> 2021年3月1日 下午12:55，Nachiket Tarate <nachiket.program...@gmail.com> 写道： > > This is an updated version of the patch in which I have added the check. If > the segments are in Fragmented MP4 format, HLS demuxer quits by giving an > error message: > > "SAMPLE-AES encryption is not supported for fragmented MP4 format yet” I don’t think is a good resolution for SAMPLE-AES encryption and decryption. You should support that if you want support SAMPLE-AES in hls, because SAMPLE-AES not only support in MPEG-TS, but also support fragment mp4. Whatever, if you only support mpegts en[de]cryption, it should be a half part patch.
> > Best Regards, > Nachiket Tarate > > On Mon, Mar 1, 2021 at 10:13 AM Steven Liu <l...@chinaffmpeg.org> wrote: > >> >> >>> 2021年3月1日 下午12:35，Nachiket Tarate <nachiket.program...@gmail.com> 写道： >>> >>> @Steven Liu <l...@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.program...@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.program...@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, ×tamp, &apic, &extra_meta); >>>> + parse_id3(pls->ctx, pb, &metadata, ×tamp, >>>> &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 >>>> >>>> >>> _______________________________________________ >>> ffmpeg-devel mailing list >>> ffmpeg-devel@ffmpeg.org >>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >>> >>> To unsubscribe, visit link above, or email >>> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe". >> >> Thanks >> >> Steven Liu >> >> >> >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> >> To unsubscribe, visit link above, or email >> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe". > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe". Thanks Steven Liu _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
