diff mbox series

[FFmpeg-devel,2/3] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer

Message ID 20210122160205.30108-1-nachiket.programmer@gmail.com
State Superseded
Headers show
Series [FFmpeg-devel,1/3] libavcodec/adts_header: add frame_length field and avpriv function to parse AAC ADTS header
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Commit Message

Nachiket Tarate Jan. 22, 2021, 4:02 p.m. UTC
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@gmail.com>
---
 libavformat/Makefile         |   2 +-
 libavformat/hls.c            |  98 ++++++--
 libavformat/hls_sample_aes.c | 463 +++++++++++++++++++++++++++++++++++
 libavformat/hls_sample_aes.h |  64 +++++
 libavformat/mpegts.c         |  12 +
 5 files changed, 623 insertions(+), 16 deletions(-)
 create mode 100644 libavformat/hls_sample_aes.c
 create mode 100644 libavformat/hls_sample_aes.h
diff mbox series

Patch

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 3a8fbcbe5f..c97930d98b 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -237,7 +237,7 @@  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 619e4800de..22e3b22650 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 +148,8 @@  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 */
@@ -986,7 +991,10 @@  fail:
 
 static struct segment *current_segment(struct playlist *pls)
 {
-    return pls->segments[pls->cur_seq_no - pls->start_seq_no];
+    int 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)
@@ -1015,10 +1023,11 @@  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);
@@ -1033,7 +1042,8 @@  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;
     }
@@ -1076,7 +1086,7 @@  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, &timestamp, &apic, &extra_meta);
+    parse_id3(pls->ctx, pb, &metadata, &timestamp, &pls->audio_setup_info, &apic, &extra_meta);
 
     if (timestamp != AV_NOPTS_VALUE) {
         pls->id3_mpegts_timestamp = timestamp;
@@ -1230,10 +1240,7 @@  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) {
@@ -1249,6 +1256,10 @@  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';
@@ -1265,13 +1276,8 @@  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 = AVERROR(ENOSYS);
+    } else
+        ret = open_url(pls->parent, in, seg->url, &c->avio_opts, opts, &is_http);
 
     /* 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
@@ -1940,6 +1946,7 @@  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);
@@ -1972,8 +1979,52 @@  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 = av_malloc(HLS_MAX_ID3_TAGS_DATA_LEN);
+            if (!buf) {
+                ret = AVERROR(ENOMEM);
+                avformat_free_context(pls->ctx);
+                pls->ctx = NULL;
+                goto fail;
+            }
+
+            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) {
+                    av_free(buf);
+                    avformat_free_context(pls->ctx);
+                    pls->ctx = NULL;
+                    goto fail;
+                }
+                ret   = 0;          /* error was end of file, nothing read */
+            }
+
+            av_free(buf);
+        }
+
+        /*
+         * 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->key_type == KEY_SAMPLE_AES && pls->is_id3_timestamped == 1 &&
+            pls->audio_setup_info.codec_id != AV_CODEC_ID_NONE) {
+            void *i = 0;
+            while ((in_fmt = (ff_const59 AVInputFormat *)av_demuxer_iterate(&i)))
+                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;
@@ -1991,6 +2042,8 @@  static int hls_read_header(AVFormatContext *s)
             goto fail;
         }
         av_free(url);
+        }
+
         pls->ctx->pb       = &pls->pb;
         pls->ctx->io_open  = nested_io_open;
         pls->ctx->flags   |= s->flags & ~AVFMT_FLAG_CUSTOM_IO;
@@ -2019,7 +2072,12 @@  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;
         }
@@ -2149,6 +2207,7 @@  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)
@@ -2167,6 +2226,15 @@  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) {
+                    HLSCryptoContext crypto_ctx;
+                    enum AVCodecID codec_id = pls->ctx->streams[pls->pkt.stream_index]->codecpar->codec_id;
+                    memcpy(crypto_ctx.iv, seg->iv, sizeof(seg->iv));
+                    memcpy(crypto_ctx.key, pls->key, sizeof(pls->key));
+                    ff_hls_decrypt_frame(codec_id, &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..d121e4736e
--- /dev/null
+++ b/libavformat/hls_sample_aes.c
@@ -0,0 +1,463 @@ 
+/*
+ * 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"
+#include "libavutil/aes.h"
+
+
+typedef struct NALUnit {
+    uint8_t     *data;
+    int         type;
+    int         length;
+} NALUnit;
+
+typedef struct AudioFrame {
+    uint8_t     *data;
+    int         length;
+    int         header_length;
+} AudioFrame;
+
+typedef struct CodecParserContext {
+    const uint8_t   *buf_in;
+    const uint8_t   *buf_end;
+    uint8_t         *buf_out;
+    int             next_start_code_length;
+} 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)
+{
+    info->codec_tag 		 = AV_RL32(buf);
+
+    if (!strncmp((const char*)&info->codec_tag, "zaac", 4))
+        info->codec_id = AV_CODEC_ID_AAC;
+    else if (!strncmp((const char*)&info->codec_tag, "zac3", 4))
+        info->codec_id = AV_CODEC_ID_AC3;
+    else if (!strncmp((const char*)&info->codec_tag, "zec3", 4))
+        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++;
+
+    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;
+
+    st->codecpar->extradata = av_mallocz(info->setup_data_length + AV_INPUT_BUFFER_PADDING_SIZE);
+
+    if (!st->codecpar->extradata)
+        return AVERROR(ENOMEM);
+
+    st->codecpar->extradata_size = info->setup_data_length;
+
+
+    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 && data[i] == 0x00 && data[i+1] == 0x00 && data[i+2] == 0x03 &&
+            (data[i+3] == 0x00 || data[i+3] == 0x01 || data[i+3] == 0x02 || data[i+3] == 0x03)) {
+            data[j] = 0x00;
+            data[j+1] = 0x00;
+            data[j+2] = data[i+3];
+            i += 4;
+            j += 3;
+        } else
+            data[j++] = data[i++];
+
+    nalu->length = j;
+}
+
+static int is_start_code(const uint8_t *buf, int zeros_in_start_code)
+{
+    for (int i = 0; i < zeros_in_start_code; i++)
+        if(*(buf++) != 0x00)
+            return 0;
+
+    if (*buf != 0x01)
+        return 0;
+
+    return 1;
+}
+
+static int get_next_nal_unit(CodecParserContext *ctx, NALUnit *nalu)
+{
+    int len = 0;
+    int nalu_start_offset = 0;
+
+    uint8_t *buf_out = ctx->buf_out;
+
+    if (ctx->next_start_code_length != 0) {
+        for (int i = 0; i < ctx->next_start_code_length - 1; i++) {
+            *buf_out++ = 0;
+            len++;
+        }
+        *buf_out++ = 1;
+        len++;
+        ctx->next_start_code_length = 0;
+    } else
+        while (ctx->buf_in < ctx->buf_end) {
+            len++;
+            if ((*buf_out++ = *ctx->buf_in++) != 0)
+                break;
+        }
+
+    if (ctx->buf_in >= ctx->buf_end)
+        if (len == 0)
+            return 0;
+        else
+            return -1;
+
+    /* No start code at the beginning of the NAL unit */
+    if(*(ctx->buf_in - 1) != 1 || len < 3)
+        return -1;
+
+    nalu_start_offset = len;
+
+    while (ctx->next_start_code_length == 0) {
+        if (ctx->buf_in >= ctx->buf_end) {
+            nalu->data   = ctx->buf_out + nalu_start_offset;
+            nalu->length = len - nalu_start_offset;
+            nalu->type   = *nalu->data & 0x1F;
+            ctx->buf_out += nalu_start_offset;
+            return 0;
+        }
+        *buf_out++ = *ctx->buf_in++;
+        len++;
+        if (is_start_code(ctx->buf_in - 4, 3))
+            ctx->next_start_code_length = 4;
+        else if (is_start_code(ctx->buf_in - 3, 2))
+            ctx->next_start_code_length = 3;
+        else
+            ctx->next_start_code_length = 0;
+    }
+
+    len -= ctx->next_start_code_length;
+
+    nalu->data	 = ctx->buf_out + nalu_start_offset;
+    nalu->length = len - nalu_start_offset;
+    nalu->type	 = *nalu->data & 0x1F;
+    ctx->buf_out += nalu_start_offset;
+    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];
+    uint8_t decrypted_block[16];
+
+    struct AVAES *aes_ctx = av_aes_alloc();
+    if (!aes_ctx)
+        return AVERROR(ENOMEM);
+
+    ret = av_aes_init(aes_ctx, crypto_ctx->key, 16 * 8, 1);
+    if (ret < 0) {
+        av_free(aes_ctx);
+        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(aes_ctx, decrypted_block, data, 1, iv, 1);
+            memcpy(iv, data, 16);
+            memcpy(data, decrypted_block, 16);
+            data += 16;
+            rem_bytes -= 16;
+        }
+        data += 144;
+        rem_bytes -= 144;
+    }
+
+    av_free(aes_ctx);
+
+    return 0;
+}
+
+static int decrypt_video_frame(HLSCryptoContext *crypto_ctx, AVPacket *pkt)
+{
+    int ret = 0;
+    CodecParserContext  ctx;
+    NALUnit nalu;
+
+    memset(&ctx, 0, sizeof(ctx));
+    ctx.buf_in  = pkt->data;
+    ctx.buf_out = pkt->data;
+    ctx.buf_end = pkt->data + pkt->size;
+
+    while (ctx.buf_in < 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) {
+            ret = decrypt_nal_unit(crypto_ctx, &nalu);
+            if (ret < 0)
+                return ret;
+        }
+        ctx.buf_out  += nalu.length;
+    }
+
+    av_shrink_packet(pkt, ctx.buf_out - 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_in < ctx->buf_end - 1) {
+        if (*ctx->buf_in == 0xFF && *(ctx->buf_in + 1) & 0xF0 == 0xF0)
+            break;
+        ctx->buf_in++;
+    }
+
+    if (ctx->buf_in >= ctx->buf_end - 1)
+        return -1;
+
+    frame->data = (uint8_t*)ctx->buf_in;
+
+    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_in < ctx->buf_end - 1) {
+        if (*ctx->buf_in == 0x0B && *(ctx->buf_in + 1) == 0x77)
+            break;
+        ctx->buf_in++;
+    }
+
+    if (ctx->buf_in >= ctx->buf_end - 1)
+        return -1;
+
+    frame->data = (uint8_t*)ctx->buf_in;
+    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;
+    uint8_t *decrypted_data;
+    int num_of_encrypted_blocks;
+
+    struct AVAES *aes_ctx = av_aes_alloc();
+    if (!aes_ctx)
+        return AVERROR(ENOMEM);
+
+    ret = av_aes_init(aes_ctx, crypto_ctx->key, 16 * 8, 1);
+    if (ret < 0) {
+        av_free(aes_ctx);
+        return ret;
+    }
+
+    data = frame->data + frame->header_length + 16;
+
+    num_of_encrypted_blocks = (frame->length - frame->header_length - 16)/16;
+
+    decrypted_data = av_mallocz(num_of_encrypted_blocks*16);
+    if (!decrypted_data)
+        return AVERROR(ENOMEM);
+
+    av_aes_crypt(aes_ctx, decrypted_data, data, num_of_encrypted_blocks, crypto_ctx->iv, 1);
+
+    if (codec_id == AV_CODEC_ID_EAC3)
+        memcpy(crypto_ctx->iv, data + (num_of_encrypted_blocks - 1)*16, 16);
+
+    memcpy(data, decrypted_data, num_of_encrypted_blocks*16);
+
+    av_free(decrypted_data);
+    av_free(aes_ctx);
+
+    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_in 	= pkt->data;
+    ctx.buf_end = pkt->data + pkt->size;
+
+    while (ctx.buf_in < 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_in += 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..ccc506161c
--- /dev/null
+++ b/libavformat/hls_sample_aes.h
@@ -0,0 +1,64 @@ 
+/*
+ * 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"
+
+#define HLS_MAX_ID3_TAGS_DATA_LEN       138
+#define HLS_MAX_AUDIO_SETUP_DATA_LEN    10
+
+
+typedef struct HLSCryptoContext {
+    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;