rtpenc: MP4A-LATM payload support

7 files changed, 142 insertions, 1 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 13fe237..55f6152 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -233,6 +233,7 @@ OBJS-$(CONFIG_RSO_MUXER)                 += rsoenc.o rso.o
 OBJS-$(CONFIG_RPL_DEMUXER)               += rpl.o
 OBJS-$(CONFIG_RTP_MUXER)                 += rtp.o         \
                                             rtpenc_aac.o     \
+                                            rtpenc_latm.o    \
                                             rtpenc_amr.o     \
                                             rtpenc_h263.o    \
                                             rtpenc_mpv.o     \
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index ec51a57..f9091f0 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -729,6 +729,7 @@ typedef struct AVFormatContext {
 #define AVFMT_FLAG_NOFILLIN     0x0010 ///< Do not infer any values from other values, just return what is stored in the container
 #define AVFMT_FLAG_NOPARSE      0x0020 ///< Do not use AVParsers, you also must set AVFMT_FLAG_NOFILLIN as the fillin code works on frames and no parsing -> no frames. Also seeking to frames can not work if parsing to find frame boundaries has been disabled
 #define AVFMT_FLAG_RTP_HINT     0x0040 ///< Add RTP hinting to the output file
+#define AVFMT_FLAG_MP4A_LATM    0x0080 ///< Enable RTP MP4A-LATM payload
 #define AVFMT_FLAG_SORT_DTS    0x10000 ///< try to interleave outputted packets by dts (using this flag can slow demuxing down)
 #define AVFMT_FLAG_PRIV_OPT    0x20000 ///< Enable use of private options by delaying codec open (this could be made default once all code is converted)
 
diff --git a/libavformat/options.c b/libavformat/options.c
index 40fd49f..82be848 100644
--- a/libavformat/options.c
+++ b/libavformat/options.c
@@ -51,6 +51,7 @@ static const AVOption options[]={
 {"igndts", "ignore dts", 0, FF_OPT_TYPE_CONST, {.dbl = AVFMT_FLAG_IGNDTS }, INT_MIN, INT_MAX, D, "fflags"},
 {"rtphint", "add rtp hinting", 0, FF_OPT_TYPE_CONST, {.dbl = AVFMT_FLAG_RTP_HINT }, INT_MIN, INT_MAX, E, "fflags"},
 {"sortdts", "try to interleave outputted packets by dts", 0, FF_OPT_TYPE_CONST, {.dbl = AVFMT_FLAG_SORT_DTS }, INT_MIN, INT_MAX, D, "fflags"},
+{"latm", "enable RTP MP4A-LATM payload", 0, FF_OPT_TYPE_CONST, {.dbl = AVFMT_FLAG_MP4A_LATM }, INT_MIN, INT_MAX, E, "fflags"},
 {"analyzeduration", "how many microseconds are analyzed to estimate duration", OFFSET(max_analyze_duration), FF_OPT_TYPE_INT, {.dbl = 5*AV_TIME_BASE }, 0, INT_MAX, D},
 {"cryptokey", "decryption key", OFFSET(key), FF_OPT_TYPE_BINARY, {.dbl = 0}, 0, 0, D},
 {"indexmem", "max memory used for timestamp index (per stream)", OFFSET(max_index_size), FF_OPT_TYPE_INT, {.dbl = 1<<20 }, 0, INT_MAX, D},
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 71ccdab..7b2e78e 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -404,7 +404,10 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
         ff_rtp_send_mpegvideo(s1, pkt->data, size);
         break;
     case CODEC_ID_AAC:
-        ff_rtp_send_aac(s1, pkt->data, size);
+        if (s1->flags & AVFMT_FLAG_MP4A_LATM)
+            ff_rtp_send_latm(s1, pkt->data, size);
+        else
+            ff_rtp_send_aac(s1, pkt->data, size);
         break;
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
diff --git a/libavformat/rtpenc.h b/libavformat/rtpenc.h
index b9663c5..d65214a 100644
--- a/libavformat/rtpenc.h
+++ b/libavformat/rtpenc.h
@@ -65,6 +65,7 @@ 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_latm(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);
 void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size);
diff --git a/libavformat/rtpenc_latm.c b/libavformat/rtpenc_latm.c
new file mode 100644
index 0000000..501fa5d
--- /dev/null
+++ b/libavformat/rtpenc_latm.c
@@ -0,0 +1,60 @@
+/*
+ * RTP Packetization of MPEG-4 Audio (RFC 3016)
+ * Copyright (c) 2011 Juan Carlos Rodriguez <ing.juancarlosrodriguez@hotmail.com>
+ *
+ * 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"
+
+void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size) {
+    /* MP4A-LATM
+     * The RTP payload format specification is described in RFC 3016
+     * The encoding specifications are provided in ISO/IEC 14496-3 */
+
+    RTPMuxContext *s = s1->priv_data;
+    int header_size;
+    int offset = 0;
+    int len    = 0;
+
+    /* skip ADTS header, if present */
+    if ((s1->streams[0]->codec->extradata_size) == 0) {
+        size -= 7;
+        buff += 7;
+    }
+
+    /* PayloadLengthInfo() */
+    header_size = size/0xFF + 1;
+    memset(s->buf, 0xFF, header_size - 1);
+    s->buf[header_size - 1] = size % 0xFF;
+
+    s->timestamp = s->cur_timestamp;
+
+    /* PayloadMux() */
+    while (size > 0) {
+        len   = FFMIN(size, s->max_payload_size - (!offset ? header_size : 0));
+        size -= len;
+        if (!offset) {
+            memcpy(s->buf + header_size, buff, len);
+            ff_rtp_send_data(s1, s->buf, header_size + len, !size);
+        } else {
+            ff_rtp_send_data(s1, buff + offset, len, !size);
+        }
+        offset += len;
+    }
+}
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index f7aec1b..c62e00d 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -23,6 +23,7 @@
 #include "libavutil/base64.h"
 #include "libavutil/parseutils.h"
 #include "libavcodec/xiph.h"
+#include "libavcodec/mpeg4audio.h"
 #include "avformat.h"
 #include "internal.h"
 #include "avc.h"
@@ -299,6 +300,69 @@ xiph_fail:
     return NULL;
 }
 
+static int latm_context2profilelevel(AVCodecContext *c) {
+    /* MP4A-LATM
+     * The RTP payload format specification is described in RFC 3016
+     * The encoding specifications are provided in ISO/IEC 14496-3 */
+
+    int profile_level = 0x2B;
+
+    /* TODO: AAC Profile only supports AAC LC Object Type.
+     * Different Object Types should implement different Profile Levels */
+
+    if (c->sample_rate <= 24000) {
+        if (c->channels <= 2)
+            profile_level = 0x28; // AAC Profile, Level 1
+    } else if (c->sample_rate <= 48000) {
+        if (c->channels <= 2) {
+            profile_level = 0x29; // AAC Profile, Level 2
+        } else if (c->channels <= 5) {
+            profile_level = 0x2A; // AAC Profile, Level 4
+        }
+    } else if (c->sample_rate <= 96000) {
+        if (c->channels <= 5) {
+            profile_level = 0x2B; // AAC Profile, Level 5
+        }
+    }
+
+    return profile_level;
+}
+
+static char *latm_context2config(AVCodecContext *c) {
+    /* MP4A-LATM
+     * The RTP payload format specification is described in RFC 3016
+     * The encoding specifications are provided in ISO/IEC 14496-3 */
+
+    uint8_t config_byte[6];
+    int rate_index;
+    char *config;
+
+    for (rate_index = 0; rate_index < 16; rate_index++)
+        if (ff_mpeg4audio_sample_rates[rate_index] == c->sample_rate)
+            break;
+    if (rate_index == 16) {
+        av_log(c, AV_LOG_ERROR, "Unsupported sample rate\n");
+        return NULL;
+    }
+
+    config_byte[0] = 0x40;
+    config_byte[1] = 0;
+    config_byte[2] = 0x20 | rate_index;
+    config_byte[3] = c->channels << 4;
+    config_byte[4] = 0x3f;
+    config_byte[5] = 0xc0;
+
+    config = av_malloc(6*2+1);
+    if (!config) {
+        av_log(c, AV_LOG_ERROR, "Cannot allocate memory for the config info.\n");
+        return NULL;
+    }
+    ff_data_to_hex(config, config_byte, 6, 1);
+    config[12] = 0;
+
+    return config;
+}
+
 static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c, int payload_type, int flags)
 {
     char *config = NULL;
@@ -334,6 +398,15 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
                                      payload_type, config ? config : "");
             break;
         case CODEC_ID_AAC:
+            if (flags & AVFMT_FLAG_MP4A_LATM) {
+                config = latm_context2config(c);
+                if (!config)
+                    return NULL;
+                av_strlcatf(buff, size, "a=rtpmap:%d MP4A-LATM/%d/%d\r\n"
+                                        "a=fmtp:%d profile-level-id=%d;cpresent=0;config=%s\r\n",
+                                         payload_type, c->sample_rate, c->channels,
+                                         payload_type, latm_context2profilelevel(c), config);
+            } else {
             if (c->extradata_size) {
                 config = extradata2config(c);
             } else {
@@ -352,6 +425,7 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
                                     "indexdeltalength=3%s\r\n",
                                      payload_type, c->sample_rate, c->channels,
                                      payload_type, config);
+            }
             break;
         case CODEC_ID_PCM_S16BE:
             if (payload_type >= RTP_PT_PRIVATE)