[FFmpeg-devel] [PATCH 2/3] libavformat/hls: add support for decryption of HLS streams in MPEG-TS format protected using SAMPLE-AES encryption
Steven Liu
lq at chinaffmpeg.org
Mon Mar 1 07:59:57 EET 2021
> 2021年3月1日 下午12:55，Nachiket Tarate <nachiket.programmer at gmail.com> 写道：
>
> This is an updated version of the patch in which I have added the check. If
> the segments are in Fragmented MP4 format, HLS demuxer quits by giving an
> error message:
>
> "SAMPLE-AES encryption is not supported for fragmented MP4 format yet”
I don’t think is a good resolution for SAMPLE-AES encryption and decryption.
You should support that if you want support SAMPLE-AES in hls,
because SAMPLE-AES not only support in MPEG-TS, but also support fragment mp4.
Whatever, if you only support mpegts en[de]cryption, it should be a half part patch.
>
> Best Regards,
> Nachiket Tarate
>
> On Mon, Mar 1, 2021 at 10:13 AM Steven Liu <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
>> ffmpeg-devel-request at ffmpeg.org with subject "unsubscribe".
> _______________________________________________
> 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
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