[PATCH 10/11] mgcp: Move transcoding to libmgcp

This patch moves the files relevant to transcoding from src/osmo-bsc_mgcp to src/libmgcp and src/include/openbsc. Makefiles and include directives are being updated accordingly. Sponsored-by: On-Waves ehf --- openbsc/contrib/testconv/Makefile | 3 +- openbsc/contrib/testconv/testconv_main.c | 2 +- openbsc/include/openbsc/Makefile.am | 3 +- openbsc/include/openbsc/mgcp_transcode.h | 36 ++ openbsc/src/libmgcp/Makefile.am | 12 +- openbsc/src/libmgcp/g711common.h | 187 ++++++++++ openbsc/src/libmgcp/mgcp_transcode.c | 553 ++++++++++++++++++++++++++++ openbsc/src/osmo-bsc_mgcp/Makefile.am | 9 +- openbsc/src/osmo-bsc_mgcp/g711common.h | 187 ---------- openbsc/src/osmo-bsc_mgcp/mgcp_main.c | 7 +- openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c | 553 ---------------------------- openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h | 36 -- openbsc/tests/mgcp/Makefile.am | 2 +- openbsc/tests/mgcp/mgcp_transcoding_test.c | 2 +- 14 files changed, 795 insertions(+), 797 deletions(-) create mode 100644 openbsc/include/openbsc/mgcp_transcode.h create mode 100644 openbsc/src/libmgcp/g711common.h create mode 100644 openbsc/src/libmgcp/mgcp_transcode.c delete mode 100644 openbsc/src/osmo-bsc_mgcp/g711common.h delete mode 100644 openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c delete mode 100644 openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h diff --git a/openbsc/contrib/testconv/Makefile b/openbsc/contrib/testconv/Makefile index 90adecc..bb856f7 100644 --- a/openbsc/contrib/testconv/Makefile +++ b/openbsc/contrib/testconv/Makefile @@ -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 $@ $< diff --git a/openbsc/contrib/testconv/testconv_main.c b/openbsc/contrib/testconv/testconv_main.c index e74c686..89dce1a 100644 --- a/openbsc/contrib/testconv/testconv_main.c +++ b/openbsc/contrib/testconv/testconv_main.c @@ -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) { diff --git a/openbsc/include/openbsc/Makefile.am b/openbsc/include/openbsc/Makefile.am index 8f7c1c4..6b08d07 100644 --- a/openbsc/include/openbsc/Makefile.am +++ b/openbsc/include/openbsc/Makefile.am @@ -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 diff --git a/openbsc/include/openbsc/mgcp_transcode.h b/openbsc/include/openbsc/mgcp_transcode.h new file mode 100644 index 0000000..0961634 --- /dev/null +++ b/openbsc/include/openbsc/mgcp_transcode.h @@ -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 */ diff --git a/openbsc/src/libmgcp/Makefile.am b/openbsc/src/libmgcp/Makefile.am index 72f625d..bd02e61 100644 --- a/openbsc/src/libmgcp/Makefile.am +++ b/openbsc/src/libmgcp/Makefile.am @@ -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 diff --git a/openbsc/src/libmgcp/g711common.h b/openbsc/src/libmgcp/g711common.h new file mode 100644 index 0000000..cb35fc6 --- /dev/null +++ b/openbsc/src/libmgcp/g711common.h @@ -0,0 +1,187 @@ +/* + * PCM - A-Law conversion + * Copyright (c) 2000 by Abramo Bagnara &lt;abramo(a)alsa-project.org&gt; + * + * Wrapper for linphone Codec class by Simon Morlat &lt;simon.morlat(a)linphone.org&gt; + * + * + * 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)); +} diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c new file mode 100644 index 0000000..edd3178 --- /dev/null +++ b/openbsc/src/libmgcp/mgcp_transcode.c @@ -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; +} diff --git a/openbsc/src/osmo-bsc_mgcp/Makefile.am b/openbsc/src/osmo-bsc_mgcp/Makefile.am index a620e7a..da02380 100644 --- a/openbsc/src/osmo-bsc_mgcp/Makefile.am +++ b/openbsc/src/osmo-bsc_mgcp/Makefile.am @@ -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 diff --git a/openbsc/src/osmo-bsc_mgcp/g711common.h b/openbsc/src/osmo-bsc_mgcp/g711common.h deleted file mode 100644 index cb35fc6..0000000 --- a/openbsc/src/osmo-bsc_mgcp/g711common.h +++ /dev/null @@ -1,187 +0,0 @@ -/* - * PCM - A-Law conversion - * Copyright (c) 2000 by Abramo Bagnara &lt;abramo(a)alsa-project.org&gt; - * - * Wrapper for linphone Codec class by Simon Morlat &lt;simon.morlat(a)linphone.org&gt; - * - * - * 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)); -} diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c index 5ac4c26..8c3808a 100644 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c +++ b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c @@ -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 */ diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c deleted file mode 100644 index edd3178..0000000 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c +++ /dev/null @@ -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; -} diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h deleted file mode 100644 index 0961634..0000000 --- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h +++ /dev/null @@ -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 */ diff --git a/openbsc/tests/mgcp/Makefile.am b/openbsc/tests/mgcp/Makefile.am index 81f28ae..2bc2da6 100644 --- a/openbsc/tests/mgcp/Makefile.am +++ b/openbsc/tests/mgcp/Makefile.am @@ -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 \ diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c index a1af157..264dbe6 100644 --- a/openbsc/tests/mgcp/mgcp_transcoding_test.c +++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c @@ -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[] = { }; -- 1.7.9.5