mirror of
https://github.com/outbackdingo/UltraGrid.git
synced 2026-03-20 23:40:18 +00:00
audio_tx_send: send indiviual channel with a func
Send channels sequentially with a new function to simplify the code. Also removed the the cumbersome multiplier packet interleaving - not sure if it worked as expected (having sliding n-packet interleave window for n-multiplied stream). Now we simply send all packets of the frame as without multiplication and then repeated with M-bit set to the latest packet. It is much easier and it is supposed to work approximately the same or even better.
This commit is contained in:
Martin Pulec
220
src/transmit.cpp
220
src/transmit.cpp
@@ -282,6 +282,7 @@ static bool set_fec(struct tx *tx, const char *fec_const)
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snprintf(msg->fec_cfg, sizeof(msg->fec_cfg), "flush");
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tx->mult_count = 1; // default
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if (strcasecmp(fec, "none") == 0) {
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tx->fec_scheme = FEC_NONE;
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} else if(strcasecmp(fec, "mult") == 0) {
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@@ -746,23 +747,22 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
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free(rtp_headers);
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}
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static void audio_tx_send_chan(struct tx *tx, struct rtp *rtp_session,
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uint32_t timestamp, const audio_frame2 *buffer,
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int channel, bool send_m);
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/*
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* This multiplication scheme relies upon the fact, that our RTP/pbuf implementation is
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* not sensitive to packet duplication. Otherwise, we can get into serious problems.
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*/
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void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 * buffer)
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void
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audio_tx_send(struct tx *tx, struct rtp *rtp_session,
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const audio_frame2 *buffer)
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{
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if (!rtp_has_receiver(rtp_session)) {
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return;
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}
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int pt = fec_pt_from_fec_type(TX_MEDIA_AUDIO, buffer->get_fec_params(0).type, tx->encryption); /* PT set for audio in our packet format */
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unsigned m = 0u;
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const char *chan_data;
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// see definition in rtp_callback.h
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uint32_t rtp_hdr[100];
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int mult_first_sent = 0;
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fec_check_messages(tx);
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const uint32_t timestamp =
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@@ -770,109 +770,113 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
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? get_local_mediatime_offset() + buffer->get_timestamp()
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: get_local_mediatime();
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for (int channel = 0; channel < buffer->get_channel_count(); ++channel)
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{
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int rtp_hdr_len = 0;
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int hdrs_len = (rtp_is_ipv6(rtp_session) ? 40 : 20) + 8 + 12; // MTU - IP hdr - UDP hdr - RTP hdr - payload_hdr
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unsigned int fec_symbol_size = buffer->get_fec_params(channel).symbol_size;
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chan_data = buffer->get_data(channel);
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unsigned pos = 0u;
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array <int, FEC_MAX_MULT> mult_pos{};
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int mult_index = 0;
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if (buffer->get_fec_params(0).type == FEC_NONE) {
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hdrs_len += (sizeof(audio_payload_hdr_t));
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rtp_hdr_len = sizeof(audio_payload_hdr_t);
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format_audio_header(buffer, channel, tx->buffer, rtp_hdr);
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} else {
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hdrs_len += (sizeof(fec_payload_hdr_t));
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rtp_hdr_len = sizeof(fec_payload_hdr_t);
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uint32_t tmp = channel << 22;
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tmp |= 0x3fffff & tx->buffer;
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// see definition in rtp_callback.h
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rtp_hdr[0] = htonl(tmp);
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rtp_hdr[2] = htonl(buffer->get_data_len(channel));
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rtp_hdr[3] = htonl(
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buffer->get_fec_params(channel).k << 19 |
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buffer->get_fec_params(channel).m << 6 |
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buffer->get_fec_params(channel).c);
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rtp_hdr[4] = htonl(buffer->get_fec_params(channel).seed);
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for (int iter = 0; iter < tx->mult_count; ++iter) {
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for (int chan = 0; chan < buffer->get_channel_count(); ++chan) {
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bool send_m = iter == tx->mult_count - 1 &&
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chan == buffer->get_channel_count() - 1;
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audio_tx_send_chan(tx, rtp_session, timestamp, buffer,
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chan, send_m);
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}
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if (tx->encryption) {
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hdrs_len += sizeof(crypto_payload_hdr_t) + tx->enc_funcs->get_overhead(tx->encryption);
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rtp_hdr[rtp_hdr_len / sizeof(uint32_t)] = htonl(DEFAULT_CIPHER_MODE << 24);
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rtp_hdr_len += sizeof(crypto_payload_hdr_t);
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}
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if (buffer->get_fec_params(0).type != FEC_NONE) {
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check_symbol_size(fec_symbol_size, tx->mtu - hdrs_len);
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}
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do {
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if(tx->fec_scheme == FEC_MULT) {
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pos = mult_pos[mult_index];
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}
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const char *data = chan_data + pos;
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int data_len = tx->mtu - hdrs_len;
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if(pos + data_len >= (unsigned int) buffer->get_data_len(channel)) {
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data_len = buffer->get_data_len(channel) - pos;
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if(channel == buffer->get_channel_count() - 1)
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m = 1;
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}
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rtp_hdr[1] = htonl(pos);
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pos += data_len;
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if(data_len) { /* check needed for FEC_MULT */
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char encrypted_data[data_len + MAX_CRYPTO_EXCEED];
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if(tx->encryption) {
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data_len = tx->enc_funcs->encrypt(tx->encryption,
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const_cast<char *>(data), data_len,
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(char *) rtp_hdr, rtp_hdr_len - sizeof(crypto_payload_hdr_t),
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encrypted_data);
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data = encrypted_data;
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}
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if (control_stats_enabled(tx->control)) {
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auto current_time_ms = time_since_epoch_in_ms();
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if(current_time_ms - tx->last_stat_report >= CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_MS){
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std::ostringstream oss;
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oss << "tx_send " << std::hex << rtp_my_ssrc(rtp_session) << std::dec << " audio " << tx->sent_since_report;
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control_report_stats(tx->control, oss.str());
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tx->last_stat_report = current_time_ms;
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tx->sent_since_report = 0;
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}
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tx->sent_since_report += data_len + rtp_hdr_len;
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}
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rtp_send_data_hdr(rtp_session, timestamp, pt, m, 0, /* contributing sources */
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0, /* contributing sources length */
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(char *) rtp_hdr, rtp_hdr_len,
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const_cast<char *>(data), data_len,
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0, 0, 0);
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}
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if(tx->fec_scheme == FEC_MULT) {
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mult_pos[mult_index] = pos;
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mult_first_sent ++;
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if(mult_index != 0 || mult_first_sent >= (tx->mult_count - 1))
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mult_index = (mult_index + 1) % tx->mult_count;
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}
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/* when trippling, we need all streams goes to end */
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if(tx->fec_scheme == FEC_MULT) {
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pos = mult_pos[tx->mult_count - 1];
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}
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} while (pos < buffer->get_data_len(channel));
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}
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tx->buffer ++;
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tx->buffer++;
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}
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static void
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audio_tx_send_chan(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
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const audio_frame2 *buffer, int channel, bool send_m)
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{
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int pt = fec_pt_from_fec_type(
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TX_MEDIA_AUDIO, buffer->get_fec_params(0).type,
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tx->encryption); /* PT set for audio in our packet format */
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unsigned m = 0U;
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// see definition in rtp_callback.h
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uint32_t rtp_hdr[100];
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int rtp_hdr_len = 0;
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int hdrs_len = (rtp_is_ipv6(rtp_session) ? 40 : 20) + 8 +
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12; // MTU - IP hdr - UDP hdr - RTP hdr - payload_hdr
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unsigned int fec_symbol_size =
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buffer->get_fec_params(channel).symbol_size;
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const char *chan_data = buffer->get_data(channel);
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unsigned pos = 0U;
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if (buffer->get_fec_params(0).type == FEC_NONE) {
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hdrs_len += (sizeof(audio_payload_hdr_t));
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rtp_hdr_len = sizeof(audio_payload_hdr_t);
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format_audio_header(buffer, channel, tx->buffer, rtp_hdr);
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} else {
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hdrs_len += (sizeof(fec_payload_hdr_t));
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rtp_hdr_len = sizeof(fec_payload_hdr_t);
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uint32_t tmp = channel << 22;
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tmp |= 0x3fffff & tx->buffer;
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// see definition in rtp_callback.h
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rtp_hdr[0] = htonl(tmp);
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rtp_hdr[2] = htonl(buffer->get_data_len(channel));
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rtp_hdr[3] = htonl(buffer->get_fec_params(channel).k << 19 |
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buffer->get_fec_params(channel).m << 6 |
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buffer->get_fec_params(channel).c);
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rtp_hdr[4] = htonl(buffer->get_fec_params(channel).seed);
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}
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if (tx->encryption) {
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hdrs_len += sizeof(crypto_payload_hdr_t) +
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tx->enc_funcs->get_overhead(tx->encryption);
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rtp_hdr[rtp_hdr_len / sizeof(uint32_t)] =
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htonl(DEFAULT_CIPHER_MODE << 24);
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rtp_hdr_len += sizeof(crypto_payload_hdr_t);
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}
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if (buffer->get_fec_params(0).type != FEC_NONE) {
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check_symbol_size(fec_symbol_size, tx->mtu - hdrs_len);
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}
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do {
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const char *data = chan_data + pos;
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int data_len = tx->mtu - hdrs_len;
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if (pos + data_len >=
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(unsigned int) buffer->get_data_len(channel)) {
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data_len = buffer->get_data_len(channel) - pos;
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if (send_m) {
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m = 1;
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}
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}
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rtp_hdr[1] = htonl(pos);
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pos += data_len;
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char encrypted_data[data_len + MAX_CRYPTO_EXCEED];
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if (tx->encryption) {
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data_len = tx->enc_funcs->encrypt(
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tx->encryption, const_cast<char *>(data), data_len,
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(char *) rtp_hdr,
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rtp_hdr_len - sizeof(crypto_payload_hdr_t),
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encrypted_data);
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data = encrypted_data;
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}
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if (control_stats_enabled(tx->control)) {
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auto current_time_ms = time_since_epoch_in_ms();
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if (current_time_ms - tx->last_stat_report >=
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CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_MS) {
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std::ostringstream oss;
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oss << "tx_send " << std::hex
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<< rtp_my_ssrc(rtp_session) << std::dec
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<< " audio " << tx->sent_since_report;
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control_report_stats(tx->control, oss.str());
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tx->last_stat_report = current_time_ms;
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tx->sent_since_report = 0;
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}
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tx->sent_since_report += data_len + rtp_hdr_len;
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}
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rtp_send_data_hdr(rtp_session, timestamp, pt, m,
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0, /* contributing sources */
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0, /* contributing sources length */
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(char *) rtp_hdr, rtp_hdr_len,
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const_cast<char *>(data), data_len, 0, 0, 0);
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} while (pos < buffer->get_data_len(channel));
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}
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/**
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