| /* |
| * (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; |
| #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 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->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_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; |
| } |
| |
| 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: %d > %d.\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_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: %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, |
| 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_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=%d\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: %d 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; |
| } |