mirror of
https://github.com/asterisk/asterisk.git
synced 2025-11-01 19:43:03 +00:00
media: Add experimental support for RTCP feedback.
This change adds experimental support for providing RTCP feedback information to codec modules so they can dynamically change themselves based on conditions. ASTERISK-26584 Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857
This commit is contained in:
@@ -55,6 +55,9 @@
|
||||
#include "asterisk/frame.h"
|
||||
#include "asterisk/linkedlists.h"
|
||||
|
||||
/* For struct ast_rtp_rtcp_report and struct ast_rtp_rtcp_report_block */
|
||||
#include "asterisk/rtp_engine.h"
|
||||
|
||||
/* codec variables */
|
||||
static int quality = 3;
|
||||
static int complexity = 2;
|
||||
@@ -64,6 +67,7 @@ static int vbr = 0;
|
||||
static float vbr_quality = 4;
|
||||
static int abr = 0;
|
||||
static int dtx = 0; /* set to 1 to enable silence detection */
|
||||
static int exp_rtcp_fb = 0; /* set to 1 to use experimental RTCP feedback for changing bitrate */
|
||||
|
||||
static int preproc = 0;
|
||||
static int pp_vad = 0;
|
||||
@@ -91,6 +95,11 @@ struct speex_coder_pvt {
|
||||
SpeexBits bits;
|
||||
int framesize;
|
||||
int silent_state;
|
||||
|
||||
int fraction_lost;
|
||||
int quality;
|
||||
int default_quality;
|
||||
|
||||
#ifdef _SPEEX_TYPES_H
|
||||
SpeexPreprocessState *pp;
|
||||
spx_int16_t buf[BUFFER_SAMPLES];
|
||||
@@ -137,6 +146,11 @@ static int speex_encoder_construct(struct ast_trans_pvt *pvt, const SpeexMode *p
|
||||
speex_encoder_ctl(tmp->speex, SPEEX_SET_DTX, &dtx);
|
||||
tmp->silent_state = 0;
|
||||
|
||||
tmp->fraction_lost = 0;
|
||||
tmp->default_quality = vbr ? vbr_quality : quality;
|
||||
tmp->quality = tmp->default_quality;
|
||||
ast_debug(3, "Default quality (%s): %d\n", vbr ? "vbr" : "cbr", tmp->default_quality);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
@@ -342,6 +356,69 @@ static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt)
|
||||
return result;
|
||||
}
|
||||
|
||||
/*! \brief handle incoming RTCP feedback and possibly edit encoder settings */
|
||||
static void lintospeex_feedback(struct ast_trans_pvt *pvt, struct ast_frame *feedback)
|
||||
{
|
||||
struct speex_coder_pvt *tmp = pvt->pvt;
|
||||
|
||||
struct ast_rtp_rtcp_report *rtcp_report;
|
||||
struct ast_rtp_rtcp_report_block *report_block;
|
||||
|
||||
int fraction_lost;
|
||||
int percent;
|
||||
int bitrate;
|
||||
int q;
|
||||
|
||||
if(!exp_rtcp_fb)
|
||||
return;
|
||||
|
||||
rtcp_report = (struct ast_rtp_rtcp_report *)feedback->data.ptr;
|
||||
if (rtcp_report->reception_report_count == 0)
|
||||
return;
|
||||
report_block = rtcp_report->report_block[0];
|
||||
fraction_lost = report_block->lost_count.fraction;
|
||||
if (fraction_lost == tmp->fraction_lost)
|
||||
return;
|
||||
/* Per RFC3550, fraction lost is defined to be the number of packets lost
|
||||
* divided by the number of packets expected. Since it's a 8-bit value,
|
||||
* and we want a percentage value, we multiply by 100 and divide by 256. */
|
||||
percent = (fraction_lost*100)/256;
|
||||
bitrate = 0;
|
||||
q = -1;
|
||||
ast_debug(3, "Fraction lost changed: %d --> %d percent loss\n", fraction_lost, percent);
|
||||
/* Handle change */
|
||||
speex_encoder_ctl(tmp->speex, SPEEX_GET_BITRATE, &bitrate);
|
||||
ast_debug(3, "Current bitrate: %d\n", bitrate);
|
||||
ast_debug(3, "Current quality: %d/%d\n", tmp->quality, tmp->default_quality);
|
||||
/* FIXME BADLY Very ugly example of how this could be handled: probably sucks */
|
||||
if (percent < 10) {
|
||||
/* Not that bad, default quality is fine */
|
||||
q = tmp->default_quality;
|
||||
} else if (percent < 20) {
|
||||
/* Quite bad, let's go down a bit */
|
||||
q = tmp->default_quality-1;
|
||||
} else if (percent < 30) {
|
||||
/* Very bad, let's go down even more */
|
||||
q = tmp->default_quality-2;
|
||||
} else {
|
||||
/* Really bad, use the lowest quality possible */
|
||||
q = 0;
|
||||
}
|
||||
if (q < 0)
|
||||
q = 0;
|
||||
if (q != tmp->quality) {
|
||||
ast_debug(3, " -- Setting to %d\n", q);
|
||||
if (vbr) {
|
||||
float vbr_q = q;
|
||||
speex_encoder_ctl(tmp->speex, SPEEX_SET_VBR_QUALITY, &vbr_q);
|
||||
} else {
|
||||
speex_encoder_ctl(tmp->speex, SPEEX_SET_QUALITY, &q);
|
||||
}
|
||||
tmp->quality = q;
|
||||
}
|
||||
tmp->fraction_lost = fraction_lost;
|
||||
}
|
||||
|
||||
static void speextolin_destroy(struct ast_trans_pvt *arg)
|
||||
{
|
||||
struct speex_coder_pvt *pvt = arg->pvt;
|
||||
@@ -400,6 +477,7 @@ static struct ast_translator lintospeex = {
|
||||
.newpvt = lintospeex_new,
|
||||
.framein = lintospeex_framein,
|
||||
.frameout = lintospeex_frameout,
|
||||
.feedback = lintospeex_feedback,
|
||||
.destroy = lintospeex_destroy,
|
||||
.sample = slin8_sample,
|
||||
.desc_size = sizeof(struct speex_coder_pvt),
|
||||
@@ -446,6 +524,7 @@ static struct ast_translator lin16tospeexwb = {
|
||||
.newpvt = lin16tospeexwb_new,
|
||||
.framein = lintospeex_framein,
|
||||
.frameout = lintospeex_frameout,
|
||||
.feedback = lintospeex_feedback,
|
||||
.destroy = lintospeex_destroy,
|
||||
.sample = slin16_sample,
|
||||
.desc_size = sizeof(struct speex_coder_pvt),
|
||||
@@ -491,6 +570,7 @@ static struct ast_translator lin32tospeexuwb = {
|
||||
.newpvt = lin32tospeexuwb_new,
|
||||
.framein = lintospeex_framein,
|
||||
.frameout = lintospeex_frameout,
|
||||
.feedback = lintospeex_feedback,
|
||||
.destroy = lintospeex_destroy,
|
||||
.desc_size = sizeof(struct speex_coder_pvt),
|
||||
.buffer_samples = BUFFER_SAMPLES,
|
||||
@@ -586,6 +666,9 @@ static int parse_config(int reload)
|
||||
pp_dereverb_level = res_f;
|
||||
} else
|
||||
ast_log(LOG_ERROR,"Error! Preprocessor Dereverb Level must be >= 0\n");
|
||||
} else if (!strcasecmp(var->name, "experimental_rtcp_feedback")) {
|
||||
exp_rtcp_fb = ast_true(var->value) ? 1 : 0;
|
||||
ast_verb(3, "CODEC SPEEX: Experimental RTCP Feedback. [%s]\n",exp_rtcp_fb ? "on" : "off");
|
||||
}
|
||||
}
|
||||
ast_config_destroy(cfg);
|
||||
|
||||
Reference in New Issue
Block a user