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;
-}