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transmit: use array for mult_pos

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+ declare some vars when initialized (audio_tx_send())
This commit is contained in:
Martin Pulec
2022-03-07 09:57:57 +01:00
parent a778460fba
commit 600b5c92fa
1 changed files with 10 additions and 28 deletions
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38
src/transmit.cpp
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@@ -84,6 +84,7 @@
#include "video_codec.h"
#include <algorithm>
#include <array>
#include <iostream>
#include <vector>
@@ -107,6 +108,7 @@
#define DEFAULT_CIPHER_MODE MODE_AES128_CFB
using std::array;
using std::vector;
static void tx_update(struct tx *tx, struct video_frame *frame, int substream);
@@ -580,7 +582,7 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
#endif
long delta, overslept = 0;
uint32_t tmp;
int mult_pos[FEC_MAX_MULT];
array <int, FEC_MAX_MULT> mult_pos{};
int mult_index = 0;
int mult_first_sent = 0;
@@ -593,14 +595,6 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
perf_record(UVP_SEND, ts);
if(tx->fec_scheme == FEC_MULT) {
int i;
for (i = 0; i < tx->mult_count; ++i) {
mult_pos[i] = 0;
}
mult_index = 0;
}
m = 0;
pos = 0;
@@ -772,12 +766,9 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
return;
}
int pt; /* PT set for audio in our packet format */
unsigned int pos = 0u,
m = 0u;
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 */
unsigned m = 0u;
const char *chan_data;
int data_len;
const char *data;
// see definition in rtp_callback.h
uint32_t rtp_hdr[100];
uint32_t timestamp;
@@ -789,8 +780,6 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
LARGE_INTEGER start, stop, freq;
#endif
long delta;
int mult_pos[FEC_MAX_MULT];
int mult_index = 0;
int mult_first_sent = 0;
fec_check_messages(tx);
@@ -798,8 +787,6 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
timestamp = get_local_mediatime();
perf_record(UVP_SEND, timestamp);
pt = fec_pt_from_fec_type(TX_MEDIA_AUDIO, buffer->get_fec_params(0).type, tx->encryption);
for (int channel = 0; channel < buffer->get_channel_count(); ++channel)
{
int rtp_hdr_len = 0;
@@ -807,15 +794,10 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
unsigned int fec_symbol_size = buffer->get_fec_params(channel).symbol_size;
chan_data = buffer->get_data(channel);
pos = 0u;
unsigned pos = 0u;
if(tx->fec_scheme == FEC_MULT) {
int i;
for (i = 0; i < tx->mult_count; ++i) {
mult_pos[i] = 0;
}
mult_index = 0;
}
array <int, FEC_MAX_MULT> mult_pos{};
int mult_index = 0;
if (buffer->get_fec_params(0).type == FEC_NONE) {
hdrs_len += (sizeof(audio_payload_hdr_t));
@@ -881,8 +863,8 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
pos = mult_pos[mult_index];
}
data = chan_data + pos;
data_len = tx->mtu - hdrs_len;
const char *data = chan_data + pos;
int data_len = tx->mtu - hdrs_len;
if(pos + data_len >= (unsigned int) buffer->get_data_len(channel)) {
data_len = buffer->get_data_len(channel) - pos;
if(channel == buffer->get_channel_count() - 1)