[mod_opus] set RTP interval for RFC 7587 compliance

Configure proper RTP timestamp increments for Opus codec to ensure
compliance with RFC 7587. This fixes timestamp handling when
transcoding between different sample rates.

Author: ticpu

src/mod/codecs/mod_opus/mod_opus.c

@@ -144,6 +144,7 @@ struct opus_context {
 	enc_stats_t encoder_stats;
 	codec_control_state_t control_state;
 	switch_bool_t recreate_decoder;
+	uint32_t encoder_samplerate;
 };
 
 struct {
@@ -177,6 +178,11 @@ static switch_bool_t switch_opus_acceptable_rate(int rate)
 	return SWITCH_TRUE;
 }
 
+static inline uint32_t opus_ts_step(uint32_t ptime_ms, uint32_t enc_rate)
+{
+	return (ptime_ms * 48); /* 48 kHz RTP clock rate */
+}
+
 static uint32_t switch_opus_encoder_set_audio_bandwidth(OpusEncoder *encoder_object,int enc_samplerate)
 {
 		if (enc_samplerate == 8000) { /* Audio Bandwidth: 0-4000Hz  Sampling Rate: 8000Hz */
@@ -618,6 +624,9 @@ static switch_status_t switch_opus_init(switch_codec_t *codec, switch_codec_flag
 			return SWITCH_STATUS_GENERR;
 		}
 
+		/* Store encoder sample rate for RTP timestamp scaling */
+		context->encoder_samplerate = enc_samplerate;
+
 		/* https://tools.ietf.org/html/rfc7587  */
 		if (opus_codec_settings.maxaveragebitrate) {
 			opus_encoder_ctl(context->encoder_object, OPUS_SET_BITRATE(opus_codec_settings.maxaveragebitrate));
@@ -680,6 +689,7 @@ static switch_status_t switch_opus_init(switch_codec_t *codec, switch_codec_flag
 		if (opus_prefs.adjust_bitrate) {
 			switch_set_flag(codec, SWITCH_CODEC_FLAG_HAS_ADJ_BITRATE);
 		}
+
 	}
 
 	if (decoding) {
@@ -718,6 +728,19 @@ static switch_status_t switch_opus_init(switch_codec_t *codec, switch_codec_flag
 		}
 	}
 
+	/* Set RTP interval for RFC 7587 compliance */
+	if (codec->session) {
+		switch_rtp_t *rtp = switch_core_media_get_rtp_session(codec->session, SWITCH_MEDIA_TYPE_AUDIO);
+		if (rtp) {
+			uint32_t ptime_ms = codec->implementation->microseconds_per_packet / 1000;
+			uint32_t step = opus_ts_step(ptime_ms, codec->implementation->actual_samples_per_second);
+			switch_rtp_set_interval(rtp, ptime_ms, step);
+		}
+	} else {
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING,
+						 "Opus codec initialized without session - RTP interval not configured\n");
+	}
+
 	context->codec_settings = opus_codec_settings;
 	codec->private_info = context;