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RTP: fix sending of RFC-compilant OPUS stream

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RTP timestamp was incorrectly computed from data_len/bps which confused VLC.

+ some cleanup
This commit is contained in:
Martin Pulec
2018-08-20 18:21:19 +02:00
parent 86874cb8a9
commit 315749bbe5
8 changed files with 79 additions and 79 deletions
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96
src/transmit.cpp
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@@ -62,13 +62,14 @@
#include "config_win32.h"
#endif // HAVE_CONFIG_H
#include "audio/audio.h"
#include "audio/codec.h"
#include "audio/utils.h"
#include "crypto/random.h"
#include "debug.h"
#include "host.h"
#include "lib_common.h"
#include "perf.h"
#include "audio/audio.h"
#include "audio/codec.h"
#include "crypto/openssl_encrypt.h"
#include "module.h"
#include "rtp/fec.h"
@@ -102,11 +103,6 @@
#define DEFAULT_CIPHER_MODE MODE_AES128_CFB
// Mulaw audio memory reservation
#define BUFFER_MTU_SIZE 1500
static char *data_buffer_mulaw;
static int buffer_mulaw_init = 0;
static void tx_update(struct tx *tx, struct video_frame *frame, int substream);
static void tx_done(struct module *tx);
static uint32_t format_interl_fps_hdr_row(enum interlacing_t interlacing, double input_fps);
@@ -147,16 +143,9 @@ struct tx {
long long int bitrate;
struct rtpenc_h264_state *rtpenc_h264_state;
char tmp_packet[RTP_MAX_MTU];
};
// Mulaw audio memory reservation
static void init_tx_mulaw_buffer() {
if (!buffer_mulaw_init) {
data_buffer_mulaw = (char *) malloc(BUFFER_MTU_SIZE*20);
buffer_mulaw_init = 1;
}
}
static void tx_update(struct tx *tx, struct video_frame *frame, int substream)
{
if(!frame) {
@@ -876,7 +865,7 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 *
tx->buffer ++;
}
/*
/**
* audio_tx_send_standard - Send interleaved channels from the audio_frame2,
* as the mulaw and A-law standards (dynamic or std PT).
*/
@@ -908,47 +897,50 @@ void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
assert(buffer->get_data_len(0) == buffer->get_data_len(i));
int data_len = buffer->get_data_len(0) * buffer->get_channel_count(); /* Number of samples to send */
int payload_size = tx->mtu - 40; /* Max size of an RTP payload field */
int payload_size = tx->mtu - 40 - 8 - 12; /* Max size of an RTP payload field (minus IPv6, UDP and RTP header lengths) */
init_tx_mulaw_buffer();
char *curr_sample = data_buffer_mulaw;
int ch, pos = 0, count = 0, pointerToSend = 0;
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;
}
} else { // we may split the data into more packets, compute chunk size
int frame_size = buffer->get_channel_count() * buffer->get_bps();
payload_size = payload_size / frame_size * frame_size; // align to frame size
}
int pos = 0;
do {
for (ch = 0; ch < buffer->get_channel_count(); ch++) {
memcpy(curr_sample, buffer->get_data(ch) + pos,
buffer->get_bps() * sizeof(char));
curr_sample += buffer->get_bps() * sizeof(char);
count += buffer->get_bps() * sizeof(char);
}
pos += buffer->get_bps() * sizeof(char);
int pkt_len = std::min(payload_size, data_len - pos);
if ((pos * buffer->get_channel_count()) % payload_size == 0) {
// Update first sample timestamp
ts = get_std_audio_local_mediatime((double)payload_size / (double)buffer->get_channel_count());
gettimeofday(&curr_time, NULL);
rtp_send_ctrl(rtp_session, ts_prev, 0, curr_time); //send RTCP SR
ts_prev = ts;
// Send the packet
rtp_send_data(rtp_session, ts, pt, 0, 0, /* contributing sources */
0, /* contributing sources length */
data_buffer_mulaw + pointerToSend, payload_size, 0, 0, 0);
pointerToSend += payload_size;
}
} while (count < data_len);
// interleave
if (buffer->get_codec() == AC_OPUS) {
assert(buffer->get_channel_count() == 1); // we cannot interleave OPUS here
memcpy(tx->tmp_packet, buffer->get_data(0), pkt_len);
} else {
for (int ch = 0; ch < buffer->get_channel_count(); ch++) {
remux_channel(tx->tmp_packet, buffer->get_data(ch) + pos / buffer->get_channel_count(), buffer->get_bps(), pkt_len / buffer->get_channel_count(), 1, buffer->get_channel_count(), 0, ch);
}
}
if ((pos * buffer->get_channel_count()) % payload_size != 0) {
// Update first sample timestamp
ts = get_std_audio_local_mediatime((double)((pos * buffer->get_channel_count()) % payload_size) / (double)buffer->get_channel_count());
gettimeofday(&curr_time, NULL);
rtp_send_ctrl(rtp_session, ts_prev, 0, curr_time); //send RTCP SR
ts_prev = ts;
// Send the packet
rtp_send_data(rtp_session, ts, pt, 0, 0, /* contributing sources */
0, /* contributing sources length */
data_buffer_mulaw + pointerToSend,
(pos * buffer->get_channel_count()) % payload_size, 0, 0, 0);
}
// Update first sample timestamp
if (buffer->get_codec() == AC_OPUS) {
/* OPUS packet will be the whole contained in one packet
* according to RFC 7587. For PCMA/PCMU there may be more
* packets so we cannot use the whole frame duration. */
ts = get_std_audio_local_mediatime(buffer->get_duration(), 48000);
} else {
ts = get_std_audio_local_mediatime((double) pkt_len / (double) buffer->get_channel_count() / (double) buffer->get_sample_rate(), buffer->get_sample_rate());
}
gettimeofday(&curr_time, NULL);
rtp_send_ctrl(rtp_session, ts_prev, 0, curr_time); //send RTCP SR
ts_prev = ts;
// Send the packet
rtp_send_data(rtp_session, ts, pt, 0, 0, /* contributing sources */
0, /* contributing sources length */
tx->tmp_packet, pkt_len, 0, 0, 0);
pos += pkt_len;
} while (pos < data_len);
}
/**