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transmit: moved stat reporting after send loop

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it simplifies the code and facilites eventual refactor
This commit is contained in:
Martin Pulec
2023-09-20 09:17:27 +02:00
parent 70d1c5186d
commit 2c8dd6743e
1 changed files with 34 additions and 28 deletions
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62
src/transmit.cpp
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@@ -86,6 +86,7 @@
#include <algorithm>
#include <array>
#include <iostream>
#include <sstream>
#include <vector>
#define MOD_NAME "[transmit] "
@@ -548,6 +549,32 @@ static vector<int> get_packet_sizes(struct video_frame *frame, int substream, in
return ret;
}
static void
report_stats(struct tx *tx, struct rtp *rtp_session, long data_sent)
{
if (!control_stats_enabled(tx->control)) {
return;
}
tx->sent_since_report += data_sent;
const time_ns_t current_time_ns = get_time_in_ns();
if (current_time_ns - tx->last_stat_report <
CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS) {
return;
}
const char *media =
tx->media_type == TX_MEDIA_VIDEO ? "video" : "audio";
std::ostringstream oss;
oss << "tx_send " << std::hex << rtp_my_ssrc(rtp_session) << std::dec
<< " " << media << " " << tx->sent_since_report;
control_report_stats(tx->control, oss.str());
tx->last_stat_report = current_time_ns;
tx->sent_since_report = 0;
}
/**
* Returns inter-packet interval in nanoseconds.
*/
@@ -703,20 +730,6 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
data = encrypted_data;
}
if (control_stats_enabled(tx->control)) {
const time_ns_t current_time_ns =
get_time_in_ns();
if (current_time_ns - tx->last_stat_report >=
CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS) {
std::ostringstream oss;
oss << "tx_send " << std::hex << rtp_my_ssrc(rtp_session) << std::dec << " video " << tx->sent_since_report;
control_report_stats(tx->control, oss.str());
tx->last_stat_report = current_time_ns;
tx->sent_since_report = 0;
}
tx->sent_since_report += data_len + rtp_hdr_len;
}
rtp_send_data_hdr(rtp_session, ts, pt, m, 0, 0,
(char *) rtp_hdr_packet, rtp_hdr_len,
data, data_len, 0, 0, 0);
@@ -744,6 +757,9 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
}
} while (pos < tile->data_len || mult_index != 0); // when multiplying, we need all streams go to the end
const long data_sent = tile->data_len + rtp_hdr_len * packet_count;
report_stats(tx, rtp_session, data_sent);
if (!tx->encryption) {
rtp_async_wait(rtp_session);
}
@@ -857,6 +873,7 @@ audio_tx_send_pkt(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
check_symbol_size(fec_symbol_size, tx->mtu - hdrs_len);
}
long data_sent = 0;
do {
const char *data = chan_data + pos;
unsigned data_len = tx->mtu - hdrs_len;
@@ -879,20 +896,7 @@ audio_tx_send_pkt(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
data = encrypted_data;
}
if (control_stats_enabled(tx->control)) {
const time_ns_t current_time_ns = get_time_in_ns();
if (current_time_ns - tx->last_stat_report >=
CONTROL_PORT_BANDWIDTH_REPORT_INTERVAL_NS) {
std::ostringstream oss;
oss << "tx_send " << std::hex
<< rtp_my_ssrc(rtp_session) << std::dec
<< " audio " << tx->sent_since_report;
control_report_stats(tx->control, oss.str());
tx->last_stat_report = current_time_ns;
tx->sent_since_report = 0;
}
tx->sent_since_report += data_len + rtp_hdr_len;
}
data_sent += data_len + rtp_hdr_len;
rtp_send_data_hdr(rtp_session, timestamp, pt, m,
0, /* contributing sources */
@@ -901,6 +905,8 @@ audio_tx_send_pkt(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
const_cast<char *>(data), data_len, 0, 0, 0);
} while (pos < len);
report_stats(tx, rtp_session, data_sent);
}
/**