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" 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".