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