toxcore/toxav/audio.c
2015-04-29 01:01:25 +02:00

424 lines
13 KiB
C

/** 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/>.
*
*/
#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(struct JitterBuffer *q, RTPMessage *m);
static RTPMessage *jbuf_read(struct JitterBuffer *q, int32_t *success);
OpusEncoder* create_audio_encoder (int32_t bit_rate, int32_t sampling_rate, int32_t channel_count);
bool reconfigure_audio_encoder(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(ToxAV* av, uint32_t friend_number, toxav_receive_audio_frame_cb *cb, void *cb_data)
{
ACSession *ac = calloc(sizeof(ACSession), 1);
if (!ac) {
LOGGER_WARNING("Allocation failed! Application might misbehave!");
return NULL;
}
if (create_recursive_mutex(ac->queue_mutex) != 0) {
LOGGER_WARNING("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("Error while starting audio decoder: %s", opus_strerror(status));
goto BASE_CLEANUP;
}
if ( !(ac->j_buf = jbuf_new(3)) ) {
LOGGER_WARNING("Jitter buffer creaton failed!");
opus_decoder_destroy(ac->decoder);
goto BASE_CLEANUP;
}
/* Initialize encoders with default values */
ac->encoder = create_audio_encoder(48000, 48000, 2);
if (ac->encoder == NULL)
goto DECODER_CLEANUP;
ac->test_encoder = create_audio_encoder(48000, 48000, 2);
if (ac->test_encoder == NULL) {
opus_encoder_destroy(ac->encoder);
goto DECODER_CLEANUP;
}
ac->last_encoding_bit_rate = 48000;
ac->last_encoding_sampling_rate = 48000;
ac->last_encoding_channel_count = 2;
ac->last_test_encoding_bit_rate = 48000;
ac->last_test_encoding_sampling_rate = 48000;
ac->last_test_encoding_channel_count = 2;
ac->last_decoding_channel_count = 2;
ac->last_decoding_sampling_rate = 48000;
ac->last_decoder_reconfiguration = 0; /* Make it possible to reconfigure straight away */
/* These need to be set in order to properly
* do error correction with opus */
ac->last_packet_frame_duration = 120;
ac->last_packet_sampling_rate = 48000;
ac->last_packet_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("Terminated audio handler: %p", ac);
free(ac);
}
void ac_do(ACSession* ac)
{
if (!ac)
return;
/* Enough space for the maximum frame size (120 ms 48 KHz audio) */
int16_t tmp[5760 * 2];
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("OPUS correction");
int fs = (ac->last_packet_sampling_rate * ac->last_packet_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
rc = convert_bw_to_sampling_rate(opus_packet_get_bandwidth(msg->data));
if (rc != -1) {
cs->last_packet_sampling_rate = rc;
} else {
LOGGER_WARNING("Failed to load packet values!");
rtp_free_msg(msg);
continue;
}*/
/* Pick up sampling rate from packet */
memcpy(&ac->last_packet_sampling_rate, msg->data, 4);
ac->last_packet_sampling_rate = ntohl(ac->last_packet_sampling_rate);
ac->last_packet_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->last_packet_sampling_rate, ac->last_packet_channel_count)) {
LOGGER_WARNING("Failed to reconfigure decoder!");
rtp_free_msg(msg);
continue;
}
rc = opus_decode(ac->decoder, msg->data + 4, msg->length - 4, tmp, 5760, 0);
rtp_free_msg(msg);
}
if (rc < 0) {
LOGGER_WARNING("Decoding error: %s", opus_strerror(rc));
} else if (ac->acb.first) {
ac->last_packet_frame_duration = (rc * 1000) / ac->last_packet_sampling_rate;
ac->acb.first(ac->av, ac->friend_number, tmp, rc * ac->last_packet_channel_count,
ac->last_packet_channel_count, ac->last_packet_sampling_rate, ac->acb.second);
}
return;
}
pthread_mutex_unlock(ac->queue_mutex);
}
int ac_queue_message(void* acp, struct RTPMessage_s *msg)
{
if (!acp || !msg)
return -1;
if ((msg->header->marker_payloadt & 0x7f) == (rtp_TypeAudio + 2) % 128) {
LOGGER_WARNING("Got dummy!");
rtp_free_msg(msg);
return 0;
}
if ((msg->header->marker_payloadt & 0x7f) != rtp_TypeAudio % 128) {
LOGGER_WARNING("Invalid payload type!");
rtp_free_msg(msg);
return -1;
}
ACSession* ac = acp;
pthread_mutex_lock(ac->queue_mutex);
int rc = jbuf_write(ac->j_buf, msg);
pthread_mutex_unlock(ac->queue_mutex);
if (rc == -1) {
LOGGER_WARNING("Could not queue the message!");
rtp_free_msg(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->encoder, bit_rate, sampling_rate, channels,
&ac->last_encoding_bit_rate, &ac->last_encoding_sampling_rate, &ac->last_encoding_channel_count))
return -1;
LOGGER_DEBUG ("Reconfigured audio encoder br: %d sr: %d cc:%d", bit_rate, sampling_rate, channels);
return 0;
}
int ac_reconfigure_test_encoder(ACSession* ac, int32_t bit_rate, int32_t sampling_rate, uint8_t channels)
{
if (!ac || !reconfigure_audio_encoder(&ac->test_encoder, bit_rate, sampling_rate, channels,
&ac->last_encoding_bit_rate, &ac->last_encoding_sampling_rate, &ac->last_encoding_channel_count))
return -1;
LOGGER_DEBUG ("Reconfigured test audio encoder br: %d sr: %d cc:%d", bit_rate, sampling_rate, channels);
return 0;
}
struct JitterBuffer {
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(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]) {
rtp_free_msg(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(struct JitterBuffer *q, RTPMessage *m)
{
uint16_t sequnum = m->header->sequnum;
unsigned int num = sequnum % q->size;
if ((uint32_t)(sequnum - q->bottom) > q->size) {
LOGGER_DEBUG("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 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]) {
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 (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_AUDIO, &status);
if ( status != OPUS_OK ) {
LOGGER_ERROR("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("Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
status = opus_encoder_ctl(rc, OPUS_SET_COMPLEXITY(10));
if ( status != OPUS_OK ) {
LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(status));
goto FAILURE;
}
return rc;
FAILURE:
opus_encoder_destroy(rc);
return NULL;
}
bool reconfigure_audio_encoder(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(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("Error while setting encoder ctl: %s", opus_strerror(status));
return false;
}
}
*old_br = new_br;
*old_sr = new_sr;
*old_ch = new_ch;
return true;
}
bool reconfigure_audio_decoder(ACSession* ac, int32_t sampling_rate, int8_t channels)
{
if (sampling_rate != ac->last_decoding_sampling_rate || channels != ac->last_decoding_channel_count) {
if (current_time_monotonic() - ac->last_decoder_reconfiguration < 500)
return false;
int status;
OpusDecoder* new_dec = opus_decoder_create(sampling_rate, channels, &status );
if ( status != OPUS_OK ) {
LOGGER_ERROR("Error while starting audio decoder(%d %d): %s", sampling_rate, channels, opus_strerror(status));
return false;
}
ac->last_decoding_sampling_rate = sampling_rate;
ac->last_decoding_channel_count = channels;
ac->last_decoder_reconfiguration = current_time_monotonic();
opus_decoder_destroy(ac->decoder);
ac->decoder = new_dec;
LOGGER_DEBUG("Reconfigured audio decoder sr: %d cc: %d", sampling_rate, channels);
}
return true;
}