lgtm after you fix nitpick about if statements in my opinon August 1, 2021 10:54 AM, "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 | 128 +++++++-- > libavformat/hls_sample_encryption.c | 389 ++++++++++++++++++++++++++++ > libavformat/hls_sample_encryption.h | 65 +++++ > libavformat/mpegts.c | 11 + > 5 files changed, 576 insertions(+), 19 deletions(-) > create mode 100644 libavformat/hls_sample_encryption.c > create mode 100644 libavformat/hls_sample_encryption.h > > diff --git a/libavformat/Makefile b/libavformat/Makefile > index 813ddd3c20..7e3c9c23c6 100644 > --- a/libavformat/Makefile > +++ b/libavformat/Makefile > @@ -238,7 +238,7 @@ 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_encryption.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 3c1b80f60c..022dae0391 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_encryption.h" > + > #define INITIAL_BUFFER_SIZE 32768 > > #define MAX_FIELD_LEN 64 > @@ -145,6 +148,8 @@ struct playlist { > int id3_changed; /* ID3 tag data has changed at some point */ > ID3v2ExtraMeta *id3_deferred_extra; /* stored here until subdemuxer is opened > */ > > + HLSAudioSetupInfo audio_setup_info; > + > int64_t seek_timestamp; > int seek_flags; > int seek_stream_index; /* into subdemuxer stream array */ > @@ -214,6 +219,7 @@ typedef struct HLSContext { > int http_multiple; > int http_seekable; > AVIOContext *playlist_pb; > + HLSCryptoContext crypto_ctx; > } HLSContext; > > static void free_segment_dynarray(struct segment **segments, int n_segments) > @@ -1006,7 +1012,10 @@ 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) > @@ -1035,17 +1044,18 @@ 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); > for (meta = *extra_meta; meta; meta = meta->next) { > if (!strcmp(meta->tag, "PRIV")) { > ID3v2ExtraMetaPRIV *priv = &meta->data.priv; > - if (priv->datasize == 8 && !strcmp(priv->owner, id3_priv_owner_ts)) { > + if (priv->datasize == 8 && !av_strncasecmp(priv->owner, id3_priv_owner_ts, > 44)) { > /* 33-bit MPEG timestamp */ > int64_t ts = AV_RB64(priv->data); > av_log(s, AV_LOG_DEBUG, "HLS ID3 audio timestamp %"PRId64"\n", ts); > @@ -1053,6 +1063,8 @@ 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 && !av_strncasecmp(priv->owner, > id3_priv_owner_audio_setup, 36)) { > + ff_hls_senc_read_audio_setup_info(audio_setup_info, priv->data, > priv->datasize); > } > } else if (!strcmp(meta->tag, "APIC") && apic) > *apic = &meta->data.apic; > @@ -1096,7 +1108,7 @@ 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; > @@ -1250,10 +1262,7 @@ 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) { > @@ -1269,6 +1278,10 @@ 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'; > @@ -1285,13 +1298,9 @@ 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 > @@ -1854,6 +1863,9 @@ static int hls_close(AVFormatContext *s) > free_variant_list(c); > free_rendition_list(c); > > + if (c->crypto_ctx.aes_ctx) > + av_free(c->crypto_ctx.aes_ctx); > + > av_dict_free(&c->avio_opts); > ff_format_io_close(c->ctx, &c->playlist_pb); > > @@ -1959,7 +1971,8 @@ static int hls_read_header(AVFormatContext *s) > struct playlist *pls = c->playlists[i]; > const AVInputFormat *in_fmt = NULL; > char *url; > - AVDictionary *seg_format_opts = NULL; > + AVDictionary *options = NULL; > + struct segment *seg = NULL; > > if (!(pls->ctx = avformat_alloc_context())) > return AVERROR(ENOMEM); > @@ -1989,8 +2002,55 @@ static int hls_read_header(AVFormatContext *s) > pls->ctx = NULL; > return AVERROR(ENOMEM); > } > + > 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; > + return ret; > + } > + } > + ret = 0; > + /* Reset reading */ > + ff_format_io_close(pls->parent, &pls->input); > + pls->input = NULL; > + pls->input_read_done = 0; > + ff_format_io_close(pls->parent, &pls->input_next); > + pls->input_next = NULL; > + pls->input_next_requested = 0; > + pls->cur_seg_offset = 0; > + pls->cur_init_section = NULL; > + /* Reset EOF flag */ > + pls->pb.eof_reached = 0; > + /* Clear any buffered data */ > + pls->pb.buf_end = pls->pb.buf_ptr = pls->pb.buffer; > + /* Reset the position */ > + pls->pb.pos = 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 = (const 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; > @@ -2008,6 +2068,24 @@ static int hls_read_header(AVFormatContext *s) > return ret; > } > av_free(url); > + } > + > + if (seg && seg->key_type == KEY_SAMPLE_AES) { > + if (strstr(in_fmt->name, "mov")) { > + char key[33]; > + ff_data_to_hex(key, pls->key, sizeof(pls->key), 0); > + key[32] = '\0'; > + av_dict_set(&options, "decryption_key", key, AV_OPT_FLAG_DECODING_PARAM); > + } else if (!c->crypto_ctx.aes_ctx) { > + c->crypto_ctx.aes_ctx = av_aes_alloc(); > + if (!c->crypto_ctx.aes_ctx) { > + avformat_free_context(pls->ctx); > + pls->ctx = NULL; > + return AVERROR(ENOMEM); > + } > + } > + } > + > pls->ctx->pb = &pls->pb; > pls->ctx->io_open = nested_io_open; > pls->ctx->flags |= s->flags & ~AVFMT_FLAG_CUSTOM_IO; > @@ -2015,10 +2093,10 @@ static int hls_read_header(AVFormatContext *s) > if ((ret = ff_copy_whiteblacklists(pls->ctx, s)) < 0) > return ret; > > - av_dict_copy(&seg_format_opts, c->seg_format_opts, 0); > + av_dict_copy(&options, c->seg_format_opts, 0); > > - ret = avformat_open_input(&pls->ctx, pls->segments[0]->url, in_fmt, > &seg_format_opts); > - av_dict_free(&seg_format_opts); > + ret = avformat_open_input(&pls->ctx, pls->segments[0]->url, in_fmt, > &options); > + av_dict_free(&options); > if (ret < 0) > return ret; > > @@ -2039,7 +2117,12 @@ 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_senc_parse_audio_setup_info(pls->ctx->streams[0], > &pls->audio_setup_info); > + else > ret = avformat_find_stream_info(pls->ctx, NULL); > + > if (ret < 0) > return ret; > } > @@ -2166,6 +2249,7 @@ 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) > @@ -2184,6 +2268,14 @@ 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 && > !strstr(pls->ctx->iformat->name, "mov")) { > + enum AVCodecID codec_id = > pls->ctx->streams[pls->pkt->stream_index]->codecpar->codec_id; > + memcpy(c->crypto_ctx.iv, seg->iv, sizeof(seg->iv)); > + memcpy(c->crypto_ctx.key, pls->key, sizeof(pls->key)); > + ff_hls_senc_decrypt_frame(codec_id, &c->crypto_ctx, pls->pkt); > + } > + > if (pls->seek_timestamp == AV_NOPTS_VALUE) > break; > > diff --git a/libavformat/hls_sample_encryption.c > b/libavformat/hls_sample_encryption.c > new file mode 100644 > index 0000000000..6538c2dceb > --- /dev/null > +++ b/libavformat/hls_sample_encryption.c > @@ -0,0 +1,389 @@ > +/* > + * 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_encryption.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_senc_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_senc_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; This should be on same line as if statement for readability and clearity of intent. Or use brackets. > + 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; Same as above. > + 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); Same thing. > + 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; > + Why stop before minus 1 when you can read the second to last and last byte? Or maybe im reading this wrong. > + 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; > + Same critque as above > + 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_senc_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_encryption.h > b/libavformat/hls_sample_encryption.h > new file mode 100644 > index 0000000000..ff3f9c22f7 > --- /dev/null > +++ b/libavformat/hls_sample_encryption.h > @@ -0,0 +1,65 @@ > +/* > + * 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_ENCRYPTION_H > +#define AVFORMAT_HLS_SAMPLE_ENCRYPTION_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_senc_read_audio_setup_info(HLSAudioSetupInfo *info, const > uint8_t *buf, size_t size); > + > +int ff_hls_senc_parse_audio_setup_info(AVStream *st, HLSAudioSetupInfo > *info); > + > +int ff_hls_senc_decrypt_frame(enum AVCodecID codec_id, HLSCryptoContext > *crypto_ctx, AVPacket > *pkt); > + > +#endif /* AVFORMAT_HLS_SAMPLE_ENCRYPTION_H */ > + > diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c > index a02965bacf..25518e3172 100644 > --- a/libavformat/mpegts.c > +++ b/libavformat/mpegts.c > @@ -839,6 +839,15 @@ 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 +957,8 @@ 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". _______________________________________________ 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".