smacker: set channel layout

[ffmpeg] / libavformat / rtpenc.c

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index ada5a3b4f35ec123672848f53f417f26be365efd..b17c4651b6bfe74728b58f1d21ae4c9403a040bc 100644 (file)
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -32,8 +32,8 @@
 
 static const AVOption options[] = {
     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 },
+    { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
+    { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
     { NULL },
 };
 
@@ -75,6 +75,9 @@ static int is_supported(enum AVCodecID id)
     case AV_CODEC_ID_ADPCM_G722:
     case AV_CODEC_ID_ADPCM_G726:
     case AV_CODEC_ID_ILBC:
+    case AV_CODEC_ID_MJPEG:
+    case AV_CODEC_ID_SPEEX:
+    case AV_CODEC_ID_OPUS:
         return 1;
     default:
         return 0;
@@ -179,15 +182,21 @@ static int rtp_write_header(AVFormatContext *s1)
         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
         s->num_frames = 0;
         goto defaultcase;
-    case AV_CODEC_ID_VP8:
-        av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
-                                 "incompatible with the latest spec drafts.\n");
-        break;
     case AV_CODEC_ID_ADPCM_G722:
         /* Due to a historical error, the clock rate for G722 in RTP is
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
+    case AV_CODEC_ID_OPUS:
+        if (st->codec->channels > 2) {
+            av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
+            goto fail;
+        }
+        /* The opus RTP RFC says that all opus streams should use 48000 Hz
+         * as clock rate, since all opus sample rates can be expressed in
+         * this clock rate, and sample rate changes on the fly are supported. */
+        avpriv_set_pts_info(st, 32, 1, 48000);
+        break;
     case AV_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");
@@ -281,8 +290,8 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 
 /* send an integer number of samples and compute time stamp and fill
    the rtp send buffer before sending. */
-static void rtp_send_samples(AVFormatContext *s1,
-                             const uint8_t *buf1, int size, int sample_size_bits)
+static int rtp_send_samples(AVFormatContext *s1,
+                            const uint8_t *buf1, int size, int sample_size_bits)
 {
     RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
@@ -292,7 +301,7 @@ static void rtp_send_samples(AVFormatContext *s1,
     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
-        av_abort();
+        return AVERROR(EINVAL);
     n = 0;
     while (size > 0) {
         s->buf_ptr = s->buf;
@@ -307,6 +316,7 @@ static void rtp_send_samples(AVFormatContext *s1,
         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
         n += (s->buf_ptr - s->buf);
     }
+    return 0;
 }
 
 static void rtp_send_mpegaudio(AVFormatContext *s1,
@@ -461,25 +471,21 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case AV_CODEC_ID_PCM_ALAW:
     case AV_CODEC_ID_PCM_U8:
     case AV_CODEC_ID_PCM_S8:
-        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
-        break;
+        return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
     case AV_CODEC_ID_PCM_U16BE:
     case AV_CODEC_ID_PCM_U16LE:
     case AV_CODEC_ID_PCM_S16BE:
     case AV_CODEC_ID_PCM_S16LE:
-        rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
-        break;
+        return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
     case AV_CODEC_ID_ADPCM_G722:
         /* The actual sample size is half a byte per sample, but since the
          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
          * the correct parameter for send_samples_bits is 8 bits per stream
          * clock. */
-        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
-        break;
+        return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
     case AV_CODEC_ID_ADPCM_G726:
-        rtp_send_samples(s1, pkt->data, size,
-                         st->codec->bits_per_coded_sample * st->codec->channels);
-        break;
+        return rtp_send_samples(s1, pkt->data, size,
+                                st->codec->bits_per_coded_sample * st->codec->channels);
     case AV_CODEC_ID_MP2:
     case AV_CODEC_ID_MP3:
         rtp_send_mpegaudio(s1, pkt->data, size);
@@ -527,6 +533,17 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case AV_CODEC_ID_ILBC:
         rtp_send_ilbc(s1, pkt->data, size);
         break;
+    case AV_CODEC_ID_MJPEG:
+        ff_rtp_send_jpeg(s1, pkt->data, size);
+        break;
+    case AV_CODEC_ID_OPUS:
+        if (size > s->max_payload_size) {
+            av_log(s1, AV_LOG_ERROR,
+                   "Packet size %d too large for max RTP payload size %d\n",
+                   size, s->max_payload_size);
+            return AVERROR(EINVAL);
+        }
+        /* Intentional fallthrough */
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);