Hi!
This patch adds support for Audible AA files.
Audible samples can be obtained from,
https://gitlab.com/vesselin.bontchev/audible-samples/tree/master
https://samples.ffmpeg.org/audible/
Currently, this code generates corrupt audio output (in some places, and
deterministically).
By posting this patch, I am hoping to get some early feedback, and help in
making this work.
Thanks,
Vesselin
From ba73543efc3fdc4b5c61e9cb56f998d748716e00 Mon Sep 17 00:00:00 2001
From: Vesselin Bontchev <vesselin.bontc...@yandex.com>
Date: Sun, 19 Jul 2015 23:16:36 +0200
Subject: [PATCH] Add support for Audible AA files
https://en.wikipedia.org/wiki/Audible.com#Quality
---
doc/demuxers.texi | 10 ++
doc/general.texi | 2 +
libavformat/Makefile | 1 +
libavformat/aadec.c | 381 +++++++++++++++++++++++++++++++++++++++++++++++
libavformat/allformats.c | 1 +
5 files changed, 395 insertions(+)
create mode 100644 libavformat/aadec.c
diff --git a/doc/demuxers.texi b/doc/demuxers.texi
index e45e1af..df95233 100644
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -18,6 +18,16 @@ enabled demuxers.
The description of some of the currently available demuxers follows.
+@section aa
+
+Audible Format 2, 3, and 4 demuxer.
+
+This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+
+@example
+ffmpeg -v debug -i input.aa -c:a copy output.wav
+@end example
+
@section applehttp
Apple HTTP Live Streaming demuxer.
diff --git a/doc/general.texi b/doc/general.texi
index a260e79..2b782e0 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -228,6 +228,8 @@ library:
@item 8088flex TMV @tab @tab X
@item AAX @tab @tab X
@tab Audible Enhanced Audio format, used in audiobooks.
+@item AA @tab @tab X
+ @tab Audible Format 2, 3, and 4, used in audiobooks.
@item ACT Voice @tab @tab X
@tab contains G.729 audio
@item Adobe Filmstrip @tab X @tab X
diff --git a/libavformat/Makefile b/libavformat/Makefile
index cc73fd8..8405f8d 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -395,6 +395,7 @@ OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o rtspdec.o httpauth.o \
OBJS-$(CONFIG_RTSP_MUXER) += rtsp.o rtspenc.o httpauth.o \
urldecode.o
OBJS-$(CONFIG_SAMI_DEMUXER) += samidec.o subtitles.o
+OBJS-$(CONFIG_AA_DEMUXER) += aadec.o
OBJS-$(CONFIG_SAP_DEMUXER) += sapdec.o
OBJS-$(CONFIG_SAP_MUXER) += sapenc.o
OBJS-$(CONFIG_SBG_DEMUXER) += sbgdec.o
diff --git a/libavformat/aadec.c b/libavformat/aadec.c
new file mode 100644
index 0000000..bd994e8
--- /dev/null
+++ b/libavformat/aadec.c
@@ -0,0 +1,381 @@
+/*
+ * Audible AA demuxer
+ * Copyright (c) 2015 Vesselin Bontchev
+ *
+ * Inspired by https://github.com/jteeuwen/audible, and
+ * https://code.google.com/p/pyaudibletags/source projects.
+ *
+ * See https://en.wikipedia.org/wiki/Audible.com#Quality for
+ * format details.
+ *
+ * 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
+ */
+
+#include "avformat.h"
+#include "internal.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/tea.h"
+#include "libavutil/fifo.h"
+#include "libavutil/opt.h"
+
+#define AA_MAGIC 1469084982 /* this identifies an audible .aa file */
+#define MAX_CODEC_SECOND_SIZE 3982
+#define MAX_TOC_ENTRIES 64
+#define TEA_BLOCK_SIZE 8
+
+typedef struct AADemuxContext {
+ int64_t data_end;
+ uint32_t HeaderSeed;
+ uint32_t Filesize;
+ uint32_t Magic;
+ uint32_t tocSize;
+ uint32_t npairs;
+ union {
+ unsigned char key[16];
+ uint32_t part[4];
+ } HeaderKey;
+ uint32_t TOC[MAX_TOC_ENTRIES][2];
+
+ void *aa_fixed_key;
+ int aa_fixed_key_size;
+ uint32_t start;
+ uint32_t end;
+ int32_t codec_second_size;
+ char codec_name[64];
+ struct AVTEA *tea_ctx;
+ uint8_t final_key[16];
+ int64_t current_chapter_size;
+ int64_t current_codec_second_size;
+ uint32_t trailing_bytes;
+ int64_t total_parsed;
+ struct AVFifoBuffer *cbuf;
+ int chapter_idx;
+} AADemuxContext;
+
+static int32_t GetSecondSizeByCodecID(char *codec_name)
+{
+ int32_t result = 0;
+
+ if (!strcmp(codec_name, "mp332")) {
+ // 0xC00D (marker in file)
+ result = 3982;
+ } else if (!strcmp(codec_name, "acelp85")) {
+ // 0xC00C
+ result = 2000;
+ } else if (!strcmp(codec_name, "acelp85")) {
+ // 0xC010
+ result = 1045;
+ }
+
+ return result;
+}
+
+static void readString(AVIOContext *pb, int size, char *output)
+{
+ avio_read(pb, output, size);
+}
+
+static int aa_read_header(AVFormatContext *s)
+{
+ AVStream *st;
+ AADemuxContext *c;
+ AVIOContext *pb;
+
+ int i, j, idx;
+ uint32_t nkey;
+ uint32_t nval;
+ char key[512] = {0};
+ char val[512] = {0};
+ unsigned char buffer[512] = {0};
+ uint32_t v0, v1;
+ unsigned char dst[8];
+ unsigned char src[8];
+ unsigned char *output = buffer;
+ int largest_idx = -1;
+ int64_t largest_size = -1;
+ int64_t current_size = -1;
+
+ c = s->priv_data;
+ pb = s->pb;
+ c->tea_ctx = av_tea_alloc();
+ if (!c->tea_ctx)
+ return AVERROR(ENOMEM);
+ c->cbuf = av_fifo_alloc(MAX_CODEC_SECOND_SIZE * 2);
+ if (!c->cbuf)
+ return AVERROR(ENOMEM);
+
+ c->Filesize = avio_rb32(pb); // file size
+ c->Magic = avio_rb32(pb); // magic string
+ c->tocSize = avio_rb32(pb); // TOC size
+ avio_rb32(pb); // unidentified integer
+ for (i = 0; i < c->tocSize; i++) { // read TOC
+ avio_rb32(pb); // TOC entry index
+ c->TOC[i][0] = avio_rb32(pb); // block Offset
+ c->TOC[i][1] = avio_rb32(pb); // block size
+ }
+ avio_read(pb, buffer, 24); // header termination block (ignored)
+ c->npairs = avio_rb32(pb); // read dictionary entries
+ for (i = 0; i < c->npairs; i++) {
+ memset(val, 0, sizeof(val));
+ memset(key, 0, sizeof(key));
+ avio_r8(pb); // unidentified byte
+ nkey = avio_rb32(pb); // key string length
+ nval = avio_rb32(pb); // value string length
+ if (nkey > 512) {
+ avio_seek(pb, nkey, SEEK_CUR);
+ } else {
+ readString(pb, nkey, key); // key string
+ }
+ if (nval > 512) {
+ avio_seek(pb, nval, SEEK_CUR);
+ } else {
+ readString(pb, nval, val); // value string.
+ }
+ if (!strncmp(key, "codec", 5)) {
+ strncpy(c->codec_name, val, sizeof(c->codec_name) - 1);
+ }
+ if (!strncmp(key, "HeaderSeed", 10)) {
+ c->HeaderSeed = atoi(val);
+ }
+ if (!strncmp(key, "HeaderKey", 9)) {
+ j = 0;
+ for (idx = 0; idx < 4; idx++) {
+ c->HeaderKey.part[idx] = (uint32_t)atoi(val + j);
+ c->HeaderKey.part[idx] = AV_RB32(c->HeaderKey.part + idx); // convert to BE!
+ if (idx == 3)
+ break;
+ while (val[j] != ' ') // find the next space
+ j++;
+ j = j + 1; // skip over the space
+ }
+ }
+ }
+
+ /* decryption key derivation */
+ av_tea_init(c->tea_ctx, c->aa_fixed_key, 16);
+ c->codec_second_size = GetSecondSizeByCodecID(c->codec_name);
+ if (c->codec_second_size == -1) {
+ return AVERROR_DECODER_NOT_FOUND;
+ }
+ memcpy(output + 0, "\x00\x00", 2); // purely for padding purposes
+ memcpy(output + 2, &c->HeaderKey, 16);
+ idx = 0;
+ for (i = 0; i < 3 * TEA_BLOCK_SIZE; i += TEA_BLOCK_SIZE) {
+ v0 = c->HeaderSeed;
+ v1 = c->HeaderSeed + 1;
+ AV_WB32(src, v0);
+ AV_WB32(src + 4, v1);
+ c->HeaderSeed = v1 + 1;
+ av_tea_crypt(c->tea_ctx, dst, src, TEA_BLOCK_SIZE, NULL, 0); // TEA ECB encrypt
+ for (j = 0; j < 8 && idx < 18; j+=1, idx+=1) {
+ output[idx] = output[idx] ^ dst[j];
+ }
+ }
+ memcpy(c->final_key, output + 2, 16); // skip first 2 bytes of output
+
+ /* decoder setup */
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ if (!strcmp(c->codec_name, "mp332")) {
+ st->codec->codec_id = AV_CODEC_ID_MP3;
+ } else if (!strcmp(c->codec_name, "acelp85")) {
+ st->codec->codec_id = AV_CODEC_ID_SIPR;
+ st->codec->block_align = 19;
+ st->codec->channels = 1;
+ st->codec->sample_rate = 8500;
+ } else if (!strcmp(c->codec_name, "acelp16")) {
+ st->codec->codec_id = AV_CODEC_ID_SIPR;
+ st->codec->block_align = 20;
+ st->codec->channels = 1;
+ st->codec->sample_rate = 16000;
+ } else {
+ return AVERROR_DECODER_NOT_FOUND;
+ }
+
+ /* determine, and jump to audio start offset */
+ for (i = 1; i < c->tocSize; i++) { // skip the first entry!
+ current_size = c->TOC[i][1];
+ if (current_size > largest_size) {
+ largest_idx = i;
+ largest_size = current_size;
+ }
+ }
+ c->start = c->TOC[largest_idx][0];
+ c->end = c->TOC[largest_idx][1];
+ avio_seek(pb, c->start, SEEK_SET);
+ c->current_chapter_size = 0;
+
+ return 0;
+}
+
+static int aa_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ AADemuxContext *c;
+ unsigned char dst[8];
+ unsigned char src[8];
+ int i;
+ int trailing_bytes;
+ int blocks;
+ uint8_t buf[MAX_CODEC_SECOND_SIZE * 2];
+ int written = 0;
+ int queued_size;
+ int ret = 0;
+ int shortfall;
+ int extra;
+
+ c = s->priv_data;
+
+ // do we already have some audio ouput from the last invocation?
+ queued_size = av_fifo_size(c->cbuf);
+ if (queued_size) {
+ // av_log(s, AV_LOG_DEBUG, "[aa] draining queued data (%d bytes)\n", queued_size);
+ av_fifo_generic_read(c->cbuf, buf, queued_size, NULL);
+ written = written + queued_size;
+ }
+
+ // are we at the start of a chapter?
+ if (c->current_chapter_size == 0) {
+ c->current_chapter_size = avio_rb32(s->pb);
+ if (c->current_chapter_size == 0) {
+ ret = AVERROR_EOF;
+ goto end;
+ }
+ av_log(s, AV_LOG_DEBUG, "[aa] chapter %d (%ld bytes)\n", c->chapter_idx, c->current_chapter_size);
+ c->chapter_idx = c->chapter_idx + 1;
+ avio_rb32(s->pb); // data start offset
+ c-> current_codec_second_size = c->codec_second_size;
+ }
+
+ // is this the last block in this chapter?
+ if (c->current_chapter_size / c->current_codec_second_size == 0) {
+ c->current_codec_second_size = c->current_chapter_size % c->current_codec_second_size;
+ }
+
+ // decrypt c->current_codec_second_size bytes
+ blocks = c->current_codec_second_size / TEA_BLOCK_SIZE;
+ for (i = 0; i < blocks; i++) {
+ avio_read(s->pb, src, TEA_BLOCK_SIZE);
+ av_tea_init(c->tea_ctx, c->final_key, 16);
+ av_tea_crypt(c->tea_ctx, dst, src, 1, NULL, 1);
+ memcpy(buf + written, dst, TEA_BLOCK_SIZE);
+ written = written + TEA_BLOCK_SIZE;
+ }
+ trailing_bytes = c->current_codec_second_size % TEA_BLOCK_SIZE;
+ if (trailing_bytes != 0) { // trailing bytes are left unencrypted!
+ avio_read(s->pb, src, trailing_bytes);
+ memcpy(buf + written, dst, trailing_bytes);
+ written = written + trailing_bytes;
+ }
+
+ // update state
+ c->current_chapter_size = c->current_chapter_size - c->current_codec_second_size;
+ if (c->current_chapter_size <= 0)
+ c->current_chapter_size = 0;
+
+ if (written < c->codec_second_size) { // don't have enough output for the decoder, pull in data from the next chapter
+ shortfall = c->codec_second_size - written;
+ c->current_chapter_size = avio_rb32(s->pb);
+ if (c->current_chapter_size == 0) {
+ ret = AVERROR_EOF;
+ goto end;
+ }
+ av_log(s, AV_LOG_DEBUG, "[aa] chapter %d (%ld bytes)\n", c->chapter_idx, c->current_chapter_size);
+ c->chapter_idx = c->chapter_idx + 1;
+ avio_rb32(s->pb); // data start offset (ignored)
+ c->current_codec_second_size = c->codec_second_size;
+ blocks = c->current_codec_second_size / TEA_BLOCK_SIZE;
+ for (i = 0; i < blocks; i++) {
+ avio_read(s->pb, src, 8);
+ av_tea_init(c->tea_ctx, c->final_key, 16);
+ av_tea_crypt(c->tea_ctx, dst, src, 1, NULL, 1);
+ av_fifo_generic_write(c->cbuf, dst, TEA_BLOCK_SIZE, NULL);
+ }
+ trailing_bytes = c->current_codec_second_size % TEA_BLOCK_SIZE;
+ if (trailing_bytes != 0) {
+ avio_read(s->pb, src, trailing_bytes);
+ av_fifo_generic_write(c->cbuf, src, trailing_bytes, NULL);
+ }
+ c->current_chapter_size = c->current_chapter_size - c->current_codec_second_size;
+ if (c->current_chapter_size <= 0)
+ c->current_chapter_size = 0;
+
+ // borrow "shortfall" bytes
+ av_fifo_generic_read(c->cbuf, buf + written, shortfall, NULL);
+ written = written + shortfall;
+ }
+
+ // give only codec_second_size bytes to the decoder
+ if (written > c->codec_second_size) { // c->cbuf drain triggers this
+ extra = written - c->codec_second_size;
+ av_fifo_generic_write(c->cbuf, buf + c->codec_second_size, extra, NULL);
+ written = c->codec_second_size;
+ }
+
+ av_init_packet(pkt);
+ pkt->data = NULL;
+ pkt->size = 0;
+ pkt->pos = avio_tell(s->pb);
+
+ // av_log(s, AV_LOG_DEBUG, "[aa] giving %d bytes to decoder\n", written);
+ av_grow_packet(pkt, written);
+ memcpy(pkt->data, buf, written);
+
+end:
+ return ret;
+}
+
+static int aa_probe(AVProbeData *p)
+{
+ unsigned char *buf = p->buf;
+
+ // first 4 bytes are file size, next 4 bytes are the magic
+ if (AV_RB32(buf+4) != AA_MAGIC)
+ return AVPROBE_SCORE_EXTENSION;
+
+ return AVPROBE_SCORE_MAX;
+}
+
+#define OFFSET(x) offsetof(AADemuxContext, x)
+#define FLAGS AV_OPT_FLAG_DECODING_PARAM
+static const AVOption aa_options[] = {
+ { "aa_fixed_key", // extracted from libAAX_SDK.so and AAXSDKWin.dll files!
+ "Fixed key used for handling Audible AA files", OFFSET(aa_fixed_key),
+ AV_OPT_TYPE_BINARY, {.str="77214d4b196a87cd520045fd2a51d673"},
+ .flags = AV_OPT_FLAG_DECODING_PARAM },
+ { NULL },
+};
+
+static const AVClass aa_class = {
+ .class_name = "aa",
+ .item_name = av_default_item_name,
+ .option = aa_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVInputFormat ff_aa_demuxer = {
+ .name = "aa",
+ .long_name = NULL_IF_CONFIG_SMALL("Audible AA format files"),
+ .priv_class = &aa_class,
+ .priv_data_size = sizeof(AADemuxContext),
+ .extensions = "aa",
+ .read_probe = aa_probe,
+ .read_header = aa_read_header,
+ .read_packet = aa_read_packet,
+ .flags = AVFMT_GENERIC_INDEX,
+};
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 181cb9e..60ec0ca 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -262,6 +262,7 @@ void av_register_all(void)
REGISTER_MUXER (RTP_MPEGTS, rtp_mpegts);
REGISTER_MUXDEMUX(RTSP, rtsp);
REGISTER_DEMUXER (SAMI, sami);
+ REGISTER_DEMUXER (AA, aa);
REGISTER_MUXDEMUX(SAP, sap);
REGISTER_DEMUXER (SBG, sbg);
REGISTER_DEMUXER (SDP, sdp);
--
2.1.4
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel