mgcp: Move the "codec" params to a struct
We might be offered multiple codecs by the remote and need to switch between them once we receive data. Do this by moving it to a struct so we can separate between proposed and current codec. In SDP we can have multiple codecs but a global ptime. The current code doesn't separate that clearly instead we write it to the main codec.
This commit is contained in:
parent
3713f78ac2
commit
cac2438b0c
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@ -64,16 +64,16 @@ int main(int argc, char **argv)
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if (argc <= 2)
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errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16} [SPP]");
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if ((src_end->payload_type = audio_name_to_type(argv[1])) == -1)
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if ((src_end->codec.payload_type = audio_name_to_type(argv[1])) == -1)
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errx(1, "invalid input format '%s'", argv[1]);
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if ((dst_end->payload_type = audio_name_to_type(argv[2])) == -1)
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if ((dst_end->codec.payload_type = audio_name_to_type(argv[2])) == -1)
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errx(1, "invalid output format '%s'", argv[2]);
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if (argc > 3)
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out_samples = atoi(argv[3]);
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if (out_samples) {
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dst_end->frame_duration_den = dst_end->rate;
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dst_end->frame_duration_num = out_samples;
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dst_end->codec.frame_duration_den = dst_end->codec.rate;
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dst_end->codec.frame_duration_num = out_samples;
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dst_end->frames_per_packet = 1;
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}
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@ -87,7 +87,7 @@ int main(int argc, char **argv)
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in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
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OSMO_ASSERT(sizeof(buf) >= in_size + 12);
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buf[1] = src_end->payload_type;
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buf[1] = src_end->codec.payload_type;
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*(uint16_t*)(buf+2) = htons(1);
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*(uint32_t*)(buf+4) = htonl(0);
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*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
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@ -67,6 +67,17 @@ struct mgcp_rtp_state {
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struct mgcp_rtp_stream_state out_stream;
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};
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struct mgcp_rtp_codec {
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uint32_t rate;
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int channels;
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uint32_t frame_duration_num;
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uint32_t frame_duration_den;
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int payload_type;
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char *audio_name;
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char *subtype_name;
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};
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struct mgcp_rtp_end {
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/* statistics */
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unsigned int packets;
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@ -77,17 +88,13 @@ struct mgcp_rtp_end {
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/* in network byte order */
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int rtp_port, rtcp_port;
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/* audio codec information */
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struct mgcp_rtp_codec codec;
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/* per endpoint data */
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int payload_type;
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uint32_t rate;
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int channels;
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uint32_t frame_duration_num;
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uint32_t frame_duration_den;
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int frames_per_packet;
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uint32_t packet_duration_ms;
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char *fmtp_extra;
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char *audio_name;
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char *subtype_name;
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int output_enabled;
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int force_output_ptime;
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@ -239,7 +239,7 @@ static int adjust_rtp_timestamp_offset(struct mgcp_endpoint *endp,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
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endp->conn_mode);
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} else {
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tsdelta = rtp_end->rate * 20 / 1000;
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tsdelta = rtp_end->codec.rate * 20 / 1000;
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LOGP(DMGCP, LOGL_NOTICE,
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"Fixed packet duration and last timestamp delta "
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"are not available on 0x%x, "
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@ -326,8 +326,8 @@ void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
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/* Use the BTS side parameters when passing the SDP data (for
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* downlink) to the net peer.
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*/
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*payload_type = endp->bts_end.payload_type;
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*audio_name = endp->bts_end.audio_name;
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*payload_type = endp->bts_end.codec.payload_type;
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*audio_name = endp->bts_end.codec.audio_name;
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*fmtp_extra = endp->bts_end.fmtp_extra;
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}
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@ -348,7 +348,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
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uint16_t seq, udelta;
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uint32_t timestamp, ssrc;
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struct rtp_hdr *rtp_hdr;
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int payload = rtp_end->payload_type;
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int payload = rtp_end->codec.payload_type;
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if (len < sizeof(*rtp_hdr))
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return;
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@ -356,7 +356,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
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rtp_hdr = (struct rtp_hdr *) data;
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seq = ntohs(rtp_hdr->sequence);
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timestamp = ntohl(rtp_hdr->timestamp);
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arrival_time = get_current_ts(rtp_end->rate);
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arrival_time = get_current_ts(rtp_end->codec.rate);
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ssrc = ntohl(rtp_hdr->ssrc);
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transit = arrival_time - timestamp;
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@ -380,7 +380,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
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endp->conn_mode);
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if (state->packet_duration == 0) {
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state->packet_duration = rtp_end->rate * 20 / 1000;
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state->packet_duration = rtp_end->codec.rate * 20 / 1000;
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LOGP(DMGCP, LOGL_NOTICE,
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"Fixed packet duration is not available on 0x%x, "
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"using fixed 20ms instead: %d from %s:%d in %d\n",
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@ -598,20 +598,20 @@ static int parse_conn_mode(const char *msg, struct mgcp_endpoint *endp)
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return ret;
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}
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static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
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static int set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
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int payload_type, const char *audio_name)
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{
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int rate = rtp->rate;
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int channels = rtp->channels;
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int rate = codec->rate;
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int channels = codec->channels;
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char audio_codec[64];
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talloc_free(rtp->subtype_name);
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rtp->subtype_name = NULL;
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talloc_free(rtp->audio_name);
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rtp->audio_name = NULL;
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talloc_free(codec->subtype_name);
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codec->subtype_name = NULL;
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talloc_free(codec->audio_name);
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codec->audio_name = NULL;
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if (payload_type != PTYPE_UNDEFINED)
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rtp->payload_type = payload_type;
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codec->payload_type = payload_type;
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if (!audio_name) {
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switch (payload_type) {
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@ -630,17 +630,17 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
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audio_codec, &rate, &channels) < 1)
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return -EINVAL;
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rtp->rate = rate;
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rtp->channels = channels;
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rtp->subtype_name = talloc_strdup(ctx, audio_codec);
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rtp->audio_name = talloc_strdup(ctx, audio_name);
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codec->rate = rate;
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codec->channels = channels;
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codec->subtype_name = talloc_strdup(ctx, audio_codec);
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codec->audio_name = talloc_strdup(ctx, audio_name);
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if (!strcmp(audio_codec, "G729")) {
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rtp->frame_duration_num = 10;
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rtp->frame_duration_den = 1000;
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codec->frame_duration_num = 10;
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codec->frame_duration_den = 1000;
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} else {
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rtp->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
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rtp->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
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codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
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codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
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}
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if (payload_type < 0) {
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@ -654,7 +654,7 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
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payload_type = 18;
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}
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rtp->payload_type = payload_type;
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codec->payload_type = payload_type;
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}
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if (channels != 1)
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@ -761,7 +761,7 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
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if (payload != audio_payload)
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break;
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set_audio_info(p->cfg, rtp, payload, audio_name);
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set_audio_info(p->cfg, &rtp->codec, payload, audio_name);
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} else if (sscanf(line, "a=ptime:%d-%d",
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&ptime, &ptime2) >= 1) {
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if (ptime2 > 0 && ptime2 != ptime)
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@ -769,8 +769,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
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else
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rtp->packet_duration_ms = ptime;
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} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
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if (ptime2 * rtp->frame_duration_den >
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rtp->frame_duration_num * 1500)
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if (ptime2 * rtp->codec.frame_duration_den >
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rtp->codec.frame_duration_num * 1500)
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/* more than 1 frame */
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rtp->packet_duration_ms = 0;
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}
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@ -785,7 +785,7 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
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rtp->rtp_port = htons(port);
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rtp->rtcp_port = htons(port + 1);
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found_media = 1;
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set_audio_info(p->cfg, rtp, audio_payload, NULL);
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set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
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}
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break;
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}
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@ -814,8 +814,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
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LOGP(DMGCP, LOGL_NOTICE,
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"Got media info via SDP: port %d, payload %d (%s), "
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"duration %d, addr %s\n",
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ntohs(rtp->rtp_port), rtp->payload_type,
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rtp->subtype_name ? rtp->subtype_name : "unknown",
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ntohs(rtp->rtp_port), rtp->codec.payload_type,
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rtp->codec.subtype_name ? rtp->codec.subtype_name : "unknown",
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rtp->packet_duration_ms, inet_ntoa(rtp->addr));
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return found_media;
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@ -872,13 +872,13 @@ uint32_t mgcp_rtp_packet_duration(struct mgcp_endpoint *endp,
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/* Get the number of frames per channel and packet */
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if (rtp->frames_per_packet)
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f = rtp->frames_per_packet;
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else if (rtp->packet_duration_ms && rtp->frame_duration_num) {
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int den = 1000 * rtp->frame_duration_num;
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f = (rtp->packet_duration_ms * rtp->frame_duration_den + den/2)
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else if (rtp->packet_duration_ms && rtp->codec.frame_duration_num) {
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int den = 1000 * rtp->codec.frame_duration_num;
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f = (rtp->packet_duration_ms * rtp->codec.frame_duration_den + den/2)
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/ den;
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}
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return rtp->rate * f * rtp->frame_duration_num / rtp->frame_duration_den;
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return rtp->codec.rate * f * rtp->codec.frame_duration_num / rtp->codec.frame_duration_den;
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}
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static int mgcp_parse_osmux_cid(const char *line)
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@ -1025,13 +1025,13 @@ mgcp_header_done:
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endp->allocated = 1;
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/* set up RTP media parameters */
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set_audio_info(p->cfg, &endp->bts_end, tcfg->audio_payload, tcfg->audio_name);
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set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
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endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints,
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tcfg->audio_fmtp_extra);
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if (have_sdp)
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parse_sdp_data(&endp->net_end, p);
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else if (endp->local_options.codec)
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set_audio_info(p->cfg, &endp->net_end,
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set_audio_info(p->cfg, &endp->net_end.codec,
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PTYPE_UNDEFINED, endp->local_options.codec);
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if (p->cfg->bts_force_ptime) {
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@ -1147,7 +1147,7 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p)
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local_options);
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if (!have_sdp && endp->local_options.codec)
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set_audio_info(p->cfg, &endp->net_end,
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set_audio_info(p->cfg, &endp->net_end.codec,
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PTYPE_UNDEFINED, endp->local_options.codec);
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setup_rtp_processing(endp);
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@ -1461,24 +1461,24 @@ static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end)
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end->dropped_packets = 0;
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memset(&end->addr, 0, sizeof(end->addr));
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end->rtp_port = end->rtcp_port = 0;
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end->payload_type = -1;
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end->codec.payload_type = -1;
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end->local_alloc = -1;
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talloc_free(end->fmtp_extra);
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end->fmtp_extra = NULL;
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talloc_free(end->subtype_name);
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end->subtype_name = NULL;
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talloc_free(end->audio_name);
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end->audio_name = NULL;
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talloc_free(end->codec.subtype_name);
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end->codec.subtype_name = NULL;
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talloc_free(end->codec.audio_name);
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end->codec.audio_name = NULL;
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talloc_free(end->rtp_process_data);
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end->rtp_process_data = NULL;
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/* Set default values */
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end->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
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end->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
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end->codec.frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
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end->codec.frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
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end->frames_per_packet = 0; /* unknown */
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end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
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end->rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
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end->channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
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end->codec.rate = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
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end->codec.channels = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
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end->output_enabled = 0;
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}
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@ -45,22 +45,22 @@ int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
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1) * state->src_frame_size;
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}
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static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
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static enum audio_format get_audio_format(const struct mgcp_rtp_codec *codec)
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{
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if (rtp_end->subtype_name) {
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if (!strcmp("GSM", rtp_end->subtype_name))
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if (codec->subtype_name) {
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if (!strcmp("GSM", codec->subtype_name))
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return AF_GSM;
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if (!strcmp("PCMA", rtp_end->subtype_name))
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if (!strcmp("PCMA", codec->subtype_name))
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return AF_PCMA;
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#ifdef HAVE_BCG729
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if (!strcmp("G729", rtp_end->subtype_name))
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if (!strcmp("G729", codec->subtype_name))
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return AF_G729;
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#endif
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if (!strcmp("L16", rtp_end->subtype_name))
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if (!strcmp("L16", codec->subtype_name))
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return AF_L16;
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}
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switch (rtp_end->payload_type) {
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switch (codec->payload_type) {
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case 3 /* GSM */:
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return AF_GSM;
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case 8 /* PCMA */:
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@ -141,6 +141,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
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{
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struct mgcp_process_rtp_state *state;
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enum audio_format src_fmt, dst_fmt;
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const struct mgcp_rtp_codec *src_codec = &src_end->codec;
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const struct mgcp_rtp_codec *dst_codec = &dst_end->codec;
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/* cleanup first */
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if (dst_end->rtp_process_data) {
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@ -151,34 +153,34 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
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if (!src_end)
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return 0;
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src_fmt = get_audio_format(src_end);
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dst_fmt = get_audio_format(dst_end);
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src_fmt = get_audio_format(src_codec);
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dst_fmt = get_audio_format(dst_codec);
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LOGP(DMGCP, LOGL_ERROR,
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"Checking transcoding: %s (%d) -> %s (%d)\n",
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src_end->subtype_name, src_end->payload_type,
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dst_end->subtype_name, dst_end->payload_type);
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src_codec->subtype_name, src_codec->payload_type,
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dst_codec->subtype_name, dst_codec->payload_type);
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if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
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if (!src_end->subtype_name || !dst_end->subtype_name)
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if (!src_codec->subtype_name || !dst_codec->subtype_name)
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/* Not enough info, do nothing */
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return 0;
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if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
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if (strcmp(src_codec->subtype_name, dst_codec->subtype_name) == 0)
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/* Nothing to do */
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return 0;
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LOGP(DMGCP, LOGL_ERROR,
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"Cannot transcode: %s codec not supported (%s -> %s).\n",
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src_fmt != AF_INVALID ? "destination" : "source",
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src_end->audio_name, dst_end->audio_name);
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src_codec->audio_name, dst_codec->audio_name);
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return -EINVAL;
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}
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if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
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if (src_codec->rate && dst_codec->rate && src_codec->rate != dst_codec->rate) {
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LOGP(DMGCP, LOGL_ERROR,
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"Cannot transcode: rate conversion (%d -> %d) not supported.\n",
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src_end->rate, dst_end->rate);
|
||||
src_codec->rate, dst_codec->rate);
|
||||
return -EINVAL;
|
||||
}
|
||||
|
||||
|
@ -269,8 +271,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
|
|||
"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);
|
||||
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;
|
||||
}
|
||||
|
@ -281,16 +283,19 @@ void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
|
|||
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;
|
||||
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 = endp->net_end.payload_type;
|
||||
*payload_type = net_codec->payload_type;
|
||||
*audio_name = net_codec->audio_name;
|
||||
*fmtp_extra = endp->net_end.fmtp_extra;
|
||||
*audio_name = endp->net_end.audio_name;
|
||||
}
|
||||
|
||||
static int decode_audio(struct mgcp_process_rtp_state *state,
|
||||
|
|
|
@ -147,6 +147,8 @@ static int config_write_mgcp(struct vty *vty)
|
|||
static void dump_rtp_end(const char *end_name, struct vty *vty,
|
||||
struct mgcp_rtp_state *state, struct mgcp_rtp_end *end)
|
||||
{
|
||||
struct mgcp_rtp_codec *codec = &end->codec;
|
||||
|
||||
vty_out(vty,
|
||||
" %s%s"
|
||||
" Timestamp Errs: %d->%d%s"
|
||||
|
@ -160,10 +162,10 @@ static void dump_rtp_end(const char *end_name, struct vty *vty,
|
|||
state->in_stream.err_ts_counter,
|
||||
state->out_stream.err_ts_counter, VTY_NEWLINE,
|
||||
end->dropped_packets, VTY_NEWLINE,
|
||||
end->payload_type, end->rate, end->channels, VTY_NEWLINE,
|
||||
end->frame_duration_num, end->frame_duration_den, VTY_NEWLINE,
|
||||
codec->payload_type, codec->rate, codec->channels, VTY_NEWLINE,
|
||||
codec->frame_duration_num, codec->frame_duration_den, VTY_NEWLINE,
|
||||
end->frames_per_packet, end->packet_duration_ms, VTY_NEWLINE,
|
||||
end->fmtp_extra, end->audio_name, end->subtype_name, VTY_NEWLINE,
|
||||
end->fmtp_extra, codec->audio_name, codec->subtype_name, VTY_NEWLINE,
|
||||
end->output_enabled, end->force_output_ptime, VTY_NEWLINE);
|
||||
}
|
||||
|
||||
|
|
|
@ -552,7 +552,7 @@ static int bsc_mgcp_policy_cb(struct mgcp_trunk_config *tcfg, int endpoint, int
|
|||
bsc_msg = bsc_mgcp_rewrite((char *) nat->mgcp_msg, nat->mgcp_length,
|
||||
sccp->bsc_endp, nat->mgcp_cfg->source_addr,
|
||||
mgcp_endp->bts_end.local_port, osmux_cid,
|
||||
&mgcp_endp->net_end.payload_type);
|
||||
&mgcp_endp->net_end.codec.payload_type);
|
||||
if (!bsc_msg) {
|
||||
LOGP(DMGCP, LOGL_ERROR, "Failed to patch the msg.\n");
|
||||
return MGCP_POLICY_CONT;
|
||||
|
@ -746,7 +746,7 @@ void bsc_mgcp_forward(struct bsc_connection *bsc, struct msgb *msg)
|
|||
output = bsc_mgcp_rewrite((char * ) msg->l2h, msgb_l2len(msg), -1,
|
||||
bsc->nat->mgcp_cfg->source_addr,
|
||||
endp->net_end.local_port, -1,
|
||||
&endp->bts_end.payload_type);
|
||||
&endp->bts_end.codec.payload_type);
|
||||
if (!output) {
|
||||
LOGP(DMGCP, LOGL_ERROR, "Failed to rewrite MGCP msg.\n");
|
||||
return;
|
||||
|
|
|
@ -410,7 +410,7 @@ static void test_messages(void)
|
|||
/* reset endpoints */
|
||||
for (i = 0; i < cfg->trunk.number_endpoints; i++) {
|
||||
endp = &cfg->trunk.endpoints[i];
|
||||
endp->net_end.payload_type = PTYPE_NONE;
|
||||
endp->net_end.codec.payload_type = PTYPE_NONE;
|
||||
endp->net_end.packet_duration_ms = -1;
|
||||
|
||||
OSMO_ASSERT(endp->conn_mode == MGCP_CONN_NONE);
|
||||
|
@ -498,18 +498,18 @@ static void test_messages(void)
|
|||
fprintf(stderr, "endpoint %d: "
|
||||
"payload type BTS %d (exp %d), NET %d (exp %d)\n",
|
||||
last_endpoint,
|
||||
endp->bts_end.payload_type, t->exp_bts_ptype,
|
||||
endp->net_end.payload_type, t->exp_net_ptype);
|
||||
endp->bts_end.codec.payload_type, t->exp_bts_ptype,
|
||||
endp->net_end.codec.payload_type, t->exp_net_ptype);
|
||||
|
||||
if (t->exp_bts_ptype != PTYPE_IGNORE)
|
||||
OSMO_ASSERT(endp->bts_end.payload_type ==
|
||||
OSMO_ASSERT(endp->bts_end.codec.payload_type ==
|
||||
t->exp_bts_ptype);
|
||||
if (t->exp_net_ptype != PTYPE_IGNORE)
|
||||
OSMO_ASSERT(endp->net_end.payload_type ==
|
||||
OSMO_ASSERT(endp->net_end.codec.payload_type ==
|
||||
t->exp_net_ptype);
|
||||
|
||||
/* Reset them again for next test */
|
||||
endp->net_end.payload_type = PTYPE_NONE;
|
||||
endp->net_end.codec.payload_type = PTYPE_NONE;
|
||||
}
|
||||
}
|
||||
|
||||
|
@ -830,7 +830,7 @@ static void test_packet_error_detection(int patch_ssrc, int patch_ts)
|
|||
|
||||
mgcp_initialize_endp(&endp);
|
||||
|
||||
rtp->payload_type = 98;
|
||||
rtp->codec.payload_type = 98;
|
||||
|
||||
for (i = 0; i < ARRAY_SIZE(test_rtp_packets1); ++i) {
|
||||
struct rtp_packet_info *info = test_rtp_packets1 + i;
|
||||
|
|
|
@ -183,14 +183,14 @@ static int given_configured_endpoint(int in_samples, int out_samples,
|
|||
mgcp_initialize_endp(endp);
|
||||
|
||||
dst_end = &endp->bts_end;
|
||||
dst_end->payload_type = audio_name_to_type(dstfmt);
|
||||
dst_end->codec.payload_type = audio_name_to_type(dstfmt);
|
||||
|
||||
src_end = &endp->net_end;
|
||||
src_end->payload_type = audio_name_to_type(srcfmt);
|
||||
src_end->codec.payload_type = audio_name_to_type(srcfmt);
|
||||
|
||||
if (out_samples) {
|
||||
dst_end->frame_duration_den = dst_end->rate;
|
||||
dst_end->frame_duration_num = out_samples;
|
||||
dst_end->codec.frame_duration_den = dst_end->codec.rate;
|
||||
dst_end->codec.frame_duration_num = out_samples;
|
||||
dst_end->frames_per_packet = 1;
|
||||
dst_end->force_output_ptime = 1;
|
||||
}
|
||||
|
@ -463,7 +463,7 @@ static int test_repacking(int in_samples, int out_samples, int no_transcode)
|
|||
out_size = mgcp_transcoding_get_frame_size(state, -1, 1);
|
||||
OSMO_ASSERT(sizeof(buf) >= out_size + 12);
|
||||
|
||||
buf[1] = endp->net_end.payload_type;
|
||||
buf[1] = endp->net_end.codec.payload_type;
|
||||
*(uint16_t*)(buf+2) = htons(1);
|
||||
*(uint32_t*)(buf+4) = htonl(0);
|
||||
*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
|
||||
|
|
Loading…
Reference in New Issue