[FFmpeg-devel] [PATCH 2/3] libavformat/hls: add support for decryption of HLS streams in MPEG-TS format protected using SAMPLE-AES encryption
Nachiket Tarate
nachiket.programmer at gmail.com
Mon Mar 1 06:55:53 EET 2021
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 <lq at chinaffmpeg.org> wrote:
>
>
> > 2021年3月1日 下午12:35，Nachiket Tarate <nachiket.programmer at gmail.com> 写道：
> >
> > @Steven Liu <lq at chinaffmpeg.org>
> >
> > Can we merge this patch ?
> I’m waiting update patch for fragment mp4 encryption.
> After new version of the patchset I will test and review.
> >
> > Best Regards,
> > Nachiket Tarate
> >
> > On Wed, Feb 24, 2021 at 4:44 PM Nachiket Tarate <
> > nachiket.programmer at gmail.com> wrote:
> >
> >> Apple HTTP Live Streaming Sample Encryption:
> >>
> >>
> >>
> https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
> >>
> >> Signed-off-by: Nachiket Tarate <nachiket.programmer at gmail.com>
> >> ---
> >> libavformat/Makefile | 2 +-
> >> libavformat/hls.c | 105 ++++++++--
> >> libavformat/hls_sample_aes.c | 391 +++++++++++++++++++++++++++++++++++
> >> libavformat/hls_sample_aes.h | 66 ++++++
> >> libavformat/mpegts.c | 12 ++
> >> 5 files changed, 562 insertions(+), 14 deletions(-)
> >> create mode 100644 libavformat/hls_sample_aes.c
> >> create mode 100644 libavformat/hls_sample_aes.h
> >>
> >> diff --git a/libavformat/Makefile b/libavformat/Makefile
> >> index fcb39ce133..62627d50a6 100644
> >> --- a/libavformat/Makefile
> >> +++ b/libavformat/Makefile
> >> @@ -236,7 +236,7 @@ OBJS-$(CONFIG_HCOM_DEMUXER) += hcom.o
> >> pcm.o
> >> OBJS-$(CONFIG_HDS_MUXER) += hdsenc.o
> >> OBJS-$(CONFIG_HEVC_DEMUXER) += hevcdec.o rawdec.o
> >> OBJS-$(CONFIG_HEVC_MUXER) += rawenc.o
> >> -OBJS-$(CONFIG_HLS_DEMUXER) += hls.o
> >> +OBJS-$(CONFIG_HLS_DEMUXER) += hls.o hls_sample_aes.o
> >> OBJS-$(CONFIG_HLS_MUXER) += hlsenc.o hlsplaylist.o avc.o
> >> OBJS-$(CONFIG_HNM_DEMUXER) += hnm.o
> >> OBJS-$(CONFIG_ICO_DEMUXER) += icodec.o
> >> diff --git a/libavformat/hls.c b/libavformat/hls.c
> >> index af2468ad9b..3cb3853c79 100644
> >> --- a/libavformat/hls.c
> >> +++ b/libavformat/hls.c
> >> @@ -2,6 +2,7 @@
> >> * Apple HTTP Live Streaming demuxer
> >> * Copyright (c) 2010 Martin Storsjo
> >> * Copyright (c) 2013 Anssi Hannula
> >> + * Copyright (c) 2021 Nachiket Tarate
> >> *
> >> * This file is part of FFmpeg.
> >> *
> >> @@ -39,6 +40,8 @@
> >> #include "avio_internal.h"
> >> #include "id3v2.h"
> >>
> >> +#include "hls_sample_aes.h"
> >> +
> >> #define INITIAL_BUFFER_SIZE 32768
> >>
> >> #define MAX_FIELD_LEN 64
> >> @@ -145,6 +148,10 @@ struct playlist {
> >> int id3_changed; /* ID3 tag data has changed at some point */
> >> ID3v2ExtraMeta *id3_deferred_extra; /* stored here until subdemuxer
> >> is opened */
> >>
> >> + /* Used in case of SAMPLE-AES encryption method */
> >> + HLSAudioSetupInfo audio_setup_info;
> >> + HLSCryptoContext crypto_ctx;
> >> +
> >> int64_t seek_timestamp;
> >> int seek_flags;
> >> int seek_stream_index; /* into subdemuxer stream array */
> >> @@ -266,6 +273,8 @@ static void free_playlist_list(HLSContext *c)
> >> pls->ctx->pb = NULL;
> >> avformat_close_input(&pls->ctx);
> >> }
> >> + if (pls->crypto_ctx.aes_ctx)
> >> + av_free(pls->crypto_ctx.aes_ctx);
> >> av_free(pls);
> >> }
> >> av_freep(&c->playlists);
> >> @@ -994,7 +1003,10 @@ fail:
> >>
> >> static struct segment *current_segment(struct playlist *pls)
> >> {
> >> - return pls->segments[pls->cur_seq_no - pls->start_seq_no];
> >> + int64_t n = pls->cur_seq_no - pls->start_seq_no;
> >> + if (n >= pls->n_segments)
> >> + return NULL;
> >> + return pls->segments[n];
> >> }
> >>
> >> static struct segment *next_segment(struct playlist *pls)
> >> @@ -1023,10 +1035,11 @@ static int read_from_url(struct playlist *pls,
> >> struct segment *seg,
> >>
> >> /* Parse the raw ID3 data and pass contents to caller */
> >> static void parse_id3(AVFormatContext *s, AVIOContext *pb,
> >> - AVDictionary **metadata, int64_t *dts,
> >> + AVDictionary **metadata, int64_t *dts,
> >> HLSAudioSetupInfo *audio_setup_info,
> >> ID3v2ExtraMetaAPIC **apic, ID3v2ExtraMeta
> >> **extra_meta)
> >> {
> >> static const char id3_priv_owner_ts[] =
> >> "com.apple.streaming.transportStreamTimestamp";
> >> + static const char id3_priv_owner_audio_setup[] =
> >> "com.apple.streaming.audioDescription";
> >> ID3v2ExtraMeta *meta;
> >>
> >> ff_id3v2_read_dict(pb, metadata, ID3v2_DEFAULT_MAGIC, extra_meta);
> >> @@ -1041,6 +1054,8 @@ static void parse_id3(AVFormatContext *s,
> >> AVIOContext *pb,
> >> *dts = ts;
> >> else
> >> av_log(s, AV_LOG_ERROR, "Invalid HLS ID3 audio
> >> timestamp %"PRId64"\n", ts);
> >> + } else if (priv->datasize >= 8 && !strcmp(priv->owner,
> >> id3_priv_owner_audio_setup)) {
> >> + ff_hls_read_audio_setup_info(audio_setup_info,
> >> priv->data, priv->datasize);
> >> }
> >> } else if (!strcmp(meta->tag, "APIC") && apic)
> >> *apic = &meta->data.apic;
> >> @@ -1084,7 +1099,7 @@ static void handle_id3(AVIOContext *pb, struct
> >> playlist *pls)
> >> ID3v2ExtraMeta *extra_meta = NULL;
> >> int64_t timestamp = AV_NOPTS_VALUE;
> >>
> >> - parse_id3(pls->ctx, pb, &metadata, ×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 at ffmpeg.org
> > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >
> > To unsubscribe, visit link above, or email
> > ffmpeg-devel-request at ffmpeg.org with subject "unsubscribe".
>
> Thanks
>
> Steven Liu
>
>
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
> To unsubscribe, visit link above, or email
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