@@ -7,6 +7,7 @@ version <next>:

 - Alpha channel scaler

 - PCX encoder

 - RTP packetization of H.263

+- RTP packetization of AMR

@@ -188,6 +188,7 @@ OBJS-$(CONFIG_ROQ_MUXER)                 += raw.o

 OBJS-$(CONFIG_RPL_DEMUXER)               += rpl.o

 OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o         \

                                             rtp_aac.o     \

+                                            rtp_amr.o     \

                                             rtp_asf.o     \

                                             rtp_h263.o    \

                                             rtp_mpv.o     \

new file mode 100644

@@ -0,0 +1,66 @@

+/*

+ * RTP packetization for AMR audio

+ * Copyright (c) 2007 Luca Abeni

+ * Copyright (c) 2009 Martin Storsjo

+ *

+ * This file is part of FFmpeg.

+ *

+ * FFmpeg is free software; you can redistribute it and/or

+ * modify it under the terms of the GNU Lesser General Public

+ * License as published by the Free Software Foundation; either

+ * version 2.1 of the License, or (at your option) any later version.

+ *

+ * FFmpeg is distributed in the hope that it will be useful,

+ * but WITHOUT ANY WARRANTY; without even the implied warranty of

+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

+ * Lesser General Public License for more details.

+ *

+ * You should have received a copy of the GNU Lesser General Public

+ * License along with FFmpeg; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include "avformat.h"

+#include "rtpenc.h"

+

+/**

+ * Packetize AMR frames into RTP packets according to RFC 3267,

+ * in octet-aligned mode.

+ */

+void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)

+{

+    RTPMuxContext *s          = s1->priv_data;

+    int max_header_toc_size   = 1 + s->max_frames_per_packet;

+    uint8_t *p;

+    int len;

+

+    /* Test if the packet must be sent. */

+    len = s->buf_ptr - s->buf;

+    if (s->num_frames == s->max_frames_per_packet || (len && len + size - 1 > s->max_payload_size)) {

+        int header_size = s->num_frames + 1;

+        p = s->buf + max_header_toc_size - header_size;

+        if (p != s->buf)

+            memmove(p, s->buf, header_size);

+

+        ff_rtp_send_data(s1, p, s->buf_ptr - p, 1);

+

+        s->num_frames = 0;

+    }

+

+    if (!s->num_frames) {

+        s->buf[0]    = 0xf0;

+        s->buf_ptr   = s->buf + max_header_toc_size;

+        s->timestamp = s->cur_timestamp;

+    } else {

+        /* Mark the previous TOC entry as having more entries following. */

+        s->buf[1 + s->num_frames - 1] |= 0x80;

+    }

+

+    /* Copy the frame type and quality bits. */

+    s->buf[1 + s->num_frames++] = buff[0] & 0x7C;

+    buff++;

+    size--;

+    memcpy(s->buf_ptr, buff, size);

+    s->buf_ptr += size;

+}

+

@@ -60,6 +60,8 @@ static int is_supported(enum CodecID id)

     case CODEC_ID_PCM_U16LE:

     case CODEC_ID_PCM_U8:

     case CODEC_ID_MPEG2TS:

+    case CODEC_ID_AMR_NB:

+    case CODEC_ID_AMR_WB:

         return 1;

     default:

         return 0;

@@ -134,6 +136,23 @@ static int rtp_write_header(AVFormatContext *s1)

         s->max_payload_size = n * TS_PACKET_SIZE;

         s->buf_ptr = s->buf;

         break;

+    case CODEC_ID_AMR_NB:

+    case CODEC_ID_AMR_WB:

+        if (!s->max_frames_per_packet)

+            s->max_frames_per_packet = 12;

+        if (st->codec->codec_id == CODEC_ID_AMR_NB)

+            n = 31;

+        else

+            n = 61;

+        /* 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;

+        }

+        if (st->codec->channels != 1) {

+            av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");

+            return -1;

+        }

     case CODEC_ID_AAC:

         s->num_frames = 0;

     default:

@@ -366,6 +385,10 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)

     case CODEC_ID_AAC:

         ff_rtp_send_aac(s1, buf1, size);

         break;

+    case CODEC_ID_AMR_NB:

+    case CODEC_ID_AMR_WB:

+        ff_rtp_send_amr(s1, buf1, size);

+        break;

     case CODEC_ID_MPEG2TS:

         rtp_send_mpegts_raw(s1, buf1, size);

         break;

@@ -59,6 +59,7 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);

 void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size);

 void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size);

 void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size);

+void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size);

 void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);

 #endif /* AVFORMAT_RTPENC_H */

@@ -228,6 +228,18 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,

                                          payload_type,

                                          c->sample_rate, c->channels);

             break;

+        case CODEC_ID_AMR_NB:

+            av_strlcatf(buff, size, "a=rtpmap:%d AMR/%d/%d\r\n"

+                                    "a=fmtp:%d octet-align=1\r\n",

+                                     payload_type, c->sample_rate, c->channels,

+                                     payload_type);

+            break;

+        case CODEC_ID_AMR_WB:

+            av_strlcatf(buff, size, "a=rtpmap:%d AMR-WB/%d/%d\r\n"

+                                    "a=fmtp:%d octet-align=1\r\n",

+                                     payload_type, c->sample_rate, c->channels,

+                                     payload_type);

+            break;

         default:

             /* Nothing special to do here... */

             break;