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-rw-r--r--codecs/codec_speex.c83
-rw-r--r--configs/samples/codecs.conf.sample3
-rw-r--r--funcs/func_frame_trace.c3
-rw-r--r--include/asterisk/frame.h2
-rw-r--r--include/asterisk/translate.h10
-rw-r--r--main/channel.c10
-rw-r--r--main/frame.c8
-rw-r--r--main/translate.c11
-rw-r--r--res/res_rtp_asterisk.c23
9 files changed, 151 insertions, 2 deletions
diff --git a/codecs/codec_speex.c b/codecs/codec_speex.c
index 49990e988..72ac22023 100644
--- a/codecs/codec_speex.c
+++ b/codecs/codec_speex.c
@@ -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);
diff --git a/configs/samples/codecs.conf.sample b/configs/samples/codecs.conf.sample
index 63d0352a8..e40aa35a3 100644
--- a/configs/samples/codecs.conf.sample
+++ b/configs/samples/codecs.conf.sample
@@ -57,6 +57,9 @@ pp_dereverb => false
pp_dereverb_decay => 0.4
pp_dereverb_level => 0.3
+; experimental bitrate changes depending on RTCP feedback [true / false]
+experimental_rtcp_feedback => false
+
[plc]
; for all codecs which do not support native PLC
diff --git a/funcs/func_frame_trace.c b/funcs/func_frame_trace.c
index 08c426161..8a0b3dd59 100644
--- a/funcs/func_frame_trace.c
+++ b/funcs/func_frame_trace.c
@@ -370,6 +370,9 @@ static void print_frame(struct ast_frame *frame)
}
ast_verbose("Bytes: %d\n", frame->datalen);
break;
+ case AST_FRAME_RTCP:
+ ast_verbose("FrameType: RTCP\n");
+ break;
case AST_FRAME_NULL:
ast_verbose("FrameType: NULL\n");
break;
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 20f40f863..45bc8fc32 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -127,6 +127,8 @@ enum ast_frame_type {
* 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 8188eb8eb..f0fa83918 100644
--- a/include/asterisk/translate.h
+++ b/include/asterisk/translate.h
@@ -121,7 +121,7 @@ enum ast_trans_cost_table {
*
* 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 @@ struct ast_translator {
/*!< 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). */
@@ -316,7 +320,9 @@ void ast_translator_free_path(struct ast_trans_pvt *tr);
/*!
* \brief translates one or more frames
* Apply an input frame into the translator and receive zero or one output frames. Consume
- * determines whether the original frame should be freed
+ * determines whether the original frame should be freed. In case the frame type is
+ * AST_FRAME_RTCP, the frame is not translated but passed to the translator codecs
+ * via the feedback callback, and a pointer to ast_null_frame is returned after that.
* \param path tr translator structure to use for translation
* \param f frame to translate
* \param consume Whether or not to free the original frame
diff --git a/main/channel.c b/main/channel.c
index 00cfa31aa..68c45a260 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -1531,6 +1531,7 @@ int ast_is_deferrable_frame(const struct ast_frame *frame)
case AST_FRAME_IAX:
case AST_FRAME_CNG:
case AST_FRAME_MODEM:
+ case AST_FRAME_RTCP:
return 0;
}
return 0;
@@ -2866,6 +2867,7 @@ int __ast_answer(struct ast_channel *chan, unsigned int delay)
case AST_FRAME_IMAGE:
case AST_FRAME_HTML:
case AST_FRAME_MODEM:
+ case AST_FRAME_RTCP:
done = 1;
break;
case AST_FRAME_CONTROL:
@@ -4348,6 +4350,14 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
*/
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 0175c7226..71feacb61 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -533,6 +533,8 @@ void ast_frame_subclass2str(struct ast_frame *f, char *subclass, size_t slen, ch
break;
}
break;
+ case AST_FRAME_RTCP:
+ ast_copy_string(subclass, "RTCP", slen);
default:
ast_copy_string(subclass, "Unknown Subclass", slen);
break;
@@ -584,6 +586,9 @@ void ast_frame_type2str(enum ast_frame_type frame_type, char *ftype, size_t len)
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;
@@ -621,6 +626,9 @@ void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix)
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 fa606e71b..168a72a4b 100644
--- a/main/translate.c
+++ b/main/translate.c
@@ -530,6 +530,17 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
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 58c217ecb..91d09b9cf 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4320,6 +4320,29 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
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[report_counter-1], sizeof(struct ast_rtp_rtcp_report_block));
+ rtcp_report2 = (struct ast_rtp_rtcp_report *)rtp->f.data.ptr;
+ rtcp_report2->report_block[report_counter-1] = 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 */
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-27 17:49:58 +0000