[FFmpeg-devel] [PATCH 2/3] libavformat/hls: add support for decryption of HLS streams in MPEG-TS format protected using SAMPLE-AES encryption

Steven Liu lq at chinaffmpeg.org
Mon Mar 1 06:42:42 EET 2021



> 2021年3月1日 下午12:35,Nachiket Tarate <nachiket.programmer at gmail.com> 写道:
> 
> @Steven Liu <lq at chinaffmpeg.org>
> 
> Can we merge this patch ?
I’m waiting update patch for fragment mp4 encryption.
After new version of the patchset I will test and review.
> 
> Best Regards,
> Nachiket Tarate
> 
> On Wed, Feb 24, 2021 at 4:44 PM Nachiket Tarate <
> nachiket.programmer at gmail.com> wrote:
> 
>> Apple HTTP Live Streaming Sample Encryption:
>> 
>> 
>> https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
>> 
>> Signed-off-by: Nachiket Tarate <nachiket.programmer at gmail.com>
>> ---
>> libavformat/Makefile         |   2 +-
>> libavformat/hls.c            | 105 ++++++++--
>> libavformat/hls_sample_aes.c | 391 +++++++++++++++++++++++++++++++++++
>> libavformat/hls_sample_aes.h |  66 ++++++
>> libavformat/mpegts.c         |  12 ++
>> 5 files changed, 562 insertions(+), 14 deletions(-)
>> create mode 100644 libavformat/hls_sample_aes.c
>> create mode 100644 libavformat/hls_sample_aes.h
>> 
>> diff --git a/libavformat/Makefile b/libavformat/Makefile
>> index fcb39ce133..62627d50a6 100644
>> --- a/libavformat/Makefile
>> +++ b/libavformat/Makefile
>> @@ -236,7 +236,7 @@ OBJS-$(CONFIG_HCOM_DEMUXER)              += hcom.o
>> pcm.o
>> OBJS-$(CONFIG_HDS_MUXER)                 += hdsenc.o
>> OBJS-$(CONFIG_HEVC_DEMUXER)              += hevcdec.o rawdec.o
>> OBJS-$(CONFIG_HEVC_MUXER)                += rawenc.o
>> -OBJS-$(CONFIG_HLS_DEMUXER)               += hls.o
>> +OBJS-$(CONFIG_HLS_DEMUXER)               += hls.o hls_sample_aes.o
>> OBJS-$(CONFIG_HLS_MUXER)                 += hlsenc.o hlsplaylist.o avc.o
>> OBJS-$(CONFIG_HNM_DEMUXER)               += hnm.o
>> OBJS-$(CONFIG_ICO_DEMUXER)               += icodec.o
>> diff --git a/libavformat/hls.c b/libavformat/hls.c
>> index af2468ad9b..3cb3853c79 100644
>> --- a/libavformat/hls.c
>> +++ b/libavformat/hls.c
>> @@ -2,6 +2,7 @@
>>  * Apple HTTP Live Streaming demuxer
>>  * Copyright (c) 2010 Martin Storsjo
>>  * Copyright (c) 2013 Anssi Hannula
>> + * Copyright (c) 2021 Nachiket Tarate
>>  *
>>  * This file is part of FFmpeg.
>>  *
>> @@ -39,6 +40,8 @@
>> #include "avio_internal.h"
>> #include "id3v2.h"
>> 
>> +#include "hls_sample_aes.h"
>> +
>> #define INITIAL_BUFFER_SIZE 32768
>> 
>> #define MAX_FIELD_LEN 64
>> @@ -145,6 +148,10 @@ struct playlist {
>>     int id3_changed; /* ID3 tag data has changed at some point */
>>     ID3v2ExtraMeta *id3_deferred_extra; /* stored here until subdemuxer
>> is opened */
>> 
>> +    /* Used in case of SAMPLE-AES encryption method */
>> +    HLSAudioSetupInfo audio_setup_info;
>> +    HLSCryptoContext  crypto_ctx;
>> +
>>     int64_t seek_timestamp;
>>     int seek_flags;
>>     int seek_stream_index; /* into subdemuxer stream array */
>> @@ -266,6 +273,8 @@ static void free_playlist_list(HLSContext *c)
>>             pls->ctx->pb = NULL;
>>             avformat_close_input(&pls->ctx);
>>         }
>> +        if (pls->crypto_ctx.aes_ctx)
>> +             av_free(pls->crypto_ctx.aes_ctx);
>>         av_free(pls);
>>     }
>>     av_freep(&c->playlists);
>> @@ -994,7 +1003,10 @@ fail:
>> 
>> static struct segment *current_segment(struct playlist *pls)
>> {
>> -    return pls->segments[pls->cur_seq_no - pls->start_seq_no];
>> +    int64_t n = pls->cur_seq_no - pls->start_seq_no;
>> +    if (n >= pls->n_segments)
>> +        return NULL;
>> +    return pls->segments[n];
>> }
>> 
>> static struct segment *next_segment(struct playlist *pls)
>> @@ -1023,10 +1035,11 @@ static int read_from_url(struct playlist *pls,
>> struct segment *seg,
>> 
>> /* Parse the raw ID3 data and pass contents to caller */
>> static void parse_id3(AVFormatContext *s, AVIOContext *pb,
>> -                      AVDictionary **metadata, int64_t *dts,
>> +                      AVDictionary **metadata, int64_t *dts,
>> HLSAudioSetupInfo *audio_setup_info,
>>                       ID3v2ExtraMetaAPIC **apic, ID3v2ExtraMeta
>> **extra_meta)
>> {
>>     static const char id3_priv_owner_ts[] =
>> "com.apple.streaming.transportStreamTimestamp";
>> +    static const char id3_priv_owner_audio_setup[] =
>> "com.apple.streaming.audioDescription";
>>     ID3v2ExtraMeta *meta;
>> 
>>     ff_id3v2_read_dict(pb, metadata, ID3v2_DEFAULT_MAGIC, extra_meta);
>> @@ -1041,6 +1054,8 @@ static void parse_id3(AVFormatContext *s,
>> AVIOContext *pb,
>>                     *dts = ts;
>>                 else
>>                     av_log(s, AV_LOG_ERROR, "Invalid HLS ID3 audio
>> timestamp %"PRId64"\n", ts);
>> +            } else if (priv->datasize >= 8 && !strcmp(priv->owner,
>> id3_priv_owner_audio_setup)) {
>> +                ff_hls_read_audio_setup_info(audio_setup_info,
>> priv->data, priv->datasize);
>>             }
>>         } else if (!strcmp(meta->tag, "APIC") && apic)
>>             *apic = &meta->data.apic;
>> @@ -1084,7 +1099,7 @@ static void handle_id3(AVIOContext *pb, struct
>> playlist *pls)
>>     ID3v2ExtraMeta *extra_meta = NULL;
>>     int64_t timestamp = AV_NOPTS_VALUE;
>> 
>> -    parse_id3(pls->ctx, pb, &metadata, &timestamp, &apic, &extra_meta);
>> +    parse_id3(pls->ctx, pb, &metadata, &timestamp,
>> &pls->audio_setup_info, &apic, &extra_meta);
>> 
>>     if (timestamp != AV_NOPTS_VALUE) {
>>         pls->id3_mpegts_timestamp = timestamp;
>> @@ -1238,10 +1253,7 @@ static int open_input(HLSContext *c, struct
>> playlist *pls, struct segment *seg,
>>     av_log(pls->parent, AV_LOG_VERBOSE, "HLS request for url '%s', offset
>> %"PRId64", playlist %d\n",
>>            seg->url, seg->url_offset, pls->index);
>> 
>> -    if (seg->key_type == KEY_NONE) {
>> -        ret = open_url(pls->parent, in, seg->url, &c->avio_opts, opts,
>> &is_http);
>> -    } else if (seg->key_type == KEY_AES_128) {
>> -        char iv[33], key[33], url[MAX_URL_SIZE];
>> +    if (seg->key_type == KEY_AES_128 || seg->key_type == KEY_SAMPLE_AES) {
>>         if (strcmp(seg->key, pls->key_url)) {
>>             AVIOContext *pb = NULL;
>>             if (open_url(pls->parent, &pb, seg->key, &c->avio_opts, opts,
>> NULL) == 0) {
>> @@ -1257,6 +1269,10 @@ static int open_input(HLSContext *c, struct
>> playlist *pls, struct segment *seg,
>>             }
>>             av_strlcpy(pls->key_url, seg->key, sizeof(pls->key_url));
>>         }
>> +    }
>> +
>> +    if (seg->key_type == KEY_AES_128) {
>> +        char iv[33], key[33], url[MAX_URL_SIZE];
>>         ff_data_to_hex(iv, seg->iv, sizeof(seg->iv), 0);
>>         ff_data_to_hex(key, pls->key, sizeof(pls->key), 0);
>>         iv[32] = key[32] = '\0';
>> @@ -1273,13 +1289,9 @@ static int open_input(HLSContext *c, struct
>> playlist *pls, struct segment *seg,
>>             goto cleanup;
>>         }
>>         ret = 0;
>> -    } else if (seg->key_type == KEY_SAMPLE_AES) {
>> -        av_log(pls->parent, AV_LOG_ERROR,
>> -               "SAMPLE-AES encryption is not supported yet\n");
>> -        ret = AVERROR_PATCHWELCOME;
>> +    } else {
>> +        ret = open_url(pls->parent, in, seg->url, &c->avio_opts, opts,
>> &is_http);
>>     }
>> -    else
>> -      ret = AVERROR(ENOSYS);
>> 
>>     /* Seek to the requested position. If this was a HTTP request, the
>> offset
>>      * should already be where want it to, but this allows e.g. local
>> testing
>> @@ -1948,6 +1960,7 @@ static int hls_read_header(AVFormatContext *s)
>>         struct playlist *pls = c->playlists[i];
>>         char *url;
>>         ff_const59 AVInputFormat *in_fmt = NULL;
>> +        struct segment *seg = NULL;
>> 
>>         if (!(pls->ctx = avformat_alloc_context())) {
>>             ret = AVERROR(ENOMEM);
>> @@ -1980,8 +1993,41 @@ static int hls_read_header(AVFormatContext *s)
>>             pls->ctx = NULL;
>>             goto fail;
>>         }
>> +
>>         ffio_init_context(&pls->pb, pls->read_buffer,
>> INITIAL_BUFFER_SIZE, 0, pls,
>>                           read_data, NULL, NULL);
>> +
>> +        /*
>> +         * If encryption scheme is SAMPLE-AES, try to read  ID3 tags of
>> +         * external audio track that contains audio setup information
>> +         */
>> +        seg = current_segment(pls);
>> +        if (seg && seg->key_type == KEY_SAMPLE_AES && pls->n_renditions >
>> 0 &&
>> +            pls->renditions[0]->type == AVMEDIA_TYPE_AUDIO) {
>> +            uint8_t buf[HLS_MAX_ID3_TAGS_DATA_LEN];
>> +            if ((ret = avio_read(&pls->pb, buf,
>> HLS_MAX_ID3_TAGS_DATA_LEN)) < 0) {
>> +                /* Fail if error was not end of file */
>> +                if (ret != AVERROR_EOF) {
>> +                    avformat_free_context(pls->ctx);
>> +                    pls->ctx = NULL;
>> +                    goto fail;
>> +                }
>> +            }
>> +            ret = 0;
>> +        }
>> +
>> +        /*
>> +         * If encryption scheme is SAMPLE-AES and audio setup information
>> is present in external audio track,
>> +         * use that information to find the media format, otherwise probe
>> input data
>> +         */
>> +        if (seg && seg->key_type == KEY_SAMPLE_AES &&
>> pls->is_id3_timestamped &&
>> +            pls->audio_setup_info.codec_id != AV_CODEC_ID_NONE) {
>> +            void *iter = NULL;
>> +            while ((in_fmt = (ff_const59 AVInputFormat
>> *)av_demuxer_iterate(&iter)))
>> +                if (in_fmt->raw_codec_id ==
>> pls->audio_setup_info.codec_id) {
>> +                    break;
>> +                }
>> +        } else {
>>         pls->ctx->probesize = s->probesize > 0 ? s->probesize : 1024 * 4;
>>         pls->ctx->max_analyze_duration = s->max_analyze_duration > 0 ?
>> s->max_analyze_duration : 4 * AV_TIME_BASE;
>>         pls->ctx->interrupt_callback = s->interrupt_callback;
>> @@ -1999,6 +2045,25 @@ static int hls_read_header(AVFormatContext *s)
>>             goto fail;
>>         }
>>         av_free(url);
>> +        }
>> +
>> +        if (seg && seg->key_type == KEY_SAMPLE_AES) {
>> +            if (!pls->is_id3_timestamped && pls->n_renditions > 0 &&
>> pls->renditions[0]->type != AVMEDIA_TYPE_AUDIO &&
>> +                strcmp(in_fmt->name, "mpegts")) {
>> +                av_log(s, AV_LOG_ERROR, "SAMPLE-AES encryption is not
>> supported for fragmented MP4 format yet\n");
>> +                ret = AVERROR_PATCHWELCOME;
>> +            } else {
>> +                pls->crypto_ctx.aes_ctx = av_aes_alloc();
>> +                if (!pls->crypto_ctx.aes_ctx)
>> +                    ret = AVERROR(ENOMEM);
>> +            }
>> +            if (ret != 0) {
>> +                avformat_free_context(pls->ctx);
>> +                pls->ctx = NULL;
>> +                goto fail;
>> +            }
>> +        }
>> +
>>         pls->ctx->pb       = &pls->pb;
>>         pls->ctx->io_open  = nested_io_open;
>>         pls->ctx->flags   |= s->flags & ~AVFMT_FLAG_CUSTOM_IO;
>> @@ -2027,7 +2092,12 @@ static int hls_read_header(AVFormatContext *s)
>>          * on us if they want to.
>>          */
>>         if (pls->is_id3_timestamped || (pls->n_renditions > 0 &&
>> pls->renditions[0]->type == AVMEDIA_TYPE_AUDIO)) {
>> +            if (seg && seg->key_type == KEY_SAMPLE_AES &&
>> pls->audio_setup_info.setup_data_length > 0 &&
>> +                pls->ctx->nb_streams == 1)
>> +                ret = ff_hls_parse_audio_setup_info(pls->ctx->streams[0],
>> &pls->audio_setup_info);
>> +            else
>>             ret = avformat_find_stream_info(pls->ctx, NULL);
>> +
>>             if (ret < 0)
>>                 goto fail;
>>         }
>> @@ -2157,6 +2227,7 @@ static int hls_read_packet(AVFormatContext *s,
>> AVPacket *pkt)
>>             while (1) {
>>                 int64_t ts_diff;
>>                 AVRational tb;
>> +                struct segment *seg = NULL;
>>                 ret = av_read_frame(pls->ctx, &pls->pkt);
>>                 if (ret < 0) {
>>                     if (!avio_feof(&pls->pb) && ret != AVERROR_EOF)
>> @@ -2175,6 +2246,14 @@ static int hls_read_packet(AVFormatContext *s,
>> AVPacket *pkt)
>>                             get_timebase(pls), AV_TIME_BASE_Q);
>>                 }
>> 
>> +                seg = current_segment(pls);
>> +                if (seg && seg->key_type == KEY_SAMPLE_AES) {
>> +                    enum AVCodecID codec_id =
>> pls->ctx->streams[pls->pkt.stream_index]->codecpar->codec_id;
>> +                    memcpy(pls->crypto_ctx.iv, seg->iv, sizeof(seg->iv));
>> +                    memcpy(pls->crypto_ctx.key, pls->key,
>> sizeof(pls->key));
>> +                    ff_hls_decrypt_frame(codec_id, &pls->crypto_ctx,
>> &pls->pkt);
>> +                }
>> +
>>                 if (pls->seek_timestamp == AV_NOPTS_VALUE)
>>                     break;
>> 
>> diff --git a/libavformat/hls_sample_aes.c b/libavformat/hls_sample_aes.c
>> new file mode 100644
>> index 0000000000..0407a15b0f
>> --- /dev/null
>> +++ b/libavformat/hls_sample_aes.c
>> @@ -0,0 +1,391 @@
>> +/*
>> + * Apple HTTP Live Streaming Sample Encryption/Decryption
>> + *
>> + * Copyright (c) 2021 Nachiket Tarate
>> + *
>> + * 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
>> + */
>> +
>> +/**
>> + * @file
>> + * Apple HTTP Live Streaming Sample Encryption
>> + *
>> https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
>> + */
>> +
>> +#include "hls_sample_aes.h"
>> +
>> +#include "libavcodec/adts_header.h"
>> +#include "libavcodec/adts_parser.h"
>> +#include "libavcodec/ac3_parser_internal.h"
>> +
>> +
>> +typedef struct NALUnit {
>> +    uint8_t     *data;
>> +    int         type;
>> +    int         length;
>> +    int         start_code_length;
>> +} NALUnit;
>> +
>> +typedef struct AudioFrame {
>> +    uint8_t     *data;
>> +    int         length;
>> +    int         header_length;
>> +} AudioFrame;
>> +
>> +typedef struct CodecParserContext {
>> +    const uint8_t   *buf_ptr;
>> +    const uint8_t   *buf_end;
>> +} CodecParserContext;
>> +
>> +static const int eac3_sample_rate_tab[] = { 48000, 44100, 32000, 0 };
>> +
>> +void ff_hls_read_audio_setup_info(HLSAudioSetupInfo *info, const uint8_t
>> *buf, size_t size)
>> +{
>> +    if (size < 8)
>> +        return;
>> +
>> +    info->codec_tag             = AV_RL32(buf);
>> +
>> +    if (info->codec_tag == MKTAG('z','a', 'a', 'c'))
>> +        info->codec_id = AV_CODEC_ID_AAC;
>> +    else if (info->codec_tag == MKTAG('z','a', 'c', '3'))
>> +        info->codec_id = AV_CODEC_ID_AC3;
>> +    else if (info->codec_tag == MKTAG('z','e', 'c', '3'))
>> +        info->codec_id = AV_CODEC_ID_EAC3;
>> +    else
>> +        info->codec_id = AV_CODEC_ID_NONE;
>> +
>> +    buf += 4;
>> +    info->priming               = AV_RL16(buf);
>> +    buf += 2;
>> +    info->version               = *buf++;
>> +    info->setup_data_length     = *buf++;
>> +
>> +    if (info->setup_data_length > size - 8)
>> +        info->setup_data_length = size - 8;
>> +
>> +    if (info->setup_data_length > HLS_MAX_AUDIO_SETUP_DATA_LEN)
>> +        return;
>> +
>> +    memcpy(info->setup_data, buf, info->setup_data_length);
>> +}
>> +
>> +int ff_hls_parse_audio_setup_info(AVStream *st, HLSAudioSetupInfo *info)
>> +{
>> +    int ret = 0;
>> +
>> +    st->codecpar->codec_tag = info->codec_tag;
>> +
>> +    if (st->codecpar->codec_id == AV_CODEC_ID_AAC)
>> +        return 0;
>> +
>> +    if (st->codecpar->codec_id != AV_CODEC_ID_AC3 &&
>> st->codecpar->codec_id != AV_CODEC_ID_EAC3)
>> +        return AVERROR_INVALIDDATA;
>> +
>> +    if (st->codecpar->codec_id == AV_CODEC_ID_AC3) {
>> +
>> +        AC3HeaderInfo *ac3hdr = NULL;
>> +
>> +        ret = avpriv_ac3_parse_header(&ac3hdr, info->setup_data,
>> info->setup_data_length);
>> +        if (ret < 0) {
>> +            if (ret != AVERROR(ENOMEM))
>> +                av_free(ac3hdr);
>> +            return ret;
>> +        }
>> +
>> +        st->codecpar->sample_rate       = ac3hdr->sample_rate;
>> +        st->codecpar->channels          = ac3hdr->channels;
>> +        st->codecpar->channel_layout    = ac3hdr->channel_layout;
>> +        st->codecpar->bit_rate          = ac3hdr->bit_rate;
>> +
>> +        av_free(ac3hdr);
>> +    } else {  /*  Parse 'dec3' EC3SpecificBox */
>> +
>> +        GetBitContext gb;
>> +        int data_rate, fscod, acmod, lfeon;
>> +
>> +        ret = init_get_bits8(&gb, info->setup_data,
>> info->setup_data_length);
>> +        if (ret < 0)
>> +            return AVERROR_INVALIDDATA;
>> +
>> +        data_rate = get_bits(&gb, 13);
>> +        skip_bits(&gb, 3);
>> +        fscod = get_bits(&gb, 2);
>> +        skip_bits(&gb, 10);
>> +        acmod = get_bits(&gb, 3);
>> +        lfeon = get_bits(&gb, 1);
>> +
>> +        st->codecpar->sample_rate = eac3_sample_rate_tab[fscod];
>> +
>> +        st->codecpar->channel_layout =
>> avpriv_ac3_channel_layout_tab[acmod];
>> +        if (lfeon)
>> +            st->codecpar->channel_layout |= AV_CH_LOW_FREQUENCY;
>> +
>> +        st->codecpar->channels =
>> av_get_channel_layout_nb_channels(st->codecpar->channel_layout);
>> +
>> +        st->codecpar->bit_rate = data_rate*1000;
>> +    }
>> +
>> +    return 0;
>> +}
>> +
>> +/*
>> + * Remove start code emulation prevention 0x03 bytes
>> + */
>> +static void remove_scep_3_bytes(NALUnit *nalu)
>> +{
>> +    int i = 0;
>> +    int j = 0;
>> +
>> +    uint8_t *data = nalu->data;
>> +
>> +    while (i < nalu->length) {
>> +        if (nalu->length - i > 3 && AV_RB24(&data[i]) == 0x000003) {
>> +            data[j++] = data[i++];
>> +            data[j++] = data[i++];
>> +            i++;
>> +        } else {
>> +            data[j++] = data[i++];
>> +        }
>> +    }
>> +
>> +    nalu->length = j;
>> +}
>> +
>> +static int get_next_nal_unit(CodecParserContext *ctx, NALUnit *nalu)
>> +{
>> +    const uint8_t *nalu_start = ctx->buf_ptr;
>> +
>> +    if (ctx->buf_end - ctx->buf_ptr >= 4 && AV_RB32(ctx->buf_ptr) ==
>> 0x00000001)
>> +        nalu->start_code_length = 4;
>> +    else if (ctx->buf_end - ctx->buf_ptr >= 3 && AV_RB24(ctx->buf_ptr) ==
>> 0x000001)
>> +        nalu->start_code_length = 3;
>> +    else /* No start code at the beginning of the NAL unit */
>> +        return -1;
>> +
>> +    ctx->buf_ptr += nalu->start_code_length;
>> +
>> +    while (ctx->buf_ptr < ctx->buf_end) {
>> +        if (ctx->buf_end - ctx->buf_ptr >= 4 && AV_RB32(ctx->buf_ptr) ==
>> 0x00000001)
>> +            break;
>> +        else if (ctx->buf_end - ctx->buf_ptr >= 3 &&
>> AV_RB24(ctx->buf_ptr) == 0x000001)
>> +            break;
>> +        ctx->buf_ptr++;
>> +    }
>> +
>> +    nalu->data  = (uint8_t *)nalu_start + nalu->start_code_length;
>> +    nalu->length = ctx->buf_ptr - nalu->data;
>> +    nalu->type  = *nalu->data & 0x1F;
>> +
>> +    return 0;
>> +}
>> +
>> +static int decrypt_nal_unit(HLSCryptoContext *crypto_ctx, NALUnit *nalu)
>> +{
>> +    int ret = 0;
>> +    int rem_bytes;
>> +    uint8_t *data;
>> +    uint8_t iv[16];
>> +
>> +    ret = av_aes_init(crypto_ctx->aes_ctx, crypto_ctx->key, 16 * 8, 1);
>> +    if (ret < 0)
>> +        return ret;
>> +
>> +    /* Remove start code emulation prevention 0x03 bytes */
>> +    remove_scep_3_bytes(nalu);
>> +
>> +    data = nalu->data + 32;
>> +    rem_bytes = nalu->length - 32;
>> +
>> +    memcpy(iv, crypto_ctx->iv, 16);
>> +
>> +    while (rem_bytes > 0) {
>> +        if (rem_bytes > 16) {
>> +            av_aes_crypt(crypto_ctx->aes_ctx, data, data, 1, iv, 1);
>> +            data += 16;
>> +            rem_bytes -= 16;
>> +        }
>> +        data += FFMIN(144, rem_bytes);
>> +        rem_bytes -= FFMIN(144, rem_bytes);
>> +    }
>> +
>> +    return 0;
>> +}
>> +
>> +static int decrypt_video_frame(HLSCryptoContext *crypto_ctx, AVPacket
>> *pkt)
>> +{
>> +    int ret = 0;
>> +    CodecParserContext  ctx;
>> +    NALUnit nalu;
>> +    uint8_t *data_ptr;
>> +    int move_nalu = 0;
>> +
>> +    memset(&ctx, 0, sizeof(ctx));
>> +    ctx.buf_ptr  = pkt->data;
>> +    ctx.buf_end = pkt->data + pkt->size;
>> +
>> +    data_ptr = pkt->data;
>> +
>> +    while (ctx.buf_ptr < ctx.buf_end) {
>> +        memset(&nalu, 0, sizeof(nalu));
>> +        ret = get_next_nal_unit(&ctx, &nalu);
>> +        if (ret < 0)
>> +            return ret;
>> +        if ((nalu.type == 0x01 || nalu.type == 0x05) && nalu.length > 48)
>> {
>> +            int encrypted_nalu_length = nalu.length;
>> +            ret = decrypt_nal_unit(crypto_ctx, &nalu);
>> +            if (ret < 0)
>> +                return ret;
>> +            move_nalu = nalu.length != encrypted_nalu_length;
>> +        }
>> +        if (move_nalu)
>> +            memmove(data_ptr, nalu.data - nalu.start_code_length,
>> nalu.start_code_length + nalu.length);
>> +        data_ptr += nalu.start_code_length + nalu.length;
>> +    }
>> +
>> +    av_shrink_packet(pkt, data_ptr - pkt->data);
>> +
>> +    return 0;
>> +}
>> +
>> +static int get_next_adts_frame(CodecParserContext *ctx, AudioFrame *frame)
>> +{
>> +    int ret = 0;
>> +
>> +    AACADTSHeaderInfo *adts_hdr = NULL;
>> +
>> +    /* Find next sync word 0xFFF */
>> +    while (ctx->buf_ptr < ctx->buf_end - 1) {
>> +        if (*ctx->buf_ptr == 0xFF && *(ctx->buf_ptr + 1) & 0xF0 == 0xF0)
>> +            break;
>> +        ctx->buf_ptr++;
>> +    }
>> +
>> +    if (ctx->buf_ptr >= ctx->buf_end - 1)
>> +        return -1;
>> +
>> +    frame->data = (uint8_t*)ctx->buf_ptr;
>> +
>> +    ret = avpriv_adts_header_parse (&adts_hdr, frame->data, ctx->buf_end
>> - frame->data);
>> +    if (ret < 0)
>> +        return ret;
>> +
>> +    frame->header_length = adts_hdr->crc_absent ? AV_AAC_ADTS_HEADER_SIZE
>> : AV_AAC_ADTS_HEADER_SIZE + 2;
>> +    frame->length = adts_hdr->frame_length;
>> +
>> +    av_free(adts_hdr);
>> +
>> +    return 0;
>> +}
>> +
>> +static int get_next_ac3_eac3_sync_frame(CodecParserContext *ctx,
>> AudioFrame *frame)
>> +{
>> +    int ret = 0;
>> +
>> +    AC3HeaderInfo *hdr = NULL;
>> +
>> +    /* Find next sync word 0x0B77 */
>> +    while (ctx->buf_ptr < ctx->buf_end - 1) {
>> +        if (*ctx->buf_ptr == 0x0B && *(ctx->buf_ptr + 1) == 0x77)
>> +            break;
>> +        ctx->buf_ptr++;
>> +    }
>> +
>> +    if (ctx->buf_ptr >= ctx->buf_end - 1)
>> +        return -1;
>> +
>> +    frame->data = (uint8_t*)ctx->buf_ptr;
>> +    frame->header_length = 0;
>> +
>> +    ret = avpriv_ac3_parse_header(&hdr, frame->data, ctx->buf_end -
>> frame->data);
>> +    if (ret < 0) {
>> +        if (ret != AVERROR(ENOMEM))
>> +            av_free(hdr);
>> +        return ret;
>> +    }
>> +
>> +    frame->length = hdr->frame_size;
>> +
>> +    av_free(hdr);
>> +
>> +    return 0;
>> +}
>> +
>> +static int get_next_sync_frame(enum AVCodecID codec_id,
>> CodecParserContext *ctx, AudioFrame *frame)
>> +{
>> +    if (codec_id == AV_CODEC_ID_AAC)
>> +        return get_next_adts_frame(ctx, frame);
>> +    else if (codec_id == AV_CODEC_ID_AC3 || codec_id == AV_CODEC_ID_EAC3)
>> +        return get_next_ac3_eac3_sync_frame(ctx, frame);
>> +    else
>> +        return AVERROR_INVALIDDATA;
>> +}
>> +
>> +static int decrypt_sync_frame(enum AVCodecID codec_id, HLSCryptoContext
>> *crypto_ctx, AudioFrame *frame)
>> +{
>> +    int ret = 0;
>> +    uint8_t *data;
>> +    int num_of_encrypted_blocks;
>> +
>> +    ret = av_aes_init(crypto_ctx->aes_ctx, crypto_ctx->key, 16 * 8, 1);
>> +    if (ret < 0)
>> +        return ret;
>> +
>> +    data = frame->data + frame->header_length + 16;
>> +
>> +    num_of_encrypted_blocks = (frame->length - frame->header_length -
>> 16)/16;
>> +
>> +    av_aes_crypt(crypto_ctx->aes_ctx, data, data,
>> num_of_encrypted_blocks, crypto_ctx->iv, 1);
>> +
>> +    return 0;
>> +}
>> +
>> +static int decrypt_audio_frame(enum AVCodecID codec_id, HLSCryptoContext
>> *crypto_ctx, AVPacket *pkt)
>> +{
>> +    int ret = 0;
>> +    CodecParserContext  ctx;
>> +    AudioFrame frame;
>> +
>> +    memset(&ctx, 0, sizeof(ctx));
>> +    ctx.buf_ptr        = pkt->data;
>> +    ctx.buf_end = pkt->data + pkt->size;
>> +
>> +    while (ctx.buf_ptr < ctx.buf_end) {
>> +        memset(&frame, 0, sizeof(frame));
>> +        ret = get_next_sync_frame(codec_id, &ctx, &frame);
>> +        if (ret < 0)
>> +            return ret;
>> +        if (frame.length - frame.header_length > 31) {
>> +            ret = decrypt_sync_frame(codec_id, crypto_ctx, &frame);
>> +            if (ret < 0)
>> +                return ret;
>> +        }
>> +        ctx.buf_ptr += frame.length;
>> +    }
>> +
>> +    return 0;
>> +}
>> +
>> +int ff_hls_decrypt_frame(enum AVCodecID codec_id, HLSCryptoContext
>> *crypto_ctx, AVPacket *pkt)
>> +{
>> +    if (codec_id == AV_CODEC_ID_H264)
>> +        return decrypt_video_frame(crypto_ctx, pkt);
>> +    else if (codec_id == AV_CODEC_ID_AAC || codec_id == AV_CODEC_ID_AC3
>> || codec_id == AV_CODEC_ID_EAC3)
>> +        return decrypt_audio_frame(codec_id, crypto_ctx, pkt);
>> +
>> +    return AVERROR_INVALIDDATA;
>> +}
>> diff --git a/libavformat/hls_sample_aes.h b/libavformat/hls_sample_aes.h
>> new file mode 100644
>> index 0000000000..cf80e41cb0
>> --- /dev/null
>> +++ b/libavformat/hls_sample_aes.h
>> @@ -0,0 +1,66 @@
>> +/*
>> + * Apple HTTP Live Streaming Sample Encryption/Decryption
>> + *
>> + * Copyright (c) 2021 Nachiket Tarate
>> + *
>> + * 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
>> + */
>> +
>> +/**
>> + * @file
>> + * Apple HTTP Live Streaming Sample Encryption
>> + *
>> https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
>> + */
>> +
>> +#ifndef AVFORMAT_HLS_SAMPLE_AES_H
>> +#define AVFORMAT_HLS_SAMPLE_AES_H
>> +
>> +#include <stdint.h>
>> +
>> +#include "avformat.h"
>> +
>> +#include "libavcodec/avcodec.h"
>> +#include "libavutil/aes.h"
>> +
>> +#define HLS_MAX_ID3_TAGS_DATA_LEN       138
>> +#define HLS_MAX_AUDIO_SETUP_DATA_LEN    10
>> +
>> +
>> +typedef struct HLSCryptoContext {
>> +    struct AVAES   *aes_ctx;
>> +    uint8_t            key[16];
>> +    uint8_t            iv[16];
>> +} HLSCryptoContext;
>> +
>> +typedef struct HLSAudioSetupInfo {
>> +    enum AVCodecID      codec_id;
>> +    uint32_t            codec_tag;
>> +    uint16_t            priming;
>> +    uint8_t             version;
>> +    uint8_t             setup_data_length;
>> +    uint8_t             setup_data[HLS_MAX_AUDIO_SETUP_DATA_LEN];
>> +} HLSAudioSetupInfo;
>> +
>> +
>> +void ff_hls_read_audio_setup_info(HLSAudioSetupInfo *info, const uint8_t
>> *buf, size_t size);
>> +
>> +int ff_hls_parse_audio_setup_info(AVStream *st, HLSAudioSetupInfo *info);
>> +
>> +int ff_hls_decrypt_frame(enum AVCodecID codec_id, HLSCryptoContext
>> *crypto_ctx, AVPacket *pkt);
>> +
>> +#endif /* AVFORMAT_HLS_SAMPLE_AES_H */
>> +
>> diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
>> index e283ec09d7..dc611ae788 100644
>> --- a/libavformat/mpegts.c
>> +++ b/libavformat/mpegts.c
>> @@ -839,6 +839,16 @@ static const StreamType MISC_types[] = {
>>     { 0 },
>> };
>> 
>> +/* HLS Sample Encryption Types  */
>> +static const StreamType HLS_SAMPLE_ENC_types[] = {
>> +    { 0xdb, AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_H264},
>> +    { 0xcf, AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_AAC },
>> +    { 0xc1, AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_AC3 },
>> +    { 0xc2, AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_EAC3},
>> +    { 0 },
>> +};
>> +
>> +
>> static const StreamType REGD_types[] = {
>>     { MKTAG('d', 'r', 'a', 'c'), AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_DIRAC },
>>     { MKTAG('A', 'C', '-', '3'), AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_AC3   },
>> @@ -948,6 +958,8 @@ static int mpegts_set_stream_info(AVStream *st,
>> PESContext *pes,
>>     }
>>     if (st->codecpar->codec_id == AV_CODEC_ID_NONE)
>>         mpegts_find_stream_type(st, pes->stream_type, MISC_types);
>> +    if (st->codecpar->codec_id == AV_CODEC_ID_NONE)
>> +        mpegts_find_stream_type(st, pes->stream_type,
>> HLS_SAMPLE_ENC_types);
>>     if (st->codecpar->codec_id == AV_CODEC_ID_NONE) {
>>         st->codecpar->codec_id  = old_codec_id;
>>         st->codecpar->codec_type = old_codec_type;
>> --
>> 2.17.1
>> 
>> 
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Thanks

Steven Liu





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