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6356eb4e4f
toxcore/toxav/audio.c
iphydf 13ae9e9a93
Move logging to a callback. 
This removes the global logger (which by the way was deleted when the first tox
was killed, so other toxes would then stop logging). Various bits of the code
now carry a logger or pass it around. It's a bit less transparent now, but now
there is no need to have a global logger, and clients can decide what to log and
where.
2016-08-27 01:16:14 +01:00

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/** audio.c
*
* Copyright (C) 2013-2015 Tox project All Rights Reserved.
*
* This file is part of Tox.
*
* Tox is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* Tox is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with Tox. If not, see <http://www.gnu.org/licenses/>.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif /* HAVE_CONFIG_H */
#include <stdlib.h>
#include "audio.h"
#include "rtp.h"
#include "../toxcore/logger.h"
static struct JitterBuffer *jbuf_new(uint32_t capacity);
static void jbuf_clear(struct JitterBuffer *q);
static void jbuf_free(struct JitterBuffer *q);
static int jbuf_write(Logger *log, struct JitterBuffer *q, struct RTPMessage *m);
static struct RTPMessage *jbuf_read(struct JitterBuffer *q, int32_t *success);
OpusEncoder *create_audio_encoder (Logger *log, int32_t bit_rate, int32_t sampling_rate, int32_t channel_count);
bool reconfigure_audio_encoder(Logger *log, OpusEncoder **e, int32_t new_br, int32_t new_sr, uint8_t new_ch,
int32_t *old_br, int32_t *old_sr, int32_t *old_ch);
bool reconfigure_audio_decoder(ACSession *ac, int32_t sampling_rate, int8_t channels);
ACSession *ac_new(Logger *log, ToxAV *av, uint32_t friend_number, toxav_audio_receive_frame_cb *cb, void *cb_data)
{
ACSession *ac = calloc(sizeof(ACSession), 1);
if (!ac) {
LOGGER_WARNING(log, "Allocation failed! Application might misbehave!");
return NULL;
}
if (create_recursive_mutex(ac->queue_mutex) != 0) {
LOGGER_WARNING(log, "Failed to create recursive mutex!");
free(ac);
return NULL;
}
int status;
ac->decoder = opus_decoder_create(48000, 2, &status);
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while starting audio decoder: %s", opus_strerror(status));
goto BASE_CLEANUP;
}
if (!(ac->j_buf = jbuf_new(3))) {
LOGGER_WARNING(log, "Jitter buffer creaton failed!");
opus_decoder_destroy(ac->decoder);
goto BASE_CLEANUP;
}
ac->log = log;
/* Initialize encoders with default values */
ac->encoder = create_audio_encoder(log, 48000, 48000, 2);
if (ac->encoder == NULL)
goto DECODER_CLEANUP;
ac->le_bit_rate = 48000;
ac->le_sample_rate = 48000;
ac->le_channel_count = 2;
ac->ld_channel_count = 2;
ac->ld_sample_rate = 48000;
ac->ldrts = 0; /* Make it possible to reconfigure straight away */
/* These need to be set in order to properly
* do error correction with opus */
ac->lp_frame_duration = 120;
ac->lp_sampling_rate = 48000;
ac->lp_channel_count = 1;
ac->av = av;
ac->friend_number = friend_number;
ac->acb.first = cb;
ac->acb.second = cb_data;
return ac;
DECODER_CLEANUP:
opus_decoder_destroy(ac->decoder);
jbuf_free(ac->j_buf);
BASE_CLEANUP:
pthread_mutex_destroy(ac->queue_mutex);
free(ac);
return NULL;
}
void ac_kill(ACSession *ac)
{
if (!ac)
return;
opus_encoder_destroy(ac->encoder);
opus_decoder_destroy(ac->decoder);
jbuf_free(ac->j_buf);
pthread_mutex_destroy(ac->queue_mutex);
LOGGER_DEBUG(ac->log, "Terminated audio handler: %p", ac);
free(ac);
}
void ac_iterate(ACSession *ac)
{
if (!ac)
return;
/* TODO fix this and jitter buffering */
/* Enough space for the maximum frame size (120 ms 48 KHz stereo audio) */
int16_t tmp[5760 * 2];
struct RTPMessage *msg;
int rc = 0;
pthread_mutex_lock(ac->queue_mutex);
while ((msg = jbuf_read(ac->j_buf, &rc)) || rc == 2) {
pthread_mutex_unlock(ac->queue_mutex);
if (rc == 2) {
LOGGER_DEBUG(ac->log, "OPUS correction");
int fs = (ac->lp_sampling_rate * ac->lp_frame_duration) / 1000;
rc = opus_decode(ac->decoder, NULL, 0, tmp, fs, 1);
} else {
/* Get values from packet and decode. */
/* NOTE: This didn't work very well */
#if 0
rc = convert_bw_to_sampling_rate(opus_packet_get_bandwidth(msg->data));
if (rc != -1) {
cs->last_packet_sampling_rate = rc;
} else {
LOGGER_WARNING(ac->log, "Failed to load packet values!");
rtp_free_msg(msg);
continue;
}
#endif
/* Pick up sampling rate from packet */
memcpy(&ac->lp_sampling_rate, msg->data, 4);
ac->lp_sampling_rate = ntohl(ac->lp_sampling_rate);
ac->lp_channel_count = opus_packet_get_nb_channels(msg->data + 4);
/** NOTE: even though OPUS supports decoding mono frames with stereo decoder and vice versa,
* it didn't work quite well.
*/
if (!reconfigure_audio_decoder(ac, ac->lp_sampling_rate, ac->lp_channel_count)) {
LOGGER_WARNING(ac->log, "Failed to reconfigure decoder!");
free(msg);
continue;
}
rc = opus_decode(ac->decoder, msg->data + 4, msg->len - 4, tmp, 5760, 0);
free(msg);
}
if (rc < 0) {
LOGGER_WARNING(ac->log, "Decoding error: %s", opus_strerror(rc));
} else if (ac->acb.first) {
ac->lp_frame_duration = (rc * 1000) / ac->lp_sampling_rate;
ac->acb.first(ac->av, ac->friend_number, tmp, rc, ac->lp_channel_count,
ac->lp_sampling_rate, ac->acb.second);
}
return;
}
pthread_mutex_unlock(ac->queue_mutex);
}
int ac_queue_message(void *acp, struct RTPMessage *msg)
{
if (!acp || !msg)
return -1;
ACSession *ac = acp;
if ((msg->header.pt & 0x7f) == (rtp_TypeAudio + 2) % 128) {
LOGGER_WARNING(ac->log, "Got dummy!");
free(msg);
return 0;
}
if ((msg->header.pt & 0x7f) != rtp_TypeAudio % 128) {
LOGGER_WARNING(ac->log, "Invalid payload type!");
free(msg);
return -1;
}
pthread_mutex_lock(ac->queue_mutex);
int rc = jbuf_write(ac->log, ac->j_buf, msg);
pthread_mutex_unlock(ac->queue_mutex);
if (rc == -1) {
LOGGER_WARNING(ac->log, "Could not queue the message!");
free(msg);
return -1;
}
return 0;
}
int ac_reconfigure_encoder(ACSession *ac, int32_t bit_rate, int32_t sampling_rate, uint8_t channels)
{
if (!ac || !reconfigure_audio_encoder(ac->log, &ac->encoder, bit_rate,
sampling_rate, channels,
&ac->le_bit_rate,
&ac->le_sample_rate,
&ac->le_channel_count))
return -1;
return 0;
}
struct JitterBuffer {
struct RTPMessage **queue;
uint32_t size;
uint32_t capacity;
uint16_t bottom;
uint16_t top;
};
static struct JitterBuffer *jbuf_new(uint32_t capacity)
{
unsigned int size = 1;
while (size <= (capacity * 4)) {
size *= 2;
}
struct JitterBuffer *q;
if (!(q = calloc(sizeof(struct JitterBuffer), 1))) return NULL;
if (!(q->queue = calloc(sizeof(struct RTPMessage *), size))) {
free(q);
return NULL;
}
q->size = size;
q->capacity = capacity;
return q;
}
static void jbuf_clear(struct JitterBuffer *q)
{
for (; q->bottom != q->top; ++q->bottom) {
if (q->queue[q->bottom % q->size]) {
free(q->queue[q->bottom % q->size]);
q->queue[q->bottom % q->size] = NULL;
}
}
}
static void jbuf_free(struct JitterBuffer *q)
{
if (!q) return;
jbuf_clear(q);
free(q->queue);
free(q);
}
static int jbuf_write(Logger *log, struct JitterBuffer *q, struct RTPMessage *m)
{
uint16_t sequnum = m->header.sequnum;
unsigned int num = sequnum % q->size;
if ((uint32_t)(sequnum - q->bottom) > q->size) {
LOGGER_DEBUG(log, "Clearing filled jitter buffer: %p", q);
jbuf_clear(q);
q->bottom = sequnum - q->capacity;
q->queue[num] = m;
q->top = sequnum + 1;
return 0;
}
if (q->queue[num])
return -1;
q->queue[num] = m;
if ((sequnum - q->bottom) >= (q->top - q->bottom))
q->top = sequnum + 1;
return 0;
}
static struct RTPMessage *jbuf_read(struct JitterBuffer *q, int32_t *success)
{
if (q->top == q->bottom) {
*success = 0;
return NULL;
}
unsigned int num = q->bottom % q->size;
if (q->queue[num]) {
struct RTPMessage *ret = q->queue[num];
q->queue[num] = NULL;
++q->bottom;
*success = 1;
return ret;
}
if ((uint32_t)(q->top - q->bottom) > q->capacity) {
++q->bottom;
*success = 2;
return NULL;
}
*success = 0;
return NULL;
}
OpusEncoder *create_audio_encoder (Logger *log, int32_t bit_rate, int32_t sampling_rate, int32_t channel_count)
{
int status = OPUS_OK;
OpusEncoder *rc = opus_encoder_create(sampling_rate, channel_count, OPUS_APPLICATION_VOIP, &status);
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while starting audio encoder: %s", opus_strerror(status));
return NULL;
}
status = opus_encoder_ctl(rc, OPUS_SET_BITRATE(bit_rate));
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
/* Enable in-band forward error correction in codec */
status = opus_encoder_ctl(rc, OPUS_SET_INBAND_FEC(1));
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
/* Make codec resistant to up to 10% packet loss
* NOTE This could also be adjusted on the fly, rather than hard-coded,
* with feedback from the receiving client.
*/
status = opus_encoder_ctl(rc, OPUS_SET_PACKET_LOSS_PERC(10));
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
/* Set algorithm to the highest complexity, maximizing compression */
status = opus_encoder_ctl(rc, OPUS_SET_COMPLEXITY(10));
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
return rc;
FAILURE:
opus_encoder_destroy(rc);
return NULL;
}
bool reconfigure_audio_encoder(Logger *log, OpusEncoder **e, int32_t new_br, int32_t new_sr, uint8_t new_ch,
int32_t *old_br, int32_t *old_sr, int32_t *old_ch)
{
/* Values are checked in toxav.c */
if (*old_sr != new_sr || *old_ch != new_ch) {
OpusEncoder *new_encoder = create_audio_encoder(log, new_br, new_sr, new_ch);
if (new_encoder == NULL)
return false;
opus_encoder_destroy(*e);
*e = new_encoder;
} else if (*old_br == new_br)
return true; /* Nothing changed */
else {
int status = opus_encoder_ctl(*e, OPUS_SET_BITRATE(new_br));
if (status != OPUS_OK) {
LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
return false;
}
}
*old_br = new_br;
*old_sr = new_sr;
*old_ch = new_ch;
LOGGER_DEBUG(log, "Reconfigured audio encoder br: %d sr: %d cc:%d", new_br, new_sr, new_ch);
return true;
}
bool reconfigure_audio_decoder(ACSession *ac, int32_t sampling_rate, int8_t channels)
{
if (sampling_rate != ac->ld_sample_rate || channels != ac->ld_channel_count) {
if (current_time_monotonic() - ac->ldrts < 500)
return false;
int status;
OpusDecoder *new_dec = opus_decoder_create(sampling_rate, channels, &status);
if (status != OPUS_OK) {
LOGGER_ERROR(ac->log, "Error while starting audio decoder(%d %d): %s", sampling_rate, channels, opus_strerror(status));
return false;
}
ac->ld_sample_rate = sampling_rate;
ac->ld_channel_count = channels;
ac->ldrts = current_time_monotonic();
opus_decoder_destroy(ac->decoder);
ac->decoder = new_dec;
LOGGER_DEBUG(ac->log, "Reconfigured audio decoder sr: %d cc: %d", sampling_rate, channels);
}
return true;
}
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