diff mbox series

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

Message ID SG2PR01MB269339627C977C841E26B05DF2020@SG2PR01MB2693.apcprd01.prod.exchangelabs.com
State New
Headers show
Series [FFmpeg-devel] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
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Context Check Description
andriy/x86_make fail Make failed
andriy/PPC64_make warning Make failed

Commit Message

Nachiket Tarate Oct. 15, 2020, 4:45 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.tarate@outlook.com>
---
 libavformat/Makefile         |   2 +-
 libavformat/hls.c            |  93 ++++++-
 libavformat/hls_sample_aes.c | 497 +++++++++++++++++++++++++++++++++++
 libavformat/hls_sample_aes.h |  64 +++++
 libavformat/mpegts.c         |  15 ++
 5 files changed, 657 insertions(+), 14 deletions(-)
 create mode 100644 libavformat/hls_sample_aes.c
 create mode 100644 libavformat/hls_sample_aes.h

Comments

Michael Niedermayer Oct. 15, 2020, 6:05 p.m. UTC | #1
On Thu, Oct 15, 2020 at 10:15:13PM +0530, Nachiket Tarate 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.tarate@outlook.com>
> ---
>  libavformat/Makefile         |   2 +-
>  libavformat/hls.c            |  93 ++++++-
>  libavformat/hls_sample_aes.c | 497 +++++++++++++++++++++++++++++++++++
>  libavformat/hls_sample_aes.h |  64 +++++
>  libavformat/mpegts.c         |  15 ++
>  5 files changed, 657 insertions(+), 14 deletions(-)
>  create mode 100644 libavformat/hls_sample_aes.c
>  create mode 100644 libavformat/hls_sample_aes.h

This seems to break fate (segfault)
I guess patchwork will notice it too but as i already tested and noticed ...

--- ./tests/ref/fate/segment-mp4-to-ts	2020-10-10 18:08:06.500253003 +0200
+++ tests/data/fate/segment-mp4-to-ts	2020-10-15 20:03:24.586303460 +0200
@@ -128,5 +128,3 @@
 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0, S=1,        1, 0x00e000e0
 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0, S=1,        1, 0x00e000e0
 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0, S=1,        1, 0x00e000e0
-0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0, S=1,        1, 0x00e000e0
-0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
Test segment-mp4-to-ts failed. Look at tests/data/fate/segment-mp4-to-ts.err for details.
tests/Makefile:255: recipe for target 'fate-segment-mp4-to-ts' failed
make: *** [fate-segment-mp4-to-ts] Error 139


[...]
Steven Liu Oct. 19, 2020, 2:13 a.m. UTC | #2
Nachiket Tarate <nachiket.tarate@outlook.com> 于2020年10月18日周日 上午8:07写道:
>
> _______________________________________
> From: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> on behalf of Michael Niedermayer <michael@niedermayer.cc>
> Sent: Thursday, October 15, 2020 11:35 PM
> To: FFmpeg development discussions and patches
> Subject: Re: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
>
> On Thu, Oct 15, 2020 at 10:15:13PM +0530, Nachiket Tarate 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.tarate@outlook.com>
> > ---
> >  libavformat/Makefile         |   2 +-
> >  libavformat/hls.c            |  93 ++++++-
> >  libavformat/hls_sample_aes.c | 497 +++++++++++++++++++++++++++++++++++
> >  libavformat/hls_sample_aes.h |  64 +++++
> >  libavformat/mpegts.c         |  15 ++
> >  5 files changed, 657 insertions(+), 14 deletions(-)
> >  create mode 100644 libavformat/hls_sample_aes.c
> >  create mode 100644 libavformat/hls_sample_aes.h
>
> This seems to break fate (segfault)
> I guess patchwork will notice it too but as i already tested and noticed ...
>
> --- ./tests/ref/fate/segment-mp4-to-ts  2020-10-10 18:08:06.500253003 +0200
> +++ tests/data/fate/segment-mp4-to-ts   2020-10-15 20:03:24.586303460 +0200
> @@ -128,5 +128,3 @@
>  0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0, S=1,        1, 0x00e000e0
>  0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0, S=1,        1, 0x00e000e0
>  0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0, S=1,        1, 0x00e000e0
> -0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0, S=1,        1, 0x00e000e0
> -0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
> Test segment-mp4-to-ts failed. Look at tests/data/fate/segment-mp4-to-ts.err for details.
> tests/Makefile:255: recipe for target 'fate-segment-mp4-to-ts' failed
> make: *** [fate-segment-mp4-to-ts] Error 139
>
>
> I ran FATE with samples again but I didn't get segfault. Can you please help me to reproduce it ?

https://patchwork.ffmpeg.org/project/ffmpeg/patch/SG2PR01MB269339627C977C841E26B05DF2020@SG2PR01MB2693.apcprd01.prod.exchangelabs.com/
fate failed message here.

TEST    segment-mp4-to-ts
--- /Users/liuqi/multimedia/upstream_ffmpeg/ffmpeg/tests/ref/fate/segment-mp4-to-ts
2020-10-19 09:24:15.000000000 +0800
+++ tests/data/fate/segment-mp4-to-ts 2020-10-19 10:09:43.000000000 +0800
@@ -128,5 +128,3 @@
 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0,
S=1,        1, 0x00e000e0
 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0,
S=1,        1, 0x00e000e0
 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0,
S=1,        1, 0x00e000e0
-0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0,
S=1,        1, 0x00e000e0
-0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
Test segment-mp4-to-ts failed. Look at
tests/data/fate/segment-mp4-to-ts.err for details.
make: *** [fate-segment-mp4-to-ts] Error 134
(base) liuqi05:ufbuild liuqi$ history |grep make | tail -n 5
  318  make -j6
  319  make fate-rsync
  320  make fate

(base) liuqi05:ufbuild liuqi$ ./ffmpeg
ffmpeg version N-99556-gf7e2f090ed Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.2)
  configuration: --cc=clang --quiet --enable-htmlpages
--enable-libx264 --enable-libxml2 --enable-gpl --extra-ldflags='-O0
-g3 -fsanitize=address -Wno-error -fPIC -I/usr/local/include'
--extra-ldflags='-O0 -g3 -fsanitize=address -Wno-error -fPIC
-L/usr/local/lib' --enable-libfreetype --enable-fontconfig
--enable-libspeex --enable-libopus --enable-libzmq --enable-libx265
--enable-libass --enable-videotoolbox --disable-optimizations
--enable-audiotoolbox --enable-opengl --disable-stripping
--samples=../../fate-suite/


need use samples --samples=../../fate-suite/
and make fate-rsync

after above two step, you can make fate reproduce it.
>
> --
> Best Regards,
> Nachiket Tarate

Thanks

Steven
Steven Liu Oct. 19, 2020, 9:47 a.m. UTC | #3
> 2020年10月19日 下午4:10,Nachiket Tarate <nachiket.tarate@outlook.com> 写道:
> 
> 
> 
> ________________________________________
> From: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> on behalf of Steven Liu <lingjiujianke@gmail.com>
> Sent: Monday, October 19, 2020 7:43 AM
> To: FFmpeg development discussions and patches
> Subject: Re: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
> 
> Nachiket Tarate <nachiket.tarate@outlook.com> 于2020年10月18日周日 上午8:07写道:
>> 
>> _______________________________________
>> From: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> on behalf of Michael Niedermayer <michael@niedermayer.cc>
>> Sent: Thursday, October 15, 2020 11:35 PM
>> To: FFmpeg development discussions and patches
>> Subject: Re: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
>> 
>> On Thu, Oct 15, 2020 at 10:15:13PM +0530, Nachiket Tarate 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.tarate@outlook.com>
>>> ---
>>> libavformat/Makefile         |   2 +-
>>> libavformat/hls.c            |  93 ++++++-
>>> libavformat/hls_sample_aes.c | 497 +++++++++++++++++++++++++++++++++++
>>> libavformat/hls_sample_aes.h |  64 +++++
>>> libavformat/mpegts.c         |  15 ++
>>> 5 files changed, 657 insertions(+), 14 deletions(-)
>>> create mode 100644 libavformat/hls_sample_aes.c
>>> create mode 100644 libavformat/hls_sample_aes.h
>> 
>> This seems to break fate (segfault)
>> I guess patchwork will notice it too but as i already tested and noticed ...
>> 
>> --- ./tests/ref/fate/segment-mp4-to-ts  2020-10-10 18:08:06.500253003 +0200
>> +++ tests/data/fate/segment-mp4-to-ts   2020-10-15 20:03:24.586303460 +0200
>> @@ -128,5 +128,3 @@
>> 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0, S=1,        1, 0x00e000e0
>> 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0, S=1,        1, 0x00e000e0
>> 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0, S=1,        1, 0x00e000e0
>> -0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0, S=1,        1, 0x00e000e0
>> -0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
>> Test segment-mp4-to-ts failed. Look at tests/data/fate/segment-mp4-to-ts.err for details.
>> tests/Makefile:255: recipe for target 'fate-segment-mp4-to-ts' failed
>> make: *** [fate-segment-mp4-to-ts] Error 139
>> 
>> 
>> I ran FATE with samples again but I didn't get segfault. Can you please help me to reproduce it ?
> 
> https://patchwork.ffmpeg.org/project/ffmpeg/patch/SG2PR01MB269339627C977C841E26B05DF2020@SG2PR01MB2693.apcprd01.prod.exchangelabs.com/
> fate failed message here.
> 
> TEST    segment-mp4-to-ts
> --- /Users/liuqi/multimedia/upstream_ffmpeg/ffmpeg/tests/ref/fate/segment-mp4-to-ts
> 2020-10-19 09:24:15.000000000 +0800
> +++ tests/data/fate/segment-mp4-to-ts 2020-10-19 10:09:43.000000000 +0800
> @@ -128,5 +128,3 @@
> 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0,
> S=1,        1, 0x00e000e0
> 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0,
> S=1,        1, 0x00e000e0
> 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0,
> S=1,        1, 0x00e000e0
> -0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0,
> S=1,        1, 0x00e000e0
> -0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
> Test segment-mp4-to-ts failed. Look at
> tests/data/fate/segment-mp4-to-ts.err for details.
> make: *** [fate-segment-mp4-to-ts] Error 134
> (base) liuqi05:ufbuild liuqi$ history |grep make | tail -n 5
>  318  make -j6
>  319  make fate-rsync
>  320  make fate
> 
> (base) liuqi05:ufbuild liuqi$ ./ffmpeg
> ffmpeg version N-99556-gf7e2f090ed Copyright (c) 2000-2020 the FFmpeg developers
>  built with Apple clang version 12.0.0 (clang-1200.0.32.2)
>  configuration: --cc=clang --quiet --enable-htmlpages
> --enable-libx264 --enable-libxml2 --enable-gpl --extra-ldflags='-O0
> -g3 -fsanitize=address -Wno-error -fPIC -I/usr/local/include'
> --extra-ldflags='-O0 -g3 -fsanitize=address -Wno-error -fPIC
> -L/usr/local/lib' --enable-libfreetype --enable-fontconfig
> --enable-libspeex --enable-libopus --enable-libzmq --enable-libx265
> --enable-libass --enable-videotoolbox --disable-optimizations
> --enable-audiotoolbox --enable-opengl --disable-stripping
> --samples=../../fate-suite/
> 
> 
> need use samples --samples=../../fate-suite/
> and make fate-rsync
> 
> after above two step, you can make fate reproduce it.
> 
> 
> Actually, I did
> 
> make fate-rsync SAMPLES=fate-suite/
> make fate SAMPLES=fate-suite/
> 
> It is same.
> 
> But segmentation fault didn't occur while executing segment-mp4-to-ts.
What your configure options?

> 
> Any idea ?
> 
> --
> Best Regards,
> Nachiket Tarate
> 
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> 
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".

Thanks

Steven Liu
Steven Liu Oct. 20, 2020, 2:25 a.m. UTC | #4
> 2020年10月19日 下午4:10,Nachiket Tarate <nachiket.tarate@outlook.com> 写道:
> 
> 
> 
> ________________________________________
> From: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> on behalf of Steven Liu <lingjiujianke@gmail.com>
> Sent: Monday, October 19, 2020 7:43 AM
> To: FFmpeg development discussions and patches
> Subject: Re: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
> 
> Nachiket Tarate <nachiket.tarate@outlook.com> 于2020年10月18日周日 上午8:07写道:
>> 
>> _______________________________________
>> From: ffmpeg-devel <ffmpeg-devel-bounces@ffmpeg.org> on behalf of Michael Niedermayer <michael@niedermayer.cc>
>> Sent: Thursday, October 15, 2020 11:35 PM
>> To: FFmpeg development discussions and patches
>> Subject: Re: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
>> 
>> On Thu, Oct 15, 2020 at 10:15:13PM +0530, Nachiket Tarate 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.tarate@outlook.com>
>>> ---
>>> libavformat/Makefile         |   2 +-
>>> libavformat/hls.c            |  93 ++++++-
>>> libavformat/hls_sample_aes.c | 497 +++++++++++++++++++++++++++++++++++
>>> libavformat/hls_sample_aes.h |  64 +++++
>>> libavformat/mpegts.c         |  15 ++
>>> 5 files changed, 657 insertions(+), 14 deletions(-)
>>> create mode 100644 libavformat/hls_sample_aes.c
>>> create mode 100644 libavformat/hls_sample_aes.h
>> 
>> This seems to break fate (segfault)
>> I guess patchwork will notice it too but as i already tested and noticed ...
>> 
>> --- ./tests/ref/fate/segment-mp4-to-ts  2020-10-10 18:08:06.500253003 +0200
>> +++ tests/data/fate/segment-mp4-to-ts   2020-10-15 20:03:24.586303460 +0200
>> @@ -128,5 +128,3 @@
>> 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0, S=1,        1, 0x00e000e0
>> 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0, S=1,        1, 0x00e000e0
>> 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0, S=1,        1, 0x00e000e0
>> -0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0, S=1,        1, 0x00e000e0
>> -0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
>> Test segment-mp4-to-ts failed. Look at tests/data/fate/segment-mp4-to-ts.err for details.
>> tests/Makefile:255: recipe for target 'fate-segment-mp4-to-ts' failed
>> make: *** [fate-segment-mp4-to-ts] Error 139
>> 
>> 
>> I ran FATE with samples again but I didn't get segfault. Can you please help me to reproduce it ?
> 
> https://patchwork.ffmpeg.org/project/ffmpeg/patch/SG2PR01MB269339627C977C841E26B05DF2020@SG2PR01MB2693.apcprd01.prod.exchangelabs.com/
> fate failed message here.
> 
> TEST    segment-mp4-to-ts
> --- /Users/liuqi/multimedia/upstream_ffmpeg/ffmpeg/tests/ref/fate/segment-mp4-to-ts
> 2020-10-19 09:24:15.000000000 +0800
> +++ tests/data/fate/segment-mp4-to-ts 2020-10-19 10:09:43.000000000 +0800
> @@ -128,5 +128,3 @@
> 0,     428400,     435600,     3600,      156, 0xd2c3406c, F=0x0,
> S=1,        1, 0x00e000e0
> 0,     432000,     439200,     3600,      330, 0x150d9b60, F=0x0,
> S=1,        1, 0x00e000e0
> 0,     435600,     446400,     3600,      324, 0x558194ee, F=0x0,
> S=1,        1, 0x00e000e0
> -0,     439200,     442800,     3600,      191, 0x108e54d1, F=0x0,
> S=1,        1, 0x00e000e0
> -0,     442800,     450000,     3600,      233, 0xac5b6486, F=0x0
> Test segment-mp4-to-ts failed. Look at
> tests/data/fate/segment-mp4-to-ts.err for details.
> make: *** [fate-segment-mp4-to-ts] Error 134
> (base) liuqi05:ufbuild liuqi$ history |grep make | tail -n 5
>  318  make -j6
>  319  make fate-rsync
>  320  make fate
> 
> (base) liuqi05:ufbuild liuqi$ ./ffmpeg
> ffmpeg version N-99556-gf7e2f090ed Copyright (c) 2000-2020 the FFmpeg developers
>  built with Apple clang version 12.0.0 (clang-1200.0.32.2)
>  configuration: --cc=clang --quiet --enable-htmlpages
> --enable-libx264 --enable-libxml2 --enable-gpl --extra-ldflags='-O0
> -g3 -fsanitize=address -Wno-error -fPIC -I/usr/local/include'
> --extra-ldflags='-O0 -g3 -fsanitize=address -Wno-error -fPIC
> -L/usr/local/lib' --enable-libfreetype --enable-fontconfig
> --enable-libspeex --enable-libopus --enable-libzmq --enable-libx265
> --enable-libass --enable-videotoolbox --disable-optimizations
> --enable-audiotoolbox --enable-opengl --disable-stripping
> --samples=../../fate-suite/
> 
> 
> need use samples --samples=../../fate-suite/
> and make fate-rsync
> 
> after above two step, you can make fate reproduce it.
> 
> 
> Actually, I did
> 
> make fate-rsync SAMPLES=fate-suite/
> make fate SAMPLES=fate-suite/
> 
> It is same.
> 
> But segmentation fault didn't occur while executing segment-mp4-to-ts.

make fate-segment-mp4-to-ts

What about just run this test?
> 
> Any idea ?
> 
> --
> Best Regards,
> Nachiket Tarate
> 
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> 
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".

Thanks

Steven Liu
diff mbox series

Patch

diff --git a/libavformat/Makefile b/libavformat/Makefile
index a5e8bddb87..0ccec2d281 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -235,7 +235,7 @@  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-$(CONFIG_HLS_MUXER)                 += hlsenc.o hlsplaylist.o
 OBJS-$(CONFIG_HNM_DEMUXER)               += hnm.o
 OBJS-$(CONFIG_ICO_DEMUXER)               += icodec.o
diff --git a/libavformat/hls.c b/libavformat/hls.c
index 72e28ab94f..63683e4742 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) 2020 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 */
@@ -1015,10 +1020,11 @@  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);
@@ -1034,6 +1040,9 @@  static void parse_id3(AVFormatContext *s, AVIOContext *pb,
                 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 +1085,7 @@  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 +1239,7 @@  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 +1255,10 @@  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';
@@ -1265,14 +1275,9 @@  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
      * without a HTTP server.
@@ -1940,6 +1945,7 @@  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 +1978,52 @@  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;
@@ -1990,6 +2040,8 @@  static int hls_read_header(AVFormatContext *s)
             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;
@@ -2018,9 +2070,14 @@  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) {
+                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;
+            }
         }
 
         pls->has_noheader_flag = !!(pls->ctx->ctx_flags & AVFMTCTX_NOHEADER);
@@ -2142,6 +2199,7 @@  static int hls_read_packet(AVFormatContext *s, AVPacket *pkt)
 
     for (i = 0; i < c->n_playlists; i++) {
         struct playlist *pls = c->playlists[i];
+        struct segment *seg = NULL;
         /* Make sure we've got one buffered packet from each open playlist
          * stream */
         if (pls->needed && !pls->pkt.data) {
@@ -2166,6 +2224,15 @@  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..41d501fde3
--- /dev/null
+++ b/libavformat/hls_sample_aes.c
@@ -0,0 +1,497 @@ 
+/*
+ * Apple HTTP Live Streaming Sample Decryption
+ *
+ * Copyright (c) 2020 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 Decryption
+ * https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
+ */
+
+#include "hls_sample_aes.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 AVParserContext {
+    const uint8_t   *buf_in;
+    const uint8_t   *buf_end;
+    uint8_t         *buf_out;
+    int             next_start_code_length;
+} AVParserContext;
+
+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);
+}
+
+/*
+ * Parse 'dec3' EC3SpecificBox
+ */
+static int parse_dec3(AC3HeaderInfo **phdr, const uint8_t *buf, size_t size)
+{
+    GetBitContext gb;
+    AC3HeaderInfo *hdr;
+    int err;
+
+    int data_rate, fscod, acmod, lfeon;
+
+    if (!*phdr)
+        *phdr = av_mallocz(sizeof(AC3HeaderInfo));
+    if (!*phdr)
+        return AVERROR(ENOMEM);
+    hdr = *phdr;
+
+    err = init_get_bits8(&gb, buf, size);
+    if (err < 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);
+
+    hdr->sample_rate = eac3_sample_rate_tab[fscod];
+
+    hdr->channel_layout = avpriv_ac3_channel_layout_tab[acmod];
+    if (lfeon)
+        hdr->channel_layout |= AV_CH_LOW_FREQUENCY;
+
+    hdr->channels = av_get_channel_layout_nb_channels(hdr->channel_layout);
+
+    hdr->bit_rate = data_rate*1000;
+
+    return 0;
+}
+
+int ff_hls_parse_audio_setup_info(AVStream *st, HLSAudioSetupInfo *info)
+{
+    int ret = 0;
+
+    AC3HeaderInfo *ac3hdr = NULL;
+
+    st->codecpar->codec_tag = info->codec_tag;
+
+    if (st->codecpar->codec_id == AV_CODEC_ID_AAC)
+        return 0;
+
+    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)
+        ret = avpriv_ac3_parse_header(&ac3hdr, info->setup_data, info->setup_data_length);
+    else if (st->codecpar->codec_id == AV_CODEC_ID_EAC3)
+        ret = parse_dec3(&ac3hdr, info->setup_data, info->setup_data_length);
+    else
+        return -1;
+
+    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);
+
+    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)
+{
+  int i;
+
+  for (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 (AVParserContext *ctx, NALUnit *nalu)
+{
+    int i;
+      int len = 0;
+    int nalu_start_offset = 0;
+
+    uint8_t *buf_out = ctx->buf_out;
+
+    if (ctx->next_start_code_length != 0) {
+        for (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) {
+        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;
+    AVParserContext  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 (AVParserContext *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 (AVParserContext *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, AVParserContext *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 -1;
+}
+
+
+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) {
+        return ret;
+    }
+
+    data = frame->data + frame->header_length + 16;
+
+    num_of_encrypted_blocks = (frame->length - frame->header_length - 16)/16;
+
+    decrypted_data = (uint8_t *)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;
+    AVParserContext  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 -1;
+}
diff --git a/libavformat/hls_sample_aes.h b/libavformat/hls_sample_aes.h
new file mode 100644
index 0000000000..7fb1d57a81
--- /dev/null
+++ b/libavformat/hls_sample_aes.h
@@ -0,0 +1,64 @@ 
+/*
+ * Apple HTTP Live Streaming Sample Decryption
+ *
+ * Copyright (c) 2020 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 Decryption
+ * https://developer.apple.com/library/ios/documentation/AudioVideo/Conceptual/HLS_Sample_Encryption
+ */
+
+#ifndef AVFORMAT_HLS_SAMPLE_DECRYPT_H
+#define AVFORMAT_HLS_SAMPLE_DECRYPT_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_DECRYPT_H */
+
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index 432b1c3ea2..cf274ca5ec 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -838,6 +838,16 @@  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   },
@@ -947,6 +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;
@@ -1987,6 +1999,9 @@  int ff_parse_mpeg2_descriptor(AVFormatContext *fc, AVStream *st, int stream_type
     case 0x05: /* registration descriptor */
         st->codecpar->codec_tag = bytestream_get_le32(pp);
         av_log(fc, AV_LOG_TRACE, "reg_desc=%.4s\n", (char *)&st->codecpar->codec_tag);
+        if (st->codecpar->codec_tag == MKTAG('a', 'p', 'a', 'd')) {
+            st->codecpar->codec_tag = bytestream_get_le32(pp);
+        }
         if (st->codecpar->codec_id == AV_CODEC_ID_NONE || st->request_probe > 0) {
             mpegts_find_stream_type(st, st->codecpar->codec_tag, REGD_types);
             if (st->codecpar->codec_tag == MKTAG('B', 'S', 'S', 'D'))