move openbsc/* to repos root

This is the first step in creating this repository from the legacy openbsc.git.

Like all other Osmocom repositories, keep the autoconf and automake files in
the repository root. openbsc.git has been the sole exception, which ends now.

Change-Id: I9c6f2a448d9cb1cc088cf1cf6918b69d7e69b4e7

1 files changed, 0 insertions, 612 deletions

diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
deleted file mode 100644
index f31e7aefb..000000000
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ /dev/null
@@ -1,612 +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 "g711common.h"
-
-#include <openbsc/debug.h>
-#include <openbsc/mgcp.h>
-#include <openbsc/mgcp_internal.h>
-#include <openbsc/mgcp_transcode.h>
-
-#include <osmocom/core/talloc.h>
-#include <osmocom/netif/rtp.h>
-
-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_codec *codec)
-{
-	if (codec->subtype_name) {
-		if (!strcasecmp("GSM", codec->subtype_name))
-			return AF_GSM;
-		if (!strcasecmp("PCMA", codec->subtype_name))
-			return AF_PCMA;
-		if (!strcasecmp("PCMU", codec->subtype_name))
-			return AF_PCMU;
-#ifdef HAVE_BCG729
-		if (!strcasecmp("G729", codec->subtype_name))
-			return AF_G729;
-#endif
-		if (!strcasecmp("L16", codec->subtype_name))
-			return AF_L16;
-	}
-
-	switch (codec->payload_type) {
-	case 0 /* PCMU */:
-		return AF_PCMU;
-	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 void ulaw_encode(short *sample, unsigned char *buf, size_t n)
-{
-	for (; n > 0; --n)
-		*(buf++) = s16_to_ulaw(*(sample++));
-}
-
-static void ulaw_decode(unsigned char *buf, short *sample, size_t n)
-{
-	for (; n > 0; --n)
-		*(sample++) = ulaw_to_s16(*(buf++));
-}
-
-static int processing_state_destructor(struct mgcp_process_rtp_state *state)
-{
-	switch (state->src_fmt) {
-	case AF_GSM:
-		if (state->src.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;
-	enum audio_format src_fmt, dst_fmt;
-	const struct mgcp_rtp_codec *dst_codec = &dst_end->codec;
-
-	/* cleanup first */
-	if (dst_end->rtp_process_data) {
-		talloc_free(dst_end->rtp_process_data);
-		dst_end->rtp_process_data = NULL;
-	}
-
-	if (!src_end)
-		return 0;
-
-	const struct mgcp_rtp_codec *src_codec = &src_end->codec;
-
-	if (endp->tcfg->no_audio_transcoding) {
-		LOGP(DMGCP, LOGL_NOTICE,
-			"Transcoding disabled on endpoint 0x%x\n",
-			ENDPOINT_NUMBER(endp));
-		return 0;
-	}
-
-	src_fmt = get_audio_format(src_codec);
-	dst_fmt = get_audio_format(dst_codec);
-
-	LOGP(DMGCP, LOGL_ERROR,
-	     "Checking transcoding: %s (%d) -> %s (%d)\n",
-	     src_codec->subtype_name, src_codec->payload_type,
-	     dst_codec->subtype_name, dst_codec->payload_type);
-
-	if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
-		if (!src_codec->subtype_name || !dst_codec->subtype_name)
-			/* Not enough info, do nothing */
-			return 0;
-
-		if (strcasecmp(src_codec->subtype_name, dst_codec->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_codec->audio_name, dst_codec->audio_name);
-		return -EINVAL;
-	}
-
-	if (src_codec->rate && dst_codec->rate && src_codec->rate != dst_codec->rate) {
-		LOGP(DMGCP, LOGL_ERROR,
-		     "Cannot transcode: rate conversion (%d -> %d) not supported.\n",
-		     src_codec->rate, dst_codec->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_PCMU:
-	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_PCMU:
-	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_codec->payload_type, src_codec->rate, src_end->fmtp_extra,
-	     dst_fmt, dst_codec->payload_type, dst_codec->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;
-	struct mgcp_rtp_codec *net_codec = &endp->net_end.codec;
-	struct mgcp_rtp_codec *bts_codec = &endp->bts_end.codec;
-
-	if (!state || net_codec->payload_type < 0) {
-		*payload_type = bts_codec->payload_type;
-		*audio_name = bts_codec->audio_name;
-		*fmtp_extra = endp->bts_end.fmtp_extra;
-		return;
-	}
-
-	*payload_type = net_codec->payload_type;
-	*audio_name = net_codec->audio_name;
-	*fmtp_extra = endp->net_end.fmtp_extra;
-}
-
-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: %zu > %zu.\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_PCMU:
-			ulaw_decode(*src, state->samples + state->sample_cnt,
-				    state->src_samples_per_frame);
-			break;
-		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: %zu > %zu.\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_PCMU:
-			ulaw_encode(state->samples + state->sample_offs, dst,
-				    state->src_samples_per_frame);
-			break;
-		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;
-}
-
-static struct mgcp_rtp_end *source_for_dest(struct mgcp_endpoint *endp,
-					struct mgcp_rtp_end *dst_end)
-{
-	if (&endp->bts_end == dst_end)
-		return &endp->net_end;
-	else if (&endp->net_end == dst_end)
-		return &endp->bts_end;
-	OSMO_ASSERT(0);
-}
-
-/*
- * With some modems we get offered multiple codecs
- * and we have selected one of them. It might not
- * be the right one and we need to detect this with
- * the first audio packets. One difficulty is that
- * we patch the rtp payload type in place, so we
- * need to discuss this.
- */
-struct mgcp_process_rtp_state *check_transcode_state(
-				struct mgcp_endpoint *endp,
-				struct mgcp_rtp_end *dst_end,
-				struct rtp_hdr *rtp_hdr)
-{
-	struct mgcp_rtp_end *src_end;
-
-	/* Only deal with messages from net to bts */
-	if (&endp->bts_end != dst_end)
-		goto done;
-
-	src_end = source_for_dest(endp, dst_end);
-
-	/* Already patched */
-	if (rtp_hdr->payload_type == dst_end->codec.payload_type)
-		goto done;
-	/* The payload we expect */
-	if (rtp_hdr->payload_type == src_end->codec.payload_type)
-		goto done;
-	/* The matching alternate payload type? Then switch */
-	if (rtp_hdr->payload_type == src_end->alt_codec.payload_type) {
-		struct mgcp_config *cfg = endp->cfg;
-		struct mgcp_rtp_codec tmp_codec = src_end->alt_codec;
-		src_end->alt_codec = src_end->codec;
-		src_end->codec = tmp_codec;
-		cfg->setup_rtp_processing_cb(endp, &endp->net_end, &endp->bts_end);
-		cfg->setup_rtp_processing_cb(endp, &endp->bts_end, &endp->net_end);
-	}
-
-done:
-	return dst_end->rtp_process_data;
-}
-
-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;
-	const size_t rtp_hdr_size = sizeof(struct rtp_hdr);
-	struct rtp_hdr *rtp_hdr = (struct rtp_hdr *) data;
-	char *payload_data = (char *) &rtp_hdr->data[0];
-	int payload_len = *len - rtp_hdr_size;
-	uint8_t *src = (uint8_t *)payload_data;
-	uint8_t *dst = (uint8_t *)payload_data;
-	size_t nbytes = payload_len;
-	size_t nsamples;
-	size_t max_samples;
-	uint32_t ts_no;
-	int rc;
-
-	state = check_transcode_state(endp, dst_end, rtp_hdr);
-	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(rtp_hdr->timestamp);
-		if (!state->is_running) {
-			state->next_seq = ntohs(rtp_hdr->sequence);
-			state->next_time = ts_no;
-			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.
-				 */
-				LOGP(DMGCP, LOGL_NOTICE,
-					"0x%x dropping sample buffer due delta=%d sample_cnt=%zu\n",
-					ENDPOINT_NUMBER(endp), delta, state->sample_cnt);
-				state->sample_cnt = 0;
-				state->next_time = ts_no;
-			} 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 */
-			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: %zu 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);
-	/*
-	 * There were no samples to encode?
-	 * TODO: how does this work for comfort noise?
-	 */
-	if (rc == 0)
-		return -ENOMSG;
-	/* Any other error during the encoding */
-	if (rc < 0)
-		return rc;
-
-	nsamples -= state->sample_cnt;
-
-	*len = rtp_hdr_size + rc;
-	rtp_hdr->sequence = htons(state->next_seq);
-	rtp_hdr->timestamp = htonl(ts_no);
-
-	state->next_seq += 1;
-	state->next_time = ts_no + nsamples;
-
-	/*
-	 * XXX: At this point we should always have consumed
-	 * samples. So doing OSMO_ASSERT(nsamples > 0) and returning
-	 * rtp_hdr_size should be fine.
-	 */
-	return nsamples ? rtp_hdr_size : 0;
-}