configure: Drop weak dependencies on external libraries for webm muxer

[ffmpeg] / libavformat / rtspenc.c

diff --git a/libavformat/rtspenc.c b/libavformat/rtspenc.c
index e3631c4290bcae7e0c368226505b1fa0273bb9a3..3db53aca076d1c16d4855545448c4120c87e0930 100644 (file)
--- a/libavformat/rtspenc.c
+++ b/libavformat/rtspenc.c
@@ -21,7 +21,6 @@
 
 #include "avformat.h"
 
-#include <sys/time.h>
 #if HAVE_POLL_H
 #include <poll.h>
 #endif
@@ -29,11 +28,21 @@
 #include "os_support.h"
 #include "rtsp.h"
 #include "internal.h"
+#include "avio_internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/avstring.h"
+#include "libavutil/time.h"
+#include "url.h"
 
 #define SDP_MAX_SIZE 16384
 
+static const AVClass rtsp_muxer_class = {
+    .class_name = "RTSP muxer",
+    .item_name  = av_default_item_name,
+    .option     = ff_rtsp_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
 {
     RTSPState *rt = s->priv_data;
@@ -46,7 +55,7 @@ int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
 
     /* Announce the stream */
     sdp = av_mallocz(SDP_MAX_SIZE);
-    if (sdp == NULL)
+    if (!sdp)
         return AVERROR(ENOMEM);
     /* We create the SDP based on the RTSP AVFormatContext where we
      * aren't allowed to change the filename field. (We create the SDP
@@ -64,7 +73,7 @@ int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
                 "rtsp", NULL, addr, -1, NULL);
     ctx_array[0] = &sdp_ctx;
-    if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
+    if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
         av_free(sdp);
         return AVERROR_INVALIDDATA;
     }
@@ -127,7 +136,7 @@ static int rtsp_write_header(AVFormatContext *s)
     return 0;
 }
 
-static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
+int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
 {
     RTSPState *rt = s->priv_data;
     AVFormatContext *rtpctx = rtsp_st->transport_priv;
@@ -136,13 +145,14 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
     uint8_t *interleave_header, *interleaved_packet;
 
     size = avio_close_dyn_buf(rtpctx->pb, &buf);
+    rtpctx->pb = NULL;
     ptr = buf;
     while (size > 4) {
         uint32_t packet_len = AV_RB32(ptr);
         int id;
         /* The interleaving header is exactly 4 bytes, which happens to be
          * the same size as the packet length header from
-         * url_open_dyn_packet_buf. So by writing the interleaving header
+         * ffio_open_dyn_packet_buf. So by writing the interleaving header
          * over these bytes, we get a consecutive interleaved packet
          * that can be written in one call. */
         interleaved_packet = interleave_header = ptr;
@@ -150,20 +160,19 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
         size -= 4;
         if (packet_len > size || packet_len < 2)
             break;
-        if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
+        if (RTP_PT_IS_RTCP(ptr[1]))
             id = rtsp_st->interleaved_max; /* RTCP */
         else
             id = rtsp_st->interleaved_min; /* RTP */
         interleave_header[0] = '$';
         interleave_header[1] = id;
         AV_WB16(interleave_header + 2, packet_len);
-        url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
+        ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
         ptr += packet_len;
         size -= packet_len;
     }
     av_free(buf);
-    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
-    return 0;
+    return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
 }
 
 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
@@ -171,7 +180,7 @@ static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
     RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;
     int n;
-    struct pollfd p = {url_get_file_handle(rt->rtsp_hd), POLLIN, 0};
+    struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
     AVFormatContext *rtpctx;
     int ret;
 
@@ -208,7 +217,7 @@ static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
      * packets, so we need to send them out on the TCP connection separately.
      */
     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
-        ret = tcp_write_packet(s, rtsp_st);
+        ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
     return ret;
 }
 
@@ -216,6 +225,11 @@ static int rtsp_write_close(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;
 
+    // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
+    // Thus call this on all streams before doing the teardown. This is
+    // done within ff_rtsp_undo_setup.
+    ff_rtsp_undo_setup(s, 1);
+
     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
 
     ff_rtsp_close_streams(s);
@@ -225,16 +239,14 @@ static int rtsp_write_close(AVFormatContext *s)
 }
 
 AVOutputFormat ff_rtsp_muxer = {
-    "rtsp",
-    NULL_IF_CONFIG_SMALL("RTSP output format"),
-    NULL,
-    NULL,
-    sizeof(RTSPState),
-    CODEC_ID_AAC,
-    CODEC_ID_MPEG4,
-    rtsp_write_header,
-    rtsp_write_packet,
-    rtsp_write_close,
-    .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
+    .name              = "rtsp",
+    .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),
+    .priv_data_size    = sizeof(RTSPState),
+    .audio_codec       = AV_CODEC_ID_AAC,
+    .video_codec       = AV_CODEC_ID_MPEG4,
+    .write_header      = rtsp_write_header,
+    .write_packet      = rtsp_write_packet,
+    .write_trailer     = rtsp_write_close,
+    .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
+    .priv_class        = &rtsp_muxer_class,
 };
-