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rtpenc: MP4A-LATM payload support

This commit is contained in:
Juan Carlos Rodriguez 2011-05-18 16:21:48 +03:00 committed by Michael Niedermayer
parent bd61b2a1ca
commit ef409645f0
7 changed files with 142 additions and 1 deletions

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@ -233,6 +233,7 @@ OBJS-$(CONFIG_RSO_MUXER) += rsoenc.o rso.o
OBJS-$(CONFIG_RPL_DEMUXER) += rpl.o OBJS-$(CONFIG_RPL_DEMUXER) += rpl.o
OBJS-$(CONFIG_RTP_MUXER) += rtp.o \ OBJS-$(CONFIG_RTP_MUXER) += rtp.o \
rtpenc_aac.o \ rtpenc_aac.o \
rtpenc_latm.o \
rtpenc_amr.o \ rtpenc_amr.o \
rtpenc_h263.o \ rtpenc_h263.o \
rtpenc_mpv.o \ rtpenc_mpv.o \

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@ -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_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_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_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_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) #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)

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@ -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"}, {"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"}, {"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"}, {"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}, {"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}, {"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}, {"indexmem", "max memory used for timestamp index (per stream)", OFFSET(max_index_size), FF_OPT_TYPE_INT, {.dbl = 1<<20 }, 0, INT_MAX, D},

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@ -404,7 +404,10 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
ff_rtp_send_mpegvideo(s1, pkt->data, size); ff_rtp_send_mpegvideo(s1, pkt->data, size);
break; break;
case CODEC_ID_AAC: 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; break;
case CODEC_ID_AMR_NB: case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB: case CODEC_ID_AMR_WB:

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@ -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_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_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_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_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_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size);

60
libavformat/rtpenc_latm.c Normal file
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@ -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;
}
}

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@ -23,6 +23,7 @@
#include "libavutil/base64.h" #include "libavutil/base64.h"
#include "libavutil/parseutils.h" #include "libavutil/parseutils.h"
#include "libavcodec/xiph.h" #include "libavcodec/xiph.h"
#include "libavcodec/mpeg4audio.h"
#include "avformat.h" #include "avformat.h"
#include "internal.h" #include "internal.h"
#include "avc.h" #include "avc.h"
@ -299,6 +300,69 @@ xiph_fail:
return NULL; 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) static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c, int payload_type, int flags)
{ {
char *config = NULL; char *config = NULL;
@ -334,6 +398,15 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
payload_type, config ? config : ""); payload_type, config ? config : "");
break; break;
case CODEC_ID_AAC: 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) { if (c->extradata_size) {
config = extradata2config(c); config = extradata2config(c);
} else { } else {
@ -352,6 +425,7 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
"indexdeltalength=3%s\r\n", "indexdeltalength=3%s\r\n",
payload_type, c->sample_rate, c->channels, payload_type, c->sample_rate, c->channels,
payload_type, config); payload_type, config);
}
break; break;
case CODEC_ID_PCM_S16BE: case CODEC_ID_PCM_S16BE:
if (payload_type >= RTP_PT_PRIVATE) if (payload_type >= RTP_PT_PRIVATE)