[Asterisk-code-review] Experimental patch to add RTCP feedback to codec modules (asterisk[14])
Lorenzo Miniero
asteriskteam at digium.com
Tue Nov 29 09:58:27 CST 2016
Lorenzo Miniero has uploaded a new change for review. ( https://gerrit.asterisk.org/4520 )
Change subject: Experimental patch to add RTCP feedback to codec modules
......................................................................
Experimental patch to add RTCP feedback to codec modules
Change-Id: Ie74d254c904faed9173e4379ecce4267bcdb3099
Fixed ISO C90 warnings
Change-Id: I4e7e9ee4e4d156dc32732695e9c45a5d4ce6a0bb
---
M codecs/codec_speex.c
M include/asterisk/frame.h
M include/asterisk/translate.h
M main/channel.c
M main/frame.c
M main/translate.c
M res/res_rtp_asterisk.c
7 files changed, 122 insertions(+), 1 deletion(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/20/4520/1
diff --git a/codecs/codec_speex.c b/codecs/codec_speex.c
index ca48eae..979b5ff 100644
--- a/codecs/codec_speex.c
+++ b/codecs/codec_speex.c
@@ -57,6 +57,9 @@
#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;
@@ -93,6 +96,10 @@
SpeexBits bits;
int framesize;
int silent_state;
+
+ int fraction_lost;
+ int quality, default_quality;
+
#ifdef _SPEEX_TYPES_H
SpeexPreprocessState *pp;
spx_int16_t buf[BUFFER_SAMPLES];
@@ -138,6 +145,11 @@
if (dtx)
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_verb(3, "Default quality (%s): %d\n", vbr ? "vbr" : "cbr", tmp->default_quality);
return 0;
}
@@ -344,6 +356,55 @@
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 *)feedback->data.ptr;
+ if(rtcp_report->reception_report_count > 0) {
+ struct ast_rtp_rtcp_report_block *report_block = rtcp_report->report_block[0];
+ int fraction_lost = report_block->lost_count.fraction;
+ if(fraction_lost != tmp->fraction_lost) {
+ int percent = (fraction_lost*100)/256;
+ int bitrate = 0;
+ int q = -1;
+ ast_verb(3, "Fraction lost changed: %d --> %d percent loss\n", fraction_lost, percent);
+ /* Handle change */
+ speex_encoder_ctl(tmp->speex, SPEEX_GET_BITRATE, &bitrate);
+ ast_verb(3, "Current bitrate: %d\n", bitrate);
+ ast_verb(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_verb(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;
@@ -402,6 +463,7 @@
.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),
@@ -448,6 +510,7 @@
.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),
@@ -493,6 +556,7 @@
.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,
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 20f40f8..45bc8fc 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -127,6 +127,8 @@
* directly into bridges.
*/
AST_FRAME_BRIDGE_ACTION_SYNC,
+ /*! RTCP feedback */
+ AST_FRAME_RTCP,
};
#define AST_FRAME_DTMF AST_FRAME_DTMF_END
diff --git a/include/asterisk/translate.h b/include/asterisk/translate.h
index b8cd219..18e34ea 100644
--- a/include/asterisk/translate.h
+++ b/include/asterisk/translate.h
@@ -121,7 +121,7 @@
*
* As a minimum, a translator should supply name, srcfmt and dstfmt,
* the required buf_size (in bytes) and buffer_samples (in samples),
- * and a few callbacks (framein, frameout, sample).
+ * and a few callbacks (framein, frameout, feedback, sample).
* The outbuf is automatically prepended by AST_FRIENDLY_OFFSET
* spare bytes so generic routines can place data in there.
*
@@ -159,6 +159,10 @@
/*!< Output frame callback. Generate a frame
* with outbuf content. */
+ void (*feedback)(struct ast_trans_pvt *pvt, struct ast_frame *feedback);
+ /*!< Feedback frame callback. Handle
+ * input frame. */
+
void (*destroy)(struct ast_trans_pvt *pvt);
/*!< cleanup private data, if needed
* (often unnecessary). */
diff --git a/main/channel.c b/main/channel.c
index b4d451a..9283e95 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -1534,6 +1534,7 @@
case AST_FRAME_IAX:
case AST_FRAME_CNG:
case AST_FRAME_MODEM:
+ case AST_FRAME_RTCP:
return 0;
}
return 0;
@@ -2869,6 +2870,7 @@
case AST_FRAME_IMAGE:
case AST_FRAME_HTML:
case AST_FRAME_MODEM:
+ case AST_FRAME_RTCP:
done = 1;
break;
case AST_FRAME_CONTROL:
@@ -4351,6 +4353,13 @@
*/
ast_read_generator_actions(chan, f);
break;
+ case AST_FRAME_RTCP:
+ /* Incoming RTCP feedback needs to get to the translator for
+ * outgoing media, which means we treat it as an ast_write */
+ if(ast_channel_writetrans(chan))
+ ast_translate(ast_channel_writetrans(chan), f, 0);
+ ast_frfree(f);
+ f = &ast_null_frame;
default:
/* Just pass it on! */
break;
diff --git a/main/frame.c b/main/frame.c
index 92b92b6..834d81f 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -535,6 +535,8 @@
break;
}
break;
+ case AST_FRAME_RTCP:
+ ast_copy_string(subclass, "RTCP", slen);
default:
ast_copy_string(subclass, "Unknown Subclass", slen);
break;
@@ -586,6 +588,9 @@
case AST_FRAME_VIDEO:
ast_copy_string(ftype, "Video", len);
break;
+ case AST_FRAME_RTCP:
+ ast_copy_string(ftype, "RTCP", len);
+ break;
default:
snprintf(ftype, len, "Unknown Frametype '%u'", frame_type);
break;
@@ -623,6 +628,9 @@
if (f->frametype == AST_FRAME_VIDEO) {
return;
}
+ if (f->frametype == AST_FRAME_RTCP) {
+ return;
+ }
ast_frame_type2str(f->frametype, ftype, sizeof(ftype));
ast_frame_subclass2str(f, subclass, sizeof(subclass), moreinfo, sizeof(moreinfo));
diff --git a/main/translate.c b/main/translate.c
index 43e6e29..ae8c0ab 100644
--- a/main/translate.c
+++ b/main/translate.c
@@ -532,6 +532,17 @@
long len;
int seqno;
+ if(f->frametype == AST_FRAME_RTCP) {
+ /* Just pass the feedback to the right callback, if it exists.
+ * This "translation" does nothing so return a null frame. */
+ struct ast_trans_pvt *tp;
+ for(tp = p; tp; tp = tp->next) {
+ if(tp->t->feedback)
+ tp->t->feedback(tp, f);
+ }
+ return &ast_null_frame;
+ }
+
has_timing_info = ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO);
ts = f->ts;
len = f->len;
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 588247b..f639fa6 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4321,6 +4321,29 @@
rtcp_report,
message_blob);
ast_json_unref(message_blob);
+
+ /* Return an AST_FRAME_RTCP frame with the ast_rtp_rtcp_report
+ * object as a its data */
+ rtp->f.frametype = AST_FRAME_RTCP;
+ rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
+ memcpy(rtp->f.data.ptr, rtcp_report, sizeof(struct ast_rtp_rtcp_report));
+ rtp->f.datalen = sizeof(struct ast_rtp_rtcp_report);
+ if(rc > 0) {
+ /* There's always a single report block stored, here */
+ struct ast_rtp_rtcp_report *rtcp_report2;
+ report_block = rtp->f.data.ptr + rtp->f.datalen + sizeof(struct ast_rtp_rtcp_report_block *);
+ memcpy(report_block, rtcp_report->report_block[0], sizeof(struct ast_rtp_rtcp_report_block));
+ rtcp_report2 = (struct ast_rtp_rtcp_report *)rtp->f.data.ptr;
+ rtcp_report2->report_block[0] = report_block;
+ rtp->f.datalen += sizeof(struct ast_rtp_rtcp_report_block);
+ }
+ rtp->f.offset = AST_FRIENDLY_OFFSET;
+ rtp->f.samples = 0;
+ rtp->f.mallocd = 0;
+ rtp->f.delivery.tv_sec = 0;
+ rtp->f.delivery.tv_usec = 0;
+ rtp->f.src = "RTP";
+ f = &rtp->f;
break;
case RTCP_PT_FUR:
/* Handle RTCP FIR as FUR */
--
To view, visit https://gerrit.asterisk.org/4520
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I4e7e9ee4e4d156dc32732695e9c45a5d4ce6a0bb
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Lorenzo Miniero <lminiero at gmail.com>
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