466 lines
11 KiB
C
466 lines
11 KiB
C
/*
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* (C) 2014 by Sysmocom s.f.m.c. GmbH
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* (C) 2014 by On-Waves
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* All Rights Reserved
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Affero General Public License as published by
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* the Free Software Foundation; either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Affero General Public License for more details.
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*
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* You should have received a copy of the GNU Affero General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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*/
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#include <stdlib.h>
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#include <string.h>
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#include <errno.h>
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#include "bscconfig.h"
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#include "g711common.h"
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#include <gsm.h>
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#ifdef HAVE_BCG729
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#include <bcg729/decoder.h>
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#include <bcg729/encoder.h>
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#endif
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#include <openbsc/debug.h>
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#include <openbsc/mgcp.h>
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#include <openbsc/mgcp_internal.h>
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#include <osmocom/core/talloc.h>
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enum audio_format {
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AF_INVALID,
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AF_S16,
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AF_L16,
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AF_GSM,
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AF_G729,
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AF_PCMA
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};
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struct mgcp_process_rtp_state {
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/* decoding */
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enum audio_format src_fmt;
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union {
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gsm gsm_handle;
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#ifdef HAVE_BCG729
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bcg729DecoderChannelContextStruct *g729_dec;
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#endif
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} src;
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size_t src_frame_size;
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size_t src_samples_per_frame;
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/* processing */
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/* encoding */
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enum audio_format dst_fmt;
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union {
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gsm gsm_handle;
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#ifdef HAVE_BCG729
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bcg729EncoderChannelContextStruct *g729_enc;
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#endif
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} dst;
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size_t dst_frame_size;
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size_t dst_samples_per_frame;
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};
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int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
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{
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struct mgcp_process_rtp_state *state = state_;
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if (dst)
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return (nsamples >= 0 ?
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nsamples / state->dst_samples_per_frame :
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1) * state->dst_frame_size;
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else
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return (nsamples >= 0 ?
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nsamples / state->src_samples_per_frame :
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1) * state->src_frame_size;
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}
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static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
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{
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if (rtp_end->subtype_name) {
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if (!strcmp("GSM", rtp_end->subtype_name))
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return AF_GSM;
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if (!strcmp("PCMA", rtp_end->subtype_name))
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return AF_PCMA;
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#ifdef HAVE_BCG729
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if (!strcmp("G729", rtp_end->subtype_name))
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return AF_G729;
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#endif
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if (!strcmp("L16", rtp_end->subtype_name))
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return AF_L16;
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}
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switch (rtp_end->payload_type) {
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case 3 /* GSM */:
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return AF_GSM;
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case 8 /* PCMA */:
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return AF_PCMA;
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#ifdef HAVE_BCG729
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case 18 /* G.729 */:
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return AF_G729;
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#endif
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case 11 /* L16 */:
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return AF_L16;
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default:
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return AF_INVALID;
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}
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}
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static void l16_encode(short *sample, unsigned char *buf, size_t n)
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{
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for (; n > 0; --n, ++sample, buf += 2) {
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buf[0] = sample[0] >> 8;
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buf[1] = sample[0] & 0xff;
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}
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}
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static void l16_decode(unsigned char *buf, short *sample, size_t n)
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{
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for (; n > 0; --n, ++sample, buf += 2)
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sample[0] = ((short)buf[0] << 8) | buf[1];
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}
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static void alaw_encode(short *sample, unsigned char *buf, size_t n)
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{
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for (; n > 0; --n)
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*(buf++) = s16_to_alaw(*(sample++));
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}
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static void alaw_decode(unsigned char *buf, short *sample, size_t n)
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{
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for (; n > 0; --n)
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*(sample++) = alaw_to_s16(*(buf++));
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}
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static int processing_state_destructor(struct mgcp_process_rtp_state *state)
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{
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switch (state->src_fmt) {
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case AF_GSM:
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if (state->dst.gsm_handle)
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gsm_destroy(state->src.gsm_handle);
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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if (state->src.g729_dec)
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closeBcg729DecoderChannel(state->src.g729_dec);
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break;
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#endif
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default:
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break;
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}
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switch (state->dst_fmt) {
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case AF_GSM:
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if (state->dst.gsm_handle)
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gsm_destroy(state->dst.gsm_handle);
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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if (state->dst.g729_enc)
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closeBcg729EncoderChannel(state->dst.g729_enc);
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break;
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#endif
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default:
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break;
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}
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return 0;
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}
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int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
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struct mgcp_rtp_end *dst_end,
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struct mgcp_rtp_end *src_end)
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{
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struct mgcp_process_rtp_state *state;
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enum audio_format src_fmt, dst_fmt;
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/* cleanup first */
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if (dst_end->rtp_process_data) {
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talloc_free(dst_end->rtp_process_data);
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dst_end->rtp_process_data = NULL;
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}
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if (!src_end)
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return 0;
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src_fmt = get_audio_format(src_end);
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dst_fmt = get_audio_format(dst_end);
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LOGP(DMGCP, LOGL_ERROR,
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"Checking transcoding: %s (%d) -> %s (%d)\n",
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src_end->subtype_name, src_end->payload_type,
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dst_end->subtype_name, dst_end->payload_type);
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if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
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if (!src_end->subtype_name || !dst_end->subtype_name)
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/* Not enough info, do nothing */
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return 0;
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if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
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/* Nothing to do */
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return 0;
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LOGP(DMGCP, LOGL_ERROR,
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"Cannot transcode: %s codec not supported (%s -> %s).\n",
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src_fmt != AF_INVALID ? "destination" : "source",
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src_end->audio_name, dst_end->audio_name);
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return -EINVAL;
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}
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if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
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LOGP(DMGCP, LOGL_ERROR,
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"Cannot transcode: rate conversion (%d -> %d) not supported.\n",
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src_end->rate, dst_end->rate);
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return -EINVAL;
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}
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state = talloc_zero(endp->tcfg->cfg, struct mgcp_process_rtp_state);
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talloc_set_destructor(state, processing_state_destructor);
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dst_end->rtp_process_data = state;
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state->src_fmt = src_fmt;
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switch (state->src_fmt) {
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case AF_L16:
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case AF_S16:
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state->src_frame_size = 80 * sizeof(short);
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state->src_samples_per_frame = 80;
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break;
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case AF_GSM:
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state->src_frame_size = sizeof(gsm_frame);
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state->src_samples_per_frame = 160;
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state->src.gsm_handle = gsm_create();
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if (!state->src.gsm_handle) {
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LOGP(DMGCP, LOGL_ERROR,
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"Failed to initialize GSM decoder.\n");
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return -EINVAL;
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}
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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state->src_frame_size = 10;
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state->src_samples_per_frame = 80;
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state->src.g729_dec = initBcg729DecoderChannel();
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if (!state->src.g729_dec) {
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LOGP(DMGCP, LOGL_ERROR,
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"Failed to initialize G.729 decoder.\n");
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return -EINVAL;
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}
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break;
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#endif
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case AF_PCMA:
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state->src_frame_size = 80;
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state->src_samples_per_frame = 80;
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break;
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default:
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break;
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}
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state->dst_fmt = dst_fmt;
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switch (state->dst_fmt) {
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case AF_L16:
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case AF_S16:
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state->dst_frame_size = 80*sizeof(short);
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state->dst_samples_per_frame = 80;
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break;
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case AF_GSM:
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state->dst_frame_size = sizeof(gsm_frame);
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state->dst_samples_per_frame = 160;
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state->dst.gsm_handle = gsm_create();
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if (!state->dst.gsm_handle) {
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LOGP(DMGCP, LOGL_ERROR,
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"Failed to initialize GSM encoder.\n");
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return -EINVAL;
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}
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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state->dst_frame_size = 10;
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state->dst_samples_per_frame = 80;
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state->dst.g729_enc = initBcg729EncoderChannel();
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if (!state->dst.g729_enc) {
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LOGP(DMGCP, LOGL_ERROR,
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"Failed to initialize G.729 decoder.\n");
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return -EINVAL;
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}
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break;
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#endif
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case AF_PCMA:
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state->dst_frame_size = 80;
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state->dst_samples_per_frame = 80;
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break;
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default:
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break;
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}
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LOGP(DMGCP, LOGL_INFO,
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"Initialized RTP processing on: 0x%x "
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"conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n",
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ENDPOINT_NUMBER(endp),
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src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra,
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dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra);
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return 0;
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}
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void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
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int *payload_type,
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const char**audio_name,
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const char**fmtp_extra)
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{
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struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data;
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if (!state || endp->net_end.payload_type < 0) {
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*payload_type = endp->bts_end.payload_type;
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*audio_name = endp->bts_end.audio_name;
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*fmtp_extra = endp->bts_end.fmtp_extra;
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return;
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}
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*payload_type = endp->net_end.payload_type;
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*fmtp_extra = endp->net_end.fmtp_extra;
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*audio_name = endp->net_end.audio_name;
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}
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int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
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struct mgcp_rtp_end *dst_end,
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char *data, int *len, int buf_size)
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{
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struct mgcp_process_rtp_state *state = dst_end->rtp_process_data;
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size_t rtp_hdr_size = 12;
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char *payload_data = data + rtp_hdr_size;
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int payload_len = *len - rtp_hdr_size;
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size_t sample_cnt = 0;
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size_t sample_idx;
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int16_t samples[10*160];
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uint8_t *src = (uint8_t *)payload_data;
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uint8_t *dst = (uint8_t *)payload_data;
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size_t nbytes = payload_len;
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size_t frame_remainder;
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if (!state)
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return 0;
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if (state->src_fmt == state->dst_fmt)
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return 0;
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/* TODO: check payload type (-> G.711 comfort noise) */
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/* Decode src into samples */
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while (nbytes >= state->src_frame_size) {
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if (sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(samples)) {
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LOGP(DMGCP, LOGL_ERROR,
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"Sample buffer too small: %d > %d.\n",
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sample_cnt + state->src_samples_per_frame,
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ARRAY_SIZE(samples));
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return -ENOSPC;
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}
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switch (state->src_fmt) {
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case AF_GSM:
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if (gsm_decode(state->src.gsm_handle,
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(gsm_byte *)src, samples + sample_cnt) < 0) {
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LOGP(DMGCP, LOGL_ERROR,
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"Failed to decode GSM.\n");
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return -EINVAL;
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}
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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bcg729Decoder(state->src.g729_dec, src, 0, samples + sample_cnt);
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break;
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#endif
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case AF_PCMA:
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alaw_decode(src, samples + sample_cnt,
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state->src_samples_per_frame);
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break;
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case AF_S16:
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memmove(samples + sample_cnt, src,
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state->src_frame_size);
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break;
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case AF_L16:
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l16_decode(src, samples + sample_cnt,
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state->src_samples_per_frame);
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break;
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default:
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break;
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}
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src += state->src_frame_size;
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nbytes -= state->src_frame_size;
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sample_cnt += state->src_samples_per_frame;
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}
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/* Add silence if necessary */
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frame_remainder = sample_cnt % state->dst_samples_per_frame;
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if (frame_remainder) {
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size_t silence = state->dst_samples_per_frame - frame_remainder;
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if (sample_cnt + silence > ARRAY_SIZE(samples)) {
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LOGP(DMGCP, LOGL_ERROR,
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"Sample buffer too small for silence: %d > %d.\n",
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sample_cnt + silence,
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ARRAY_SIZE(samples));
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return -ENOSPC;
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}
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while (silence > 0) {
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samples[sample_cnt] = 0;
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sample_cnt += 1;
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silence -= 1;
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}
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}
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/* Encode samples into dst */
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sample_idx = 0;
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nbytes = 0;
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while (sample_idx + state->dst_samples_per_frame <= sample_cnt) {
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if (nbytes + state->dst_frame_size > buf_size) {
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LOGP(DMGCP, LOGL_ERROR,
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"Encoding (RTP) buffer too small: %d > %d.\n",
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nbytes + state->dst_frame_size, buf_size);
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return -ENOSPC;
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}
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switch (state->dst_fmt) {
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case AF_GSM:
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gsm_encode(state->dst.gsm_handle,
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samples + sample_idx, dst);
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break;
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#ifdef HAVE_BCG729
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case AF_G729:
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bcg729Encoder(state->dst.g729_enc,
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samples + sample_idx, dst);
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break;
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#endif
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case AF_PCMA:
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alaw_encode(samples + sample_idx, dst,
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state->src_samples_per_frame);
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break;
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case AF_S16:
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memmove(dst, samples + sample_idx, state->dst_frame_size);
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break;
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case AF_L16:
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l16_encode(samples + sample_idx, dst,
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state->src_samples_per_frame);
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break;
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default:
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break;
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}
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dst += state->dst_frame_size;
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nbytes += state->dst_frame_size;
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sample_idx += state->dst_samples_per_frame;
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}
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*len = rtp_hdr_size + nbytes;
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/* Patch payload type */
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data[1] = (data[1] & 0x80) | (dst_end->payload_type & 0x7f);
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return 0;
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}
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