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83e2d7087aab7fa8308fb1d2ecb9a944668d496e
UltraGrid/src/audio/audio.cpp
Martin Pulec 83e2d7087a Fixed Speex references to SpeexDSP 
To be clear that we are actually using SpeexDSP.
2021-03-03 15:01:54 +01:00

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/*
* FILE: audio/audio.cpp
* AUTHORS: Martin Benes <martinbenesh@gmail.com>
* Lukas Hejtmanek <xhejtman@ics.muni.cz>
* Petr Holub <hopet@ics.muni.cz>
* Milos Liska <xliska@fi.muni.cz>
* Jiri Matela <matela@ics.muni.cz>
* Dalibor Matura <255899@mail.muni.cz>
* Martin Pulec <martin.pulec@cesnet.cz>
* Ian Wesley-Smith <iwsmith@cct.lsu.edu>
*
* Copyright (c) 2005-2020 CESNET z.s.p.o.
*
* Redistribution and use in source and binary forms, with or without
* modification, is permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
*
* This product includes software developed by CESNET z.s.p.o.
*
* 4. Neither the name of CESNET nor the names of its contributors may be used
* to endorse or promote products derived from this software without specific
* prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#include "config_unix.h"
#include "config_win32.h"
#endif
#include <cinttypes>
#include <iomanip>
#include <iostream>
#include <sstream>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include "audio/audio.h"
#include "audio/codec.h"
#include "audio/echo.h"
#include "audio/audio_capture.h"
#include "audio/audio_playback.h"
#include "audio/capture/sdi.h"
#include "audio/playback/sdi.h"
#include "audio/jack.h"
#include "audio/utils.h"
#include "compat/platform_semaphore.h"
#include "compat/platform_semaphore.h"
#include "debug.h"
#include "../export.h" // not audio/export.h
#include "host.h"
#include "module.h"
#include "perf.h"
#include "rang.hpp"
#include "rtp/audio_decoders.h"
#include "rtp/rtp.h"
#include "rtp/rtp_callback.h"
#include "rtp/pbuf.h"
#include "tv.h"
#include "transmit.h"
#include "pdb.h"
#include "utils/thread.h"
#include "utils/worker.h"
using namespace std;
using rang::fg;
using rang::style;
enum audio_transport_device {
NET_NATIVE,
NET_JACK,
NET_STANDARD
};
#define DEFAULT_AUDIO_RECV_BUF_SIZE (256 * 1024)
struct audio_network_parameters {
char *addr = nullptr;
int recv_port = 0;
int send_port = 0;
struct pdb *participants = 0;
int force_ip_version = 0;
char *mcast_if = nullptr;
};
struct state_audio {
state_audio(struct module *parent) {
module_init_default(&mod);
mod.priv_data = this;
mod.cls = MODULE_CLASS_AUDIO;
module_register(&mod, parent);
module_init_default(&audio_receiver_module);
audio_receiver_module.cls = MODULE_CLASS_RECEIVER;
audio_receiver_module.priv_data = this;
module_register(&audio_receiver_module, &mod);
module_init_default(&audio_sender_module);
audio_sender_module.cls = MODULE_CLASS_SENDER;
audio_sender_module.priv_data = this;
module_register(&audio_sender_module, &mod);
gettimeofday(&t0, NULL);
}
~state_audio() {
module_done(&audio_receiver_module);
module_done(&audio_sender_module);
module_done(&mod);
}
struct module mod;
struct state_audio_capture *audio_capture_device = nullptr;
struct state_audio_playback *audio_playback_device = nullptr;
struct module audio_receiver_module;
struct module audio_sender_module;
struct audio_codec_state *audio_coder = nullptr;
struct audio_network_parameters audio_network_parameters{};
struct rtp *audio_network_device = nullptr;
struct pdb *audio_participants = nullptr;
std::string proto_cfg; // audio network protocol options
void *jack_connection = nullptr;
enum audio_transport_device sender = NET_NATIVE;
enum audio_transport_device receiver = NET_NATIVE;
std::chrono::steady_clock::time_point start_time;
struct timeval t0; // for statistics
audio_frame2 captured;
struct tx *tx_session = nullptr;
pthread_t audio_sender_thread_id{},
audio_receiver_thread_id{};
bool audio_sender_thread_started = false,
audio_receiver_thread_started = false;
char *audio_channel_map = nullptr;
const char *audio_scale = nullptr;
echo_cancellation_t *echo_state = nullptr;
struct exporter *exporter = nullptr;
int resample_to = 0;
char *requested_encryption = nullptr;
volatile bool paused = false; // for CoUniverse...
int audio_tx_mode = 0;
double volume = 1.0; // receiver volume scale
bool muted_receiver = false;
bool muted_sender = false;
size_t recv_buf_size = DEFAULT_AUDIO_RECV_BUF_SIZE;
};
/**
* Copies one input channel into n output (interlaced).
*
* Input and output data may overlap.
*/
typedef void (*audio_device_help_t)(void);
static void *audio_sender_thread(void *arg);
static void *audio_receiver_thread(void *arg);
static struct rtp *initialize_audio_network(struct audio_network_parameters *params);
static struct response *audio_receiver_process_message(struct state_audio *s, struct msg_receiver *msg);
static struct response *audio_sender_process_message(struct state_audio *s, struct msg_sender *msg);
static void audio_channel_map_usage(void);
static void audio_scale_usage(void);
static void audio_channel_map_usage(void)
{
printf("\t--audio-channel-map <mapping> mapping of input audio channels\n");
printf("\t to output audio channels comma-separated\n");
printf("\t list of channel mapping\n");
printf("\t eg. 0:0,1:0 - mixes first 2 channels\n");
printf("\t 0:0 - play only first channel\n");
printf("\t 0:0,:1 - sets second channel to\n");
printf("\t a silence, first one is\n");
printf("\t left as is\n");
printf("\t 0:0,0:1 - splits mono into\n");
printf("\t 2 channels\n");
}
static void audio_scale_usage(void)
{
printf("\t--audio-scale [<factor>|<method>]\n");
printf("\t Floating point number that tells\n");
printf("\t a static scaling factor for all\n");
printf("\t output channels.\n");
printf("\t Scaling method can be one from these:\n");
printf("\t mixauto - automatically adjust\n");
printf("\t volume if using channel\n");
printf("\t mixing/remapping\n");
printf("\t (default)\n");
printf("\t auto - automatically adjust volume\n");
printf("\t none - no scaling will be performed\n");
}
/**
* take care that addrs can also be comma-separated list of addresses !
*/
struct state_audio * audio_cfg_init(struct module *parent, const char *addrs, int recv_port, int send_port,
const char *send_cfg, const char *recv_cfg,
const char *proto, const char *proto_cfg,
const char *fec_cfg, const char *encryption,
char *audio_channel_map, const char *audio_scale,
bool echo_cancellation, int force_ip_version, const char *mcast_if,
const char *audio_codec_cfg,
long long int bitrate, volatile int *audio_delay, const std::chrono::steady_clock::time_point *start_time, int mtu, struct exporter *exporter)
{
struct state_audio *s = NULL;
char *tmp, *unused = NULL;
UNUSED(unused);
char *addr;
int resample_to = get_audio_codec_sample_rate(audio_codec_cfg);
assert(send_cfg != NULL);
assert(recv_cfg != NULL);
if(audio_channel_map &&
strcmp("help", audio_channel_map) == 0) {
audio_channel_map_usage();
exit_uv(0);
return NULL;
}
if(audio_scale &&
strcmp("help", audio_scale) == 0) {
audio_scale_usage();
exit_uv(0);
return NULL;
}
s = new state_audio(parent);
s->start_time = *start_time;
s->audio_channel_map = audio_channel_map;
s->audio_scale = audio_scale;
s->audio_sender_thread_started = s->audio_receiver_thread_started = false;
s->resample_to = resample_to;
s->audio_coder = audio_codec_init_cfg(audio_codec_cfg, AUDIO_CODER);
if(!s->audio_coder) {
goto error;
}
s->exporter = exporter;
if(echo_cancellation) {
#ifdef HAVE_SPEEXDSP
//s->echo_state = echo_cancellation_init();
fprintf(stderr, "Echo cancellation is currently broken "
"in UltraGrid.\nPlease write to %s "
"if you wish to use this feature.\n",
PACKAGE_BUGREPORT);
goto error;
#else
fprintf(stderr, "Speex not compiled in. Could not enable echo cancellation.\n");
delete s;
goto error;
#endif /* HAVE_SPEEXDSP */
} else {
s->echo_state = NULL;
}
if(encryption) {
s->requested_encryption = strdup(encryption);
}
assert(addrs != nullptr);
tmp = strdup(addrs);
s->audio_participants = pdb_init(audio_delay);
addr = strtok_r(tmp, ",", &unused);
assert(addr != nullptr);
s->audio_network_parameters.addr = strdup(addr);
s->audio_network_parameters.recv_port = recv_port;
s->audio_network_parameters.send_port = send_port;
s->audio_network_parameters.participants = s->audio_participants;
s->audio_network_parameters.force_ip_version = force_ip_version;
s->audio_network_parameters.mcast_if = mcast_if
? strdup(mcast_if) : NULL;
if ((s->audio_network_device = initialize_audio_network(
&s->audio_network_parameters))
== NULL) {
printf("Unable to open audio network\n");
free(tmp);
goto error;
}
free(tmp);
if (strcmp(send_cfg, "none") != 0) {
char *cfg = NULL;
char *device = strdup(send_cfg);
if(strchr(device, ':')) {
char *delim = strchr(device, ':');
*delim = '\0';
cfg = delim + 1;
}
int ret = audio_capture_init(device, cfg, &s->audio_capture_device);
free(device);
if(ret < 0) {
fprintf(stderr, "Error initializing audio capture.\n");
goto error;
}
if(ret > 0) {
goto error;
}
s->tx_session = tx_init(&s->audio_sender_module, mtu, TX_MEDIA_AUDIO, fec_cfg, encryption, bitrate);
if(!s->tx_session) {
fprintf(stderr, "Unable to initialize audio transmit.\n");
goto error;
}
s->audio_tx_mode |= MODE_SENDER;
} else {
s->audio_capture_device = audio_capture_init_null_device();
}
if (strcmp(recv_cfg, "none") != 0) {
char *cfg = NULL;
char *device = strdup(recv_cfg);
if(strchr(device, ':')) {
char *delim = strchr(device, ':');
*delim = '\0';
cfg = delim + 1;
}
int ret = audio_playback_init(device, cfg, &s->audio_playback_device);
free(device);
if(ret < 0) {
fprintf(stderr, "Error initializing audio playback.\n");
goto error;
}
if(ret > 0) {
goto error;
}
size_t len = sizeof(struct rtp *);
audio_playback_ctl(s->audio_playback_device, AUDIO_PLAYBACK_PUT_NETWORK_DEVICE,
&s->audio_network_device, &len);
s->audio_tx_mode |= MODE_RECEIVER;
} else {
s->audio_playback_device = audio_playback_init_null_device();
}
s->proto_cfg = proto_cfg;
if (strcasecmp(proto, "ultragrid_rtp") == 0) {
s->sender = NET_NATIVE;
s->receiver = NET_NATIVE;
} else if (strcasecmp(proto, "rtsp") == 0 || strcasecmp(proto, "sdp") == 0) {
s->receiver = NET_STANDARD;
s->sender = NET_STANDARD;
} else if (strcasecmp(proto, "JACK") == 0) {
#ifndef HAVE_JACK_TRANS
fprintf(stderr, "[Audio] JACK transport requested, "
"but JACK support isn't compiled.\n");
goto error;
#else
s->sender = NET_JACK;
s->receiver = NET_JACK;
#endif
} else {
log_msg(LOG_LEVEL_ERROR, "Unknow audio protocol: %s\n", proto);
goto error;
}
return s;
error:
if(s->tx_session)
module_done(CAST_MODULE(s->tx_session));
if(s->audio_participants) {
pdb_destroy(&s->audio_participants);
}
audio_codec_done(s->audio_coder);
delete s;
exit_uv(EXIT_FAIL_AUDIO);
return NULL;
}
void audio_start(struct state_audio *s) {
#ifdef HAVE_JACK_TRANS
if (s->sender == NET_JACK || s->receiver == NET_JACK) {
s->jack_connection = jack_start(s->proto_cfg.c_str());
if (s->jack_connection) {
if (!is_jack_sender(s->jack_connection)) {
s->sender = NET_NATIVE;
}
if (!is_jack_receiver(s->jack_connection)) {
s->receiver = NET_NATIVE;
}
} else {
log_msg(LOG_LEVEL_FATAL, "JACK transport initialization failed!\n");
exit_uv(EXIT_FAIL_AUDIO);
}
}
#endif
if (s->audio_tx_mode & MODE_SENDER) {
if (pthread_create
(&s->audio_sender_thread_id, NULL, audio_sender_thread, (void *)s) != 0) {
log_msg(LOG_LEVEL_FATAL, "Error creating audio thread. Quitting\n");
exit_uv(EXIT_FAIL_AUDIO);
} else {
s->audio_sender_thread_started = true;
}
}
if (s->audio_tx_mode & MODE_RECEIVER) {
if (pthread_create
(&s->audio_receiver_thread_id, NULL, audio_receiver_thread, (void *)s) != 0) {
log_msg(LOG_LEVEL_FATAL, "Error creating audio thread. Quitting\n");
exit_uv(EXIT_FAIL_AUDIO);
} else {
s->audio_receiver_thread_started = true;
}
}
}
void audio_join(struct state_audio *s) {
if(s) {
if(s->audio_receiver_thread_started)
pthread_join(s->audio_receiver_thread_id, NULL);
if(s->audio_sender_thread_started)
pthread_join(s->audio_sender_thread_id, NULL);
}
}
void audio_done(struct state_audio *s)
{
if (!s) {
return;
}
audio_playback_done(s->audio_playback_device);
audio_capture_done(s->audio_capture_device);
// process remaining messages
struct message *msg;
while ((msg = check_message(&s->audio_receiver_module))) {
struct response *r = audio_receiver_process_message(s, (struct msg_receiver *) msg);
free_message(msg, r);
}
while ((msg = check_message(&s->audio_sender_module))) {
struct response *r = audio_sender_process_message(s, (struct msg_sender *) msg);
free_message(msg, r);
}
module_done(CAST_MODULE(s->tx_session));
if(s->audio_network_device)
rtp_done(s->audio_network_device);
if(s->audio_participants) {
pdb_destroy(&s->audio_participants);
}
free(s->requested_encryption);
free(s->audio_network_parameters.addr);
free(s->audio_network_parameters.mcast_if);
audio_codec_done(s->audio_coder);
delete s;
}
static struct rtp *initialize_audio_network(struct audio_network_parameters *params)
{
struct rtp *r;
double rtcp_bw = 1024 * 512; // FIXME: something about 5% for rtcp is said in rfc
r = rtp_init_if(params->addr, params->mcast_if, params->recv_port,
params->send_port, 255, rtcp_bw,
FALSE, rtp_recv_callback,
(uint8_t *) params->participants,
params->force_ip_version, false);
if (r != NULL) {
pdb_add(params->participants, rtp_my_ssrc(r));
rtp_set_option(r, RTP_OPT_WEAK_VALIDATION, TRUE);
rtp_set_option(r, RTP_OPT_PROMISC, TRUE);
rtp_set_option(r, RTP_OPT_RECORD_SOURCE, TRUE);
rtp_set_sdes(r, rtp_my_ssrc(r), RTCP_SDES_TOOL,
PACKAGE_STRING, strlen(PACKAGE_VERSION));
rtp_set_recv_buf(r, DEFAULT_AUDIO_RECV_BUF_SIZE);
}
return r;
}
struct audio_decoder {
bool enabled;
struct pbuf_audio_data pbuf_data;
};
static struct response * audio_receiver_process_message(struct state_audio *s, struct msg_receiver *msg)
{
switch (msg->type) {
case RECEIVER_MSG_CHANGE_RX_PORT:
{
assert(s->audio_tx_mode == MODE_RECEIVER); // receiver only
struct rtp *old_audio_network_device = s->audio_network_device;
int old_rx_port = s->audio_network_parameters.recv_port;
s->audio_network_parameters.recv_port = msg->new_rx_port;
s->audio_network_device = initialize_audio_network(
&s->audio_network_parameters);
if (!s->audio_network_device) {
s->audio_network_parameters.recv_port = old_rx_port;
s->audio_network_device = old_audio_network_device;
string err = string("Changing audio RX port to ") +
to_string(msg->new_rx_port) + " failed!";
LOG(LOG_LEVEL_ERROR) << err << "\n";
return new_response(RESPONSE_INT_SERV_ERR, err.c_str());
} else {
rtp_done(old_audio_network_device);
LOG(LOG_LEVEL_INFO) << "Successfully changed audio "
"RX port to " << msg->new_rx_port << ".\n";
}
break;
}
case RECEIVER_MSG_GET_VOLUME:
{
double ret = s->volume;
char volume_str[128] = "";
snprintf(volume_str, sizeof volume_str, "%f", ret);
return new_response(RESPONSE_OK, volume_str);
break;
}
case RECEIVER_MSG_INCREASE_VOLUME:
case RECEIVER_MSG_DECREASE_VOLUME:
case RECEIVER_MSG_MUTE:
{
if (msg->type == RECEIVER_MSG_MUTE) {
s->muted_receiver = !s->muted_receiver;
} else if (msg->type == RECEIVER_MSG_INCREASE_VOLUME) {
s->volume *= 1.1;
} else {
s->volume /= 1.1;
}
double new_volume = s->muted_receiver ? 0.0 : s->volume;
double db = 20.0 * log10(new_volume);
log_msg(LOG_LEVEL_INFO, "Playback volume: %.2f%% (%+.2f dB)\n", new_volume * 100.0, db);
struct pdb_e *cp;
pdb_iter_t it;
cp = pdb_iter_init(s->audio_participants, &it);
while (cp != NULL) {
struct audio_decoder *dec_state = (struct audio_decoder *) cp->decoder_state;
if (dec_state) {
audio_decoder_set_volume(dec_state->pbuf_data.decoder, new_volume);
}
cp = pdb_iter_next(&it);
}
pdb_iter_done(&it);
break;
}
default:
abort();
}
return new_response(RESPONSE_OK, NULL);
}
static void audio_decoder_state_deleter(void *state)
{
struct audio_decoder *s = (struct audio_decoder *) state;
free(s->pbuf_data.buffer.data);
audio_decoder_destroy(s->pbuf_data.decoder);
free(s);
}
static void audio_update_recv_buf(struct state_audio *s, size_t curr_frame_len)
{
size_t new_size = curr_frame_len * 2;
if (new_size > s->recv_buf_size) {
s->recv_buf_size = new_size;
debug_msg("[Audio receiver] Recv buffer adjusted to %zd\n", new_size);
rtp_set_recv_buf(s->audio_network_device, s->recv_buf_size);
}
}
static void *audio_receiver_thread(void *arg)
{
set_thread_name(__func__);
struct state_audio *s = (struct state_audio *) arg;
// rtp variables
struct timeval timeout, curr_time;
uint32_t ts;
struct pdb_e *cp;
struct audio_desc device_desc{};
bool playback_supports_multiple_streams;
struct pbuf_audio_data *current_pbuf = NULL;
#ifdef HAVE_JACK_TRANS
struct pbuf_audio_data jack_pbuf{};
current_pbuf = &jack_pbuf;
#endif
size_t len = sizeof playback_supports_multiple_streams;
if (!audio_playback_ctl(s->audio_playback_device, AUDIO_PLAYBACK_CTL_MULTIPLE_STREAMS,
&playback_supports_multiple_streams, &len)) {
playback_supports_multiple_streams = false;
}
printf("Audio receiving started.\n");
while (!should_exit) {
struct message *msg;
while((msg= check_message(&s->audio_receiver_module))) {
struct response *r = audio_receiver_process_message(s, (struct msg_receiver *) msg);
free_message(msg, r);
}
bool decoded = false;
if (s->receiver == NET_NATIVE || s->receiver == NET_STANDARD) {
gettimeofday(&curr_time, NULL);
auto curr_time_hr = std::chrono::high_resolution_clock::now();
ts = std::chrono::duration_cast<std::chrono::duration<double>>(s->start_time - std::chrono::steady_clock::now()).count() * 90000;
rtp_update(s->audio_network_device, curr_time);
rtp_send_ctrl(s->audio_network_device, ts, 0, curr_time);
timeout.tv_sec = 0;
// timeout.tv_usec = 999999 / 59.94; // audio goes almost always at the same rate
// as video frames
timeout.tv_usec = 1000; // this stuff really smells !!!
rtp_recv_r(s->audio_network_device, &timeout, ts);
pdb_iter_t it;
cp = pdb_iter_init(s->audio_participants, &it);
while (cp != NULL) {
if (cp->decoder_state == NULL &&
!pbuf_is_empty(cp->playout_buffer)) { // the second check is need ed because we want to assign display to participant that really sends data
// disable all previous sources
if (!playback_supports_multiple_streams) {
pdb_iter_t it;
struct pdb_e *cp = pdb_iter_init(s->audio_participants, &it);
while (cp != NULL) {
if(cp->decoder_state) {
((struct audio_decoder *) cp->decoder_state)->enabled = false;
}
cp = pdb_iter_next(&it);
}
pdb_iter_done(&it);
}
struct audio_decoder *dec_state;
dec_state = (struct audio_decoder *) calloc(1, sizeof(struct audio_decoder));
if (get_commandline_param("low-latency-audio")) {
pbuf_set_playout_delay(cp->playout_buffer, strcmp(get_commandline_param("low-latency-audio"), "ultra") == 0 ? 0.001 :0.005);
}
assert(dec_state != NULL);
cp->decoder_state = dec_state;
dec_state->enabled = true;
dec_state->pbuf_data.decoder = (struct state_audio_decoder *) audio_decoder_init(s->audio_channel_map, s->audio_scale, s->requested_encryption, (audio_playback_ctl_t) audio_playback_ctl, s->audio_playback_device, &s->audio_receiver_module);
audio_decoder_set_volume(dec_state->pbuf_data.decoder, s->muted_receiver ? 0.0 : s->volume);
assert(dec_state->pbuf_data.decoder != NULL);
cp->decoder_state_deleter = audio_decoder_state_deleter;
}
struct audio_decoder *dec_state = (struct audio_decoder *) cp->decoder_state;
if (dec_state && dec_state->enabled) {
dec_state->pbuf_data.buffer.data_len = 0;
// We iterate in loop since there can be more than one frmae present in
// the playout buffer and it would be discarded by following pbuf_remove()
// call.
while (pbuf_decode(cp->playout_buffer, curr_time_hr, s->receiver == NET_NATIVE ? decode_audio_frame : decode_audio_frame_mulaw, &dec_state->pbuf_data)) {
current_pbuf = &dec_state->pbuf_data;
decoded = true;
}
}
pbuf_remove(cp->playout_buffer, curr_time_hr);
cp = pdb_iter_next(&it);
if (decoded && !playback_supports_multiple_streams)
break;
}
pdb_iter_done(&it);
}else { /* NET_JACK */
#ifdef HAVE_JACK_TRANS
decoded = jack_receive(s->jack_connection, &jack_pbuf);
#endif
}
if (decoded) {
bool failed = false;
if(s->echo_state) {
#ifdef HAVE_SPEEXDSP
echo_play(s->echo_state, &current_pbuf->buffer);
#endif
}
struct audio_desc curr_desc;
curr_desc = audio_desc_from_audio_frame(&current_pbuf->buffer);
if(!audio_desc_eq(device_desc, curr_desc)) {
int log_l;
string msg;
if (audio_playback_reconfigure(s->audio_playback_device, curr_desc.bps * 8,
curr_desc.ch_count,
curr_desc.sample_rate) != TRUE) {
log_l = LOG_LEVEL_ERROR;
msg = "Audio reconfiguration failed";
failed = true;
}
else {
log_l = LOG_LEVEL_INFO;
msg = "Audio reconfiguration succeeded";
device_desc = curr_desc;
rtp_flush_recv_buf(s->audio_network_device);
}
LOG(log_l) << msg << " (" << curr_desc << ")" << (log_l < LOG_LEVEL_WARNING ? "!" : ".") << "\n";
rtp_flush_recv_buf(s->audio_network_device);
}
if (failed) {
continue;
}
audio_update_recv_buf(s, current_pbuf->frame_size);
if (!playback_supports_multiple_streams) {
audio_playback_put_frame(s->audio_playback_device, &current_pbuf->buffer);
} else {
pdb_iter_t it;
cp = pdb_iter_init(s->audio_participants, &it);
while (cp != NULL) {
struct audio_decoder *dec_state = (struct audio_decoder *) cp->decoder_state;
if (dec_state && dec_state->enabled && dec_state->pbuf_data.buffer.data_len > 0) {
struct audio_frame *f = &dec_state->pbuf_data.buffer;
f->network_source = &dec_state->pbuf_data.source;
audio_playback_put_frame(s->audio_playback_device, f);
}
cp = pdb_iter_next(&it);
}
pdb_iter_done(&it);
}
}
}
#ifdef HAVE_JACK_TRANS
free(jack_pbuf.buffer.data);
#endif
return NULL;
}
static struct response *audio_sender_process_message(struct state_audio *s, struct msg_sender *msg)
{
switch (msg->type) {
case SENDER_MSG_CHANGE_RECEIVER:
{
assert(s->audio_tx_mode == MODE_SENDER);
auto old_device = s->audio_network_device;
auto old_receiver = s->audio_network_parameters.addr;
s->audio_network_parameters.addr = strdup(msg->receiver);
s->audio_network_device =
initialize_audio_network(&s->audio_network_parameters);
if (!s->audio_network_device) {
s->audio_network_device = old_device;
free(s->audio_network_parameters.addr);
s->audio_network_parameters.addr = old_receiver;
return new_response(RESPONSE_INT_SERV_ERR, "Changing receiver failed!");
} else {
free(old_receiver);
rtp_done(old_device);
}
break;
}
case SENDER_MSG_CHANGE_PORT:
{
assert(s->audio_tx_mode == MODE_SENDER);
auto old_device = s->audio_network_device;
auto old_port = s->audio_network_parameters.send_port;
s->audio_network_parameters.send_port = msg->tx_port;
if (msg->rx_port) {
s->audio_network_parameters.recv_port = msg->rx_port;
}
s->audio_network_device =
initialize_audio_network(&s->audio_network_parameters);
if (!s->audio_network_device) {
s->audio_network_device = old_device;
s->audio_network_parameters.send_port = old_port;
return new_response(RESPONSE_INT_SERV_ERR, "Changing receiver failed!");
} else {
rtp_done(old_device);
}
break;
}
break;
case SENDER_MSG_GET_STATUS:
{
char status[128] = "";
snprintf(status, sizeof status, "%d", (int) s->muted_sender);
return new_response(RESPONSE_OK, status);
break;
}
case SENDER_MSG_PAUSE:
s->paused = true;
break;
case SENDER_MSG_MUTE:
s->muted_sender = !s->muted_sender;
log_msg(LOG_LEVEL_NOTICE, "Audio sender %smuted.\n", s->muted_sender ? "" : "un");
break;
case SENDER_MSG_PLAY:
s->paused = false;
break;
case SENDER_MSG_QUERY_VIDEO_MODE:
return new_response(RESPONSE_BAD_REQUEST, NULL);
case SENDER_MSG_RESET_SSRC:
{
assert(s->audio_tx_mode == MODE_SENDER);
uint32_t old_ssrc = rtp_my_ssrc(s->audio_network_device);
auto old_devices = s->audio_network_device;
s->audio_network_device =
initialize_audio_network(&s->audio_network_parameters);
if (!s->audio_network_device) {
s->audio_network_device = old_devices;
log_msg(LOG_LEVEL_ERROR, "[control] Audio: Unable to change SSRC!\n");
return new_response(RESPONSE_INT_SERV_ERR, NULL);
} else {
rtp_done(old_devices);
log_msg(LOG_LEVEL_NOTICE, "[control] Audio: changed SSRC from 0x%08" PRIx32 " to "
"0x%08" PRIx32 ".\n", old_ssrc, rtp_my_ssrc(s->audio_network_device));
}
}
break;
case SENDER_MSG_CHANGE_FEC:
if (strcmp(msg->fec_cfg, "flush") != 0) {
LOG(LOG_LEVEL_ERROR) << "Not implemented!\n";
}
return new_response(RESPONSE_NOT_IMPL, NULL);
}
return new_response(RESPONSE_OK, NULL);
}
struct asend_stats_processing_data {
audio_frame2 frame;
double seconds;
};
static void *asend_compute_and_print_stats(void *arg) {
auto d = (struct asend_stats_processing_data*) arg;
log_msg(LOG_LEVEL_INFO, "[Audio sender] Sent %d samples in last %f seconds.\n",
d->frame.get_sample_count(),
d->seconds);
for (int i = 0; i < d->frame.get_channel_count(); ++i) {
double rms, peak;
rms = calculate_rms(&d->frame, i, &peak);
LOG(LOG_LEVEL_INFO) << "[Audio sender] Channel " << i << " - volume: " << setprecision(2) << fixed << fg::green << style::bold << 20 * log(rms) / log(10) << style::reset << fg::reset << " dBFS RMS, " << fg::green << style::bold << 20 * log(peak) / log(10) << style::reset << fg::reset << " dBFS peak.\n";
}
delete d;
return NULL;
}
static void process_statistics(struct state_audio *s, audio_frame2 *buffer)
{
if (!s->captured.has_same_prop_as(*buffer)) {
s->captured.init(buffer->get_channel_count(), buffer->get_codec(),
buffer->get_bps(), buffer->get_sample_rate());
}
s->captured.append(*buffer);
struct timeval t;
double seconds;
gettimeofday(&t, 0);
seconds = tv_diff(t, s->t0);
if (seconds > 5.0) {
auto d = new asend_stats_processing_data;
std::swap(d->frame, s->captured);
d->seconds = seconds;
task_run_async_detached(asend_compute_and_print_stats, d);
s->t0 = t;
}
}
static int find_codec_sample_rate(int sample_rate, const int *supported) {
if (!supported) {
return 0;
}
int rate_hi = 0, // nearest high
rate_lo = 0; // nearest low
const int *tmp = supported;
while (*tmp != 0) {
if (*tmp == sample_rate) {
return sample_rate;
}
if (*tmp > sample_rate && (rate_hi == 0 || *tmp < rate_hi)) {
rate_hi = *tmp;
}
if (*tmp < sample_rate && *tmp > rate_lo) {
rate_lo = *tmp;
}
tmp++;
}
return rate_hi > 0 ? rate_hi : rate_lo;
}
static void *audio_sender_thread(void *arg)
{
set_thread_name(__func__);
struct state_audio *s = (struct state_audio *) arg;
struct audio_frame *buffer = NULL;
audio_frame2_resampler resampler_state;
printf("Audio sending started.\n");
while (!should_exit) {
struct message *msg;
while((msg = check_message(&s->audio_sender_module))) {
struct response *r = audio_sender_process_message(s, (struct msg_sender *) msg);
free_message(msg, r);
}
if ((s->audio_tx_mode & MODE_RECEIVER) == 0) { // otherwise receiver thread does the stuff...
struct timeval curr_time;
uint32_t ts;
gettimeofday(&curr_time, NULL);
ts = std::chrono::duration_cast<std::chrono::duration<double>>(s->start_time - std::chrono::steady_clock::now()).count() * 90000;
rtp_update(s->audio_network_device, curr_time);
rtp_send_ctrl(s->audio_network_device, ts, 0, curr_time);
// receive RTCP
struct timeval timeout;
timeout.tv_sec = 0;
timeout.tv_usec = 0;
rtcp_recv_r(s->audio_network_device, &timeout, ts);
}
buffer = audio_capture_read(s->audio_capture_device);
if(buffer) {
if (s->muted_sender) {
memset(buffer->data, 0, buffer->data_len);
}
export_audio(s->exporter, buffer);
if(s->echo_state) {
#ifdef HAVE_SPEEXDSP
buffer = echo_cancel(s->echo_state, buffer);
if(!buffer)
continue;
#endif
}
if (s->paused) {
continue;
}
audio_frame2 bf_n(buffer);
// RESAMPLE
int resample_to = s->resample_to;
if (resample_to == 0) {
const int *supp_sample_rates = audio_codec_get_supported_samplerates(s->audio_coder);
resample_to = find_codec_sample_rate(bf_n.get_sample_rate(),
supp_sample_rates);
}
if (resample_to != 0 && bf_n.get_sample_rate() != s->resample_to) {
if (bf_n.get_bps() != 2) {
bf_n.change_bps(2);
}
bf_n.resample(resampler_state, resample_to);
}
// COMPRESS
process_statistics(s, &bf_n);
// SEND
if(s->sender == NET_NATIVE) {
audio_frame2 *uncompressed = &bf_n;
const audio_frame2 *compressed = NULL;
while((compressed = audio_codec_compress(s->audio_coder, uncompressed))) {
audio_tx_send(s->tx_session, s->audio_network_device, compressed);
uncompressed = NULL;
}
}else if(s->sender == NET_STANDARD){
audio_frame2 *uncompressed = &bf_n;
const audio_frame2 *compressed = NULL;
while((compressed = audio_codec_compress(s->audio_coder, uncompressed))) {
//TODO to be dynamic as a function of the selected codec, now only accepting mulaw without checking errors
audio_tx_send_standard(s->tx_session, s->audio_network_device, compressed);
uncompressed = NULL;
}
}
#ifdef HAVE_JACK_TRANS
else
jack_send(s->jack_connection, buffer);
#endif
}
}
return NULL;
}
void audio_sdi_send(struct state_audio *s, struct audio_frame *frame) {
void *sdi_capture;
if (!s->audio_capture_device)
return;
if(!audio_capture_get_vidcap_flags(audio_capture_get_driver_name(s->audio_capture_device)))
return;
sdi_capture = audio_capture_get_state_pointer(s->audio_capture_device);
sdi_capture_new_incoming_frame(sdi_capture, frame);
}
void audio_register_display_callbacks(struct state_audio *s, void *udata, void (*putf)(void *, struct audio_frame *), int (*reconfigure)(void *, int, int, int), int (*get_property)(void *, int, void *, size_t *))
{
struct state_sdi_playback *sdi_playback;
if(!audio_playback_get_display_flags(s->audio_playback_device))
return;
sdi_playback = (struct state_sdi_playback *) audio_playback_get_state_pointer(s->audio_playback_device);
sdi_register_display_callbacks(sdi_playback, udata, putf, reconfigure, get_property);
}
unsigned int audio_get_display_flags(struct state_audio *s)
{
return audio_playback_get_display_flags(s->audio_playback_device);
}
struct audio_desc audio_desc_from_frame(struct audio_frame *frame)
{
return (struct audio_desc) { frame->bps, frame->sample_rate,
frame->ch_count, AC_PCM };
}
std::ostream& operator<<(std::ostream& os, const audio_desc& desc)
{
os << desc.ch_count << " channel" << (desc.ch_count > 1 ? "s" : "") << ", " << desc.bps << " Bps, " << desc.sample_rate << " Hz, codec: " << get_name_to_audio_codec(desc.codec);
return os;
}
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