[FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
Nachiket Tarate
nachiket.tarate at outlook.com
Sat Oct 10 18:06:31 EEST 2020
@Martin Storsj? <martin at martin.st>
If you get time, kindly review this patch.
--
Best Regards,
Nachiket Tarate
________________________________
From: ffmpeg-devel <ffmpeg-devel-bounces at ffmpeg.org> on behalf of Nachiket Tarate <nachiket.tarate at outlook.com>
Sent: Saturday, October 10, 2020 8:30 PM
To: ffmpeg-devel at ffmpeg.org <ffmpeg-devel at ffmpeg.org>
Subject: [FFmpeg-devel] [PATCH] libavformat/hls: add support for SAMPLE-AES decryption in HLS demuxer
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 at outlook.com>
---
libavformat/Makefile | 2 +-
libavformat/hls.c | 132 ++++++---
libavformat/hls_sample_aes.c | 499 +++++++++++++++++++++++++++++++++++
libavformat/hls_sample_aes.h | 64 +++++
libavformat/mpegts.c | 15 ++
5 files changed, 679 insertions(+), 33 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 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..3cdbc6dd94 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, ×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;
@@ -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 +1275,10 @@ 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
* without a HTTP server.
@@ -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);
@@ -1972,24 +1979,70 @@ 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);
- 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;
- url = av_strdup(pls->segments[0]->url);
- ret = av_probe_input_buffer(&pls->pb, &in_fmt, url, NULL, 0, 0);
- av_free(url);
- if (ret < 0) {
- /* Free the ctx - it isn't initialized properly at this point,
- * so avformat_close_input shouldn't be called. If
- * avformat_open_input fails below, it frees and zeros the
- * context, so it doesn't need any special treatment like this. */
- av_log(s, AV_LOG_ERROR, "Error when loading first segment '%s'\n", pls->segments[0]->url);
- avformat_free_context(pls->ctx);
- pls->ctx = NULL;
- goto fail;
+
+ /*
+ * 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;
+ url = av_strdup(pls->segments[0]->url);
+ ret = av_probe_input_buffer(&pls->pb, &in_fmt, url, NULL, 0, 0);
+ av_free(url);
+ if (ret < 0) {
+ /* Free the ctx - it isn't initialized properly at this point,
+ * so avformat_close_input shouldn't be called. If
+ * avformat_open_input fails below, it frees and zeros the
+ * context, so it doesn't need any special treatment like this. */
+ av_log(s, AV_LOG_ERROR, "Error when loading first segment '%s'\n", pls->segments[0]->url);
+ 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;
@@ -2018,9 +2071,14 @@ 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)) {
- ret = avformat_find_stream_info(pls->ctx, NULL);
- if (ret < 0)
- goto fail;
+ 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 +2200,7 @@ 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) {
@@ -2165,7 +2224,16 @@ static int hls_read_packet(AVFormatContext *s, AVPacket *pkt)
c->first_timestamp = av_rescale_q(pls->pkt.dts,
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..c6c4c74247
--- /dev/null
+++ b/libavformat/hls_sample_aes.c
@@ -0,0 +1,499 @@
+/*
+ * 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..f3ce63188a
--- /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..83a9e18fdb 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'))
--
2.17.1
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