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audio_tx_send_standard: validation to separate fn

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This commit is contained in:
Martin Pulec
2024-02-05 10:04:43 +01:00
parent 731a48e94f
commit c64cc2df9c
1 changed files with 20 additions and 13 deletions
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33
src/transmit.cpp
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@@ -874,6 +874,22 @@ audio_tx_send_chan(struct tx *tx, struct rtp *rtp_session, uint32_t timestamp,
report_stats(tx, rtp_session, data_sent);
}
static bool
validate_std_audio(const audio_frame2 * buffer, int payload_size, int data_len)
{
if (buffer->get_codec() == AC_OPUS &&
buffer->get_channel_count() > 1) { // we cannot interleave Opus here
MSG(ERROR, "Opus can currently have only 1 channel in "
"RFC-compliant mode! Discarding...\n");
return false;
}
if (buffer->get_codec() == AC_OPUS && payload_size < data_len) {
MSG(ERROR, "Opus frame larger than packet! Discarding...\n");
return false;
}
return true;
}
/**
* audio_tx_send_standard - Send interleaved channels from the audio_frame2,
* as the mulaw and A-law standards (dynamic or std PT).
@@ -886,13 +902,6 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
buffer->get_codec() == AC_MP3 ||
buffer->get_codec() == AC_OPUS);
if (buffer->get_codec() == AC_OPUS &&
buffer->get_channel_count() > 1) { // we cannot interleave Opus here
MSG(ERROR, "Opus can currently have only 1 channel in "
"RFC-compliant mode! Discarding...\n");
return;
}
uint32_t ts;
static uint32_t ts_prev = 0;
@@ -916,12 +925,6 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
int data_len = buffer->get_data_len(0) * buffer->get_channel_count(); /* Number of samples to send */
int payload_size = tx->mtu - 40 - 8 - 12; /* Max size of an RTP payload field (minus IPv6, UDP and RTP header lengths) */
if (buffer->get_codec() == AC_OPUS) { // OPUS needs to fit one package
if (payload_size < data_len) {
log_msg(LOG_LEVEL_ERROR, "Transmit: Opus frame larger than packet! Discarding...\n");
return;
}
}
if (pt == PT_ITU_T_G711_PCMU ||
pt == PT_ITU_T_G711_PCMA) { // we may split the data into more
// packets, compute chunk size
@@ -932,6 +935,10 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
payload_size -= sizeof(mpa_hdr_t);
}
if (!validate_std_audio(buffer, payload_size, data_len)) {
return;
}
int pos = 0;
do {
int pkt_len = std::min(payload_size, data_len - pos);