diff --git a/libavformat/Makefile b/libavformat/Makefile index 13fe2371bf..55f6152f8d 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 ec51a57ca8..f9091f0afd 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 40fd49ff8b..82be8487eb 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 71ccdabf4a..7b2e78e88e 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 b9663c55b0..d65214aeb0 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 0000000000..501fa5d5d5 --- /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 + * + * 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 f7aec1b766..c62e00d775 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)