> 2021年3月1日 下午12:35，Nachiket Tarate <[email protected]> 写道： > > @Steven Liu <[email protected]> > > 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 < > [email protected]> wrote: > >> Apple HTTP Live Streaming Sample Encryption: >> >> >> https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption >> >> Signed-off-by: Nachiket Tarate <[email protected]> >> --- >> 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 > [email protected] > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > [email protected] with subject "unsubscribe".
Thanks Steven Liu _______________________________________________ ffmpeg-devel mailing list [email protected] https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
