tls: cafile, cert, key options

[ffmpeg] / libavformat / rtpenc.c

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index c2736b17289eb2e9669867b4c757feebc4f6500f..0a3071d190aabbc96e0c9a61a4873befe482eacc 100644 (file)
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -31,8 +31,9 @@
 //#define DEBUG
 
 static const AVOption options[] = {
-    FF_RTP_FLAG_OPTS(RTPMuxContext, flags)
+    FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
+    { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
     { NULL },
 };
 
@@ -73,6 +74,7 @@ static int is_supported(enum CodecID id)
     case CODEC_ID_VP8:
     case CODEC_ID_ADPCM_G722:
     case CODEC_ID_ADPCM_G726:
+    case CODEC_ID_ILBC:
         return 1;
     default:
         return 0;
@@ -101,7 +103,8 @@ static int rtp_write_header(AVFormatContext *s1)
     s->base_timestamp = av_get_random_seed();
     s->timestamp = s->base_timestamp;
     s->cur_timestamp = 0;
-    s->ssrc = av_get_random_seed();
+    if (!s->ssrc)
+        s->ssrc = av_get_random_seed();
     s->first_packet = 1;
     s->first_rtcp_ntp_time = ff_ntp_time();
     if (s1->start_time_realtime)
@@ -185,6 +188,16 @@ static int rtp_write_header(AVFormatContext *s1)
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
+    case CODEC_ID_ILBC:
+        if (st->codec->block_align != 38 && st->codec->block_align != 50) {
+            av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
+            goto fail;
+        }
+        if (!s->max_frames_per_packet)
+            s->max_frames_per_packet = 1;
+        s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
+                                         s->max_payload_size / st->codec->block_align);
+        goto defaultcase;
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
         if (!s->max_frames_per_packet)
@@ -196,11 +209,11 @@ static int rtp_write_header(AVFormatContext *s1)
         /* max_header_toc_size + the largest AMR payload must fit */
         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
-            return -1;
+            goto fail;
         }
         if (st->codec->channels != 1) {
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
-            return -1;
+            goto fail;
         }
     case CODEC_ID_AAC:
         s->num_frames = 0;
@@ -214,6 +227,10 @@ defaultcase:
     }
 
     return 0;
+
+fail:
+    av_freep(&s->buf);
+    return AVERROR(EINVAL);
 }
 
 /* send an rtcp sender report packet */
@@ -389,6 +406,36 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
     }
 }
 
+static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
+{
+    RTPMuxContext *s = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int frame_duration = av_get_audio_frame_duration(st->codec, 0);
+    int frame_size = st->codec->block_align;
+    int frames = size / frame_size;
+
+    while (frames > 0) {
+        int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
+
+        if (!s->num_frames) {
+            s->buf_ptr = s->buf;
+            s->timestamp = s->cur_timestamp;
+        }
+        memcpy(s->buf_ptr, buf, n * frame_size);
+        frames           -= n;
+        s->num_frames    += n;
+        s->buf_ptr       += n * frame_size;
+        buf              += n * frame_size;
+        s->cur_timestamp += n * frame_duration;
+
+        if (s->num_frames == s->max_frames_per_packet) {
+            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+            s->num_frames = 0;
+        }
+    }
+    return 0;
+}
+
 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     RTPMuxContext *s = s1->priv_data;
@@ -477,6 +524,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case CODEC_ID_VP8:
         ff_rtp_send_vp8(s1, pkt->data, size);
         break;
+    case CODEC_ID_ILBC:
+        rtp_send_ilbc(s1, pkt->data, size);
+        break;
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);