_offs = 0;
-
- /* Append decoded audio to samples */
- decode_audio(state, &src, &nbytes);
-
- if (nbytes > 0)
- LOGP(DMGCP, LOGL_NOTICE,
- "Skipped audio frame in RTP packet: %d octets\n",
- nbytes);
- } else
- ts_no = state->next_time;
-
- if (state->sample_cnt < state->dst_packet_duration)
- return -EAGAIN;
-
- max_samples =
- state->dst_packet_duration ?
- state->dst_packet_duration : state->sample_cnt;
-
- nsamples = state->sample_cnt;
-
- rc = encode_audio(state, dst, buf_size, max_samples);
- if (rc <= 0)
- return rc;
-
- nsamples -= state->sample_cnt;
- fprintf(stderr, "Wrote %d samples to buffer (offset %d)\n", nsamples, state->sample_offs);
-
- *len = rtp_hdr_size + rc;
- *(uint16_t*)(data+2) = htonl(state->next_seq);
- *(uint32_t*)(data+4) = htonl(ts_no);
-
- state->next_seq += 1;
- state->next_time = ts_no + nsamples;
-
- return nsamples ? rtp_hdr_size : 0;
-}
deleted file mode 100644
@@ -1,36 +0,0 @@
-/*
- * (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);
-
-int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst);
-#endif /* OPENBSC_MGCP_TRANSCODE_H */
@@ -18,7 +18,7 @@ mgcp_test_LDADD = $(top_builddir)/src/libbsc/libbsc.a \
$(LIBOSMOCORE_LIBS) -lrt -lm $(LIBOSMOSCCP_LIBS) $(LIBOSMOVTY_LIBS) \
$(LIBRARY_DL)
-mgcp_transcoding_test_SOURCES = mgcp_transcoding_test.c $(top_builddir)/src/osmo-bsc_mgcp/mgcp_transcode.c
+mgcp_transcoding_test_SOURCES = mgcp_transcoding_test.c
mgcp_transcoding_test_LDADD = \
$(top_builddir)/src/libbsc/libbsc.a \
@@ -17,7 +17,7 @@
#error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)"
#endif
-#include "src/osmo-bsc_mgcp/mgcp_transcode.h"
+#include "openbsc/mgcp_transcode.h"
uint8_t *audio_frame_l16[] = {
};