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Added OPUS stereo for Asterisk Master (Version 15) #8

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Added OPUS stereo for Asterisk Master (Version 15) #8

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87 changes: 70 additions & 17 deletions codecs/codec_opus_open_source.c
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Original file line number Diff line number Diff line change
Expand Up @@ -60,7 +60,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision: $")

#include "asterisk/opus.h" /* for CODEC_OPUS_DEFAULT_* */

#define BUFFER_SAMPLES 5760
#define BUFFER_SAMPLES 11520
#define MAX_CHANNELS 2
#define OPUS_SAMPLES 960

Expand Down Expand Up @@ -90,6 +90,7 @@ struct opus_coder_pvt {
int multiplier;
int id;
int16_t buf[BUFFER_SAMPLES];
int16_t buf_stereo[BUFFER_SAMPLES * 2];
int framesize;
int inited;
int channels;
Expand All @@ -106,7 +107,7 @@ struct opus_attr {
unsigned int fec;
unsigned int dtx;
unsigned int spropmaxcapturerate; /* FIXME: not utilised, yet */
unsigned int spropstereo; /* FIXME: currently, we are just mono */
unsigned int spropstereo;
};

/* Helper methods */
Expand All @@ -116,7 +117,7 @@ static int opus_encoder_construct(struct ast_trans_pvt *pvt, int sampling_rate)
struct opus_attr *attr = pvt->explicit_dst ? ast_format_get_attribute_data(pvt->explicit_dst) : NULL;
const opus_int32 bitrate = attr ? attr->maxbitrate : CODEC_OPUS_DEFAULT_BITRATE;
const int maxplayrate = attr ? attr->maxplayrate : CODEC_OPUS_DEFAULT_MAX_PLAYBACK_RATE;
const int channels = attr ? attr->stereo + 1 : CODEC_OPUS_DEFAULT_STEREO + 1;
const int channels = attr ? attr->spropstereo + 1 : CODEC_OPUS_DEFAULT_STEREO + 1;
const opus_int32 vbr = attr ? !(attr->cbr) : !CODEC_OPUS_DEFAULT_CBR;
const opus_int32 fec = attr ? attr->fec : CODEC_OPUS_DEFAULT_FEC;
const opus_int32 dtx = attr ? attr->dtx : CODEC_OPUS_DEFAULT_DTX;
Expand Down Expand Up @@ -150,24 +151,29 @@ static int opus_encoder_construct(struct ast_trans_pvt *pvt, int sampling_rate)
opvt->sampling_rate = sampling_rate;
opvt->multiplier = 48000 / sampling_rate;
opvt->framesize = sampling_rate / 50;
opvt->channels = channels;
opvt->id = ast_atomic_fetchadd_int(&usage.encoder_id, 1) + 1;

ast_atomic_fetchadd_int(&usage.encoders, +1);

ast_debug(3, "Created encoder #%d (%d -> opus)\n", opvt->id, sampling_rate);
ast_debug(3, "Created encoder #%d (%d -> opus) (channels %d)\n", opvt->id, sampling_rate, opvt->channels);

return 0;
}

static int opus_decoder_construct(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
struct opus_coder_pvt *opvt = pvt->pvt;
/* struct opus_attr *attr = ast_format_get_attribute_data(f->subclass.format); */
struct opus_attr *attr = ast_format_get_attribute_data(f->subclass.format);
int error = 0;

opvt->sampling_rate = pvt->t->dst_codec.sample_rate;
opvt->multiplier = 48000 / opvt->sampling_rate;
opvt->channels = /* attr ? attr->spropstereo + 1 :*/ 1; /* FIXME */;

opvt->channels = CODEC_OPUS_DEFAULT_STEREO + 1;
if (attr != NULL) {
opvt->channels = attr->stereo + 1;
}

opvt->opus = opus_decoder_create(opvt->sampling_rate, opvt->channels, &error);

Expand All @@ -180,7 +186,7 @@ static int opus_decoder_construct(struct ast_trans_pvt *pvt, struct ast_frame *f

ast_atomic_fetchadd_int(&usage.decoders, +1);

ast_debug(3, "Created decoder #%d (opus -> %d)\n", opvt->id, opvt->sampling_rate);
ast_debug(3, "Created decoder #%d (opus -> %d) (channels %d)\n", opvt->id, opvt->sampling_rate, opvt->channels);

return 0;
}
Expand All @@ -205,6 +211,16 @@ static int lintoopus_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
struct opus_coder_pvt *opvt = pvt->pvt;

/* One time upgrade to stereo. */
struct opus_attr *attr = ast_format_get_attribute_data(pvt->f.subclass.format);
if (attr != NULL) {
if (attr->spropstereo + 1 != opvt->channels) {
ast_log(LOG_ERROR, "Changing attr %d opvt channels %d \n", attr->stereo, opvt->channels);
opus_encoder_destroy(opvt->opus);
opus_encoder_construct(pvt, pvt->t->src_codec.sample_rate);
}
}

/* XXX We should look at how old the rest of our stream is, and if it
is too old, then we should overwrite it entirely, otherwise we can
get artifacts of earlier talk that do not belong */
Expand All @@ -220,17 +236,30 @@ static struct ast_frame *lintoopus_frameout(struct ast_trans_pvt *pvt)
struct ast_frame *result = NULL;
struct ast_frame *last = NULL;
int samples = 0; /* output samples */
int interleaved = pvt->interleaved_stereo ? 2 : 1;
int status;
int i;
int k;

while (pvt->samples >= opvt->framesize * interleaved) {

if (opvt->channels == 2 && pvt->interleaved_stereo == 0) {
/* No stereo samples, but stereo output (put the same audio on both channels). */
opus_int16 stereobuf[opvt->framesize * 2];
k = 0;
for (i = 0; i < opvt->framesize * 2; i += 2) {
stereobuf[i] = opvt->buf[samples + k];
stereobuf[i + 1] = opvt->buf[samples + k];
k++;
}
status = opus_encode(opvt->opus, stereobuf, opvt->framesize, pvt->outbuf.uc, BUFFER_SAMPLES);
} else {
/* Stereo source (interleaved format) and stereo output or everything mono. */
status = opus_encode(opvt->opus, opvt->buf + samples, opvt->framesize, pvt->outbuf.uc, BUFFER_SAMPLES);
}

while (pvt->samples >= opvt->framesize) {
/* status is either error or output bytes */
const int status = opus_encode(opvt->opus,
opvt->buf + samples,
opvt->framesize,
pvt->outbuf.uc,
BUFFER_SAMPLES);

samples += opvt->framesize;
pvt->samples -= opvt->framesize;
samples += opvt->framesize * interleaved;
pvt->samples -= opvt->framesize * interleaved;

if (status < 0) {
ast_log(LOG_ERROR, "Error encoding the Opus frame: %s\n", opus_strerror(status));
Expand Down Expand Up @@ -267,6 +296,8 @@ static int opustolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
opus_int32 len;
unsigned char *src;
int status;
int k;
int i;

if (!opvt->inited && f->datalen == 0) {
return 0; /* we cannot start without data */
Expand All @@ -291,6 +322,28 @@ static int opustolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
}
decode_fec = opvt->decode_fec_incoming;

/*
* In case of stereo, we do not use FEC or PLC (yet).
*/
if (opvt->channels == 2) {
/*
* If we have incoming stereo signals, we will only copy one channel to lin.
* Asterisk can't handle stereo signals internally at the moment.
*/
status = opus_decode(opvt->opus, f->data.ptr, f->datalen, opvt->buf_stereo, BUFFER_SAMPLES, decode_fec);
if (status < 0) {
ast_log(LOG_ERROR, "Error decoding the Opus stereo frame: %s\n", opus_strerror(status));
}
k = 0;
for (i = 0; i < status * 2; i += 2) {
pvt->outbuf.i16[k] = opvt->buf_stereo[i];
k++;
}
pvt->samples += status;
pvt->datalen += status * opvt->channels * sizeof(int16_t);
return 0;
}

/*
* The Opus Codec, actually its library allows
* - Forward-Error Correction (FEC), and
Expand Down
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