avutil/hwcontext_vulkan: fix format specifiers for some printed variables

[ffmpeg] / libavformat / rtpdec.c

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 51cf0fde07a1d6f98cec7bf24eed7ccd1a095a07..6b0da9e63695261b193b3e9ca98d46c8c9b568b0 100644 (file)
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -36,48 +36,48 @@
 
 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
 
-static RTPDynamicProtocolHandler l24_dynamic_handler = {
+static const RTPDynamicProtocolHandler l24_dynamic_handler = {
     .enc_name   = "L24",
     .codec_type = AVMEDIA_TYPE_AUDIO,
     .codec_id   = AV_CODEC_ID_PCM_S24BE,
 };
 
-static RTPDynamicProtocolHandler gsm_dynamic_handler = {
+static const RTPDynamicProtocolHandler gsm_dynamic_handler = {
     .enc_name   = "GSM",
     .codec_type = AVMEDIA_TYPE_AUDIO,
     .codec_id   = AV_CODEC_ID_GSM,
 };
 
-static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
+static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
     .enc_name   = "X-MP3-draft-00",
     .codec_type = AVMEDIA_TYPE_AUDIO,
     .codec_id   = AV_CODEC_ID_MP3ADU,
 };
 
-static RTPDynamicProtocolHandler speex_dynamic_handler = {
+static const RTPDynamicProtocolHandler speex_dynamic_handler = {
     .enc_name   = "speex",
     .codec_type = AVMEDIA_TYPE_AUDIO,
     .codec_id   = AV_CODEC_ID_SPEEX,
 };
 
-static RTPDynamicProtocolHandler opus_dynamic_handler = {
+static const RTPDynamicProtocolHandler opus_dynamic_handler = {
     .enc_name   = "opus",
     .codec_type = AVMEDIA_TYPE_AUDIO,
     .codec_id   = AV_CODEC_ID_OPUS,
 };
 
-static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
+static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
     .enc_name   = "t140",
     .codec_type = AVMEDIA_TYPE_SUBTITLE,
     .codec_id   = AV_CODEC_ID_TEXT,
 };
 
-extern RTPDynamicProtocolHandler ff_rdt_video_handler;
-extern RTPDynamicProtocolHandler ff_rdt_audio_handler;
-extern RTPDynamicProtocolHandler ff_rdt_live_video_handler;
-extern RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_video_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_audio_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
 
-static const RTPDynamicProtocolHandler *rtp_dynamic_protocol_handler_list[] = {
+static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = {
     /* rtp */
     &ff_ac3_dynamic_handler,
     &ff_amr_nb_dynamic_handler,
@@ -404,38 +404,26 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
 
 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
 {
-    AVIOContext *pb;
-    uint8_t *buf;
-    int len;
+    uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf;
 
     /* Send a small RTP packet */
-    if (avio_open_dyn_buf(&pb) < 0)
-        return;
 
-    avio_w8(pb, (RTP_VERSION << 6));
-    avio_w8(pb, 0); /* Payload type */
-    avio_wb16(pb, 0); /* Seq */
-    avio_wb32(pb, 0); /* Timestamp */
-    avio_wb32(pb, 0); /* SSRC */
+    bytestream_put_byte(&ptr, (RTP_VERSION << 6));
+    bytestream_put_byte(&ptr, 0); /* Payload type */
+    bytestream_put_be16(&ptr, 0); /* Seq */
+    bytestream_put_be32(&ptr, 0); /* Timestamp */
+    bytestream_put_be32(&ptr, 0); /* SSRC */
 
-    len = avio_close_dyn_buf(pb, &buf);
-    if ((len > 0) && buf)
-        ffurl_write(rtp_handle, buf, len);
-    av_free(buf);
+    ffurl_write(rtp_handle, buf, ptr - buf);
 
     /* Send a minimal RTCP RR */
-    if (avio_open_dyn_buf(&pb) < 0)
-        return;
-
-    avio_w8(pb, (RTP_VERSION << 6));
-    avio_w8(pb, RTCP_RR); /* receiver report */
-    avio_wb16(pb, 1); /* length in words - 1 */
-    avio_wb32(pb, 0); /* our own SSRC */
+    ptr = buf;
+    bytestream_put_byte(&ptr, (RTP_VERSION << 6));
+    bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */
+    bytestream_put_be16(&ptr, 1); /* length in words - 1 */
+    bytestream_put_be32(&ptr, 0); /* our own SSRC */
 
-    len = avio_close_dyn_buf(pb, &buf);
-    if ((len > 0) && buf)
-        ffurl_write(rtp_handle, buf, len);
-    av_free(buf);
+    ffurl_write(rtp_handle, buf, ptr - buf);
 }
 
 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
@@ -634,6 +622,24 @@ void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
         s->srtp_enabled = 1;
 }
 
+static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
+    int64_t rtcp_time, delta_timestamp, delta_time;
+
+    AVProducerReferenceTime *prft =
+        (AVProducerReferenceTime *) av_packet_new_side_data(
+            pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
+    if (!prft)
+        return AVERROR(ENOMEM);
+
+    rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US;
+    delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
+    delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
+
+    prft->wallclock = rtcp_time + delta_time;
+    prft->flags = 24;
+    return 0;
+}
+
 /**
  * This was the second switch in rtp_parse packet.
  * Normalizes time, if required, sets stream_index, etc.
@@ -645,6 +651,12 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
     if (timestamp == RTP_NOTS_VALUE)
         return;
 
+    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+        if (rtp_set_prft(s, pkt, timestamp) < 0) {
+            av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
+        }
+    }
+
     if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
         int64_t addend;
         int delta_timestamp;
@@ -976,7 +988,7 @@ int ff_parse_fmtp(AVFormatContext *s,
 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
 {
     int ret;
-    av_init_packet(pkt);
+    av_packet_unref(pkt);
 
     pkt->size         = avio_close_dyn_buf(*dyn_buf, &pkt->data);
     pkt->stream_index = stream_idx;