diff options
Diffstat (limited to 'src/libmgcp/mgcp_transcode.c')
| -rw-r--r-- | src/libmgcp/mgcp_transcode.c | 612 |
1 files changed, 612 insertions, 0 deletions
diff --git a/src/libmgcp/mgcp_transcode.c b/src/libmgcp/mgcp_transcode.c new file mode 100644 index 000000000..f31e7aefb --- /dev/null +++ b/src/libmgcp/mgcp_transcode.c @@ -0,0 +1,612 @@ +/* + * (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; +} |