mgcp: Add RTP audio transcoding
This patch implements audio transcoding between the formats GSM,
PCMA, L16, and optionally G.729.
The feature needs to be enabled by using the autoconf option
'--enable-mgcp-transcoding'. In this case mgcp_transcode.c will
be compiled and linked to osmo-bsc_mgcp, and the transcoding
functions provided will be registered as processing callbacks.
If G.729 support is required, libcg729 needs to be installed and
'--with-g729' must be passed to ./configure.
Ticket: OW#1111
Sponsored-by: On-Waves ehf
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
new file mode 100644
index 0000000..cadc876
--- /dev/null
+++ b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
@@ -0,0 +1,452 @@
+/*
+ * (C) 2014 by Sysmocom s.f.m.c. GmbH
+ * (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;
+};
+
+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;
+ enum audio_format src_fmt, dst_fmt;
+
+ /* 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;
+
+ 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;
+ }
+
+ 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;
+}
+
+
+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_cnt = 0;
+ size_t sample_idx;
+ int16_t samples[10*160];
+ uint8_t *src = (uint8_t *)payload_data;
+ uint8_t *dst = (uint8_t *)payload_data;
+ size_t nbytes = payload_len;
+ size_t frame_remainder;
+
+ if (!state)
+ return 0;
+
+ if (state->src_fmt == state->dst_fmt)
+ return 0;
+
+ /* TODO: check payload type (-> G.711 comfort noise) */
+
+ /* Decode src into samples */
+ while (nbytes >= state->src_frame_size) {
+ if (sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(samples)) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Sample buffer too small: %d > %d.\n",
+ sample_cnt + state->src_samples_per_frame,
+ ARRAY_SIZE(samples));
+ return -ENOSPC;
+ }
+ switch (state->src_fmt) {
+ case AF_GSM:
+ if (gsm_decode(state->src.gsm_handle,
+ (gsm_byte *)src, samples + 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, samples + sample_cnt);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_decode(src, samples + sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(samples + sample_cnt, src,
+ state->src_frame_size);
+ break;
+ case AF_L16:
+ l16_decode(src, samples + sample_cnt,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ src += state->src_frame_size;
+ nbytes -= state->src_frame_size;
+ sample_cnt += state->src_samples_per_frame;
+ }
+
+ /* Add silence if necessary */
+ frame_remainder = sample_cnt % state->dst_samples_per_frame;
+ if (frame_remainder) {
+ size_t silence = state->dst_samples_per_frame - frame_remainder;
+ if (sample_cnt + silence > ARRAY_SIZE(samples)) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "Sample buffer too small for silence: %d > %d.\n",
+ sample_cnt + silence,
+ ARRAY_SIZE(samples));
+ return -ENOSPC;
+ }
+
+ while (silence > 0) {
+ samples[sample_cnt] = 0;
+ sample_cnt += 1;
+ silence -= 1;
+ }
+ }
+
+ /* Encode samples into dst */
+ sample_idx = 0;
+ nbytes = 0;
+ while (sample_idx + state->dst_samples_per_frame <= sample_cnt) {
+ if (nbytes + state->dst_frame_size > buf_size) {
+ LOGP(DMGCP, LOGL_ERROR,
+ "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,
+ samples + sample_idx, dst);
+ break;
+#ifdef HAVE_BCG729
+ case AF_G729:
+ bcg729Encoder(state->dst.g729_enc,
+ samples + sample_idx, dst);
+ break;
+#endif
+ case AF_PCMA:
+ alaw_encode(samples + sample_idx, dst,
+ state->src_samples_per_frame);
+ break;
+ case AF_S16:
+ memmove(dst, samples + sample_idx, state->dst_frame_size);
+ break;
+ case AF_L16:
+ l16_encode(samples + sample_idx, dst,
+ state->src_samples_per_frame);
+ break;
+ default:
+ break;
+ }
+ dst += state->dst_frame_size;
+ nbytes += state->dst_frame_size;
+ sample_idx += state->dst_samples_per_frame;
+ }
+
+ *len = rtp_hdr_size + nbytes;
+ /* Patch payload type */
+ data[1] = (data[1] & 0x80) | (dst_end->payload_type & 0x7f);
+
+ return 0;
+}