Re: [FFmpeg-devel] [PATCH] avformat: add DAT (Digital Audio Tape) demuxer

[Jerome Martinez <jerome@mediaarea.net>]

[-- Attachment #1: Type: text/plain, Size: 802 bytes --]

Le 25/01/2025 à 01:46, Michael Niedermayer a écrit :
> [...]
> this passes tests.
>
> but if you want, you could instead of testing "extra metadata (not needed for decoding)"
> test more than 1 packet
>
> having a best case score of 1 seems to be something that will
> likely fail sooner or later by not detecting a dat file

Maybe misunderstanding, the cdxl parser provides score of 1, the DAT 
patches provide a score of the count of detected frames so better.
But attached is a v2 of the 2nd patch, less compilation warnings, using 
less metadata not impacting the decoding and handling corner cases like 
bad metadata e.g. 16-bit 4-ch (impossible) and 32 kHz 12-bit 4-ch.
12-bit is not yet handled but the patch provides the detection of such 
file and a smooth rejection of the file.

Jérôme

[-- Attachment #2: 0002-avformat-dat-improve-DAT-demuxer.patch --]
[-- Type: text/plain, Size: 4208 bytes --]

From 90211198a936ca7087dbf04e5d636fd9992a8332 Mon Sep 17 00:00:00 2001
From: Jerome Martinez <jerome@mediaarea.net>
Date: Wed, 22 Jan 2025 16:08:18 +0100
Subject: [PATCH 2/4] avformat/dat: improve DAT demuxer

Less false positive detection
Better computation of data size with 12-bit
---
 libavformat/dat.c | 49 ++++++++++++++++++++++++++++++++++-------------
 1 file changed, 36 insertions(+), 13 deletions(-)

diff --git a/libavformat/dat.c b/libavformat/dat.c
index 37548a8a73..071af9a12a 100644
--- a/libavformat/dat.c
+++ b/libavformat/dat.c
@@ -26,8 +26,9 @@
 #define DAT_PACKET_SIZE 5822
 #define DAT_OFFSET 5760
 
-static const uint32_t encoded_rate[] = { 48000, 44100, 32000, 0 };
-static const uint16_t encoded_size[] = { 5760, 5292, 3840, 0 };
+static const uint16_t encoded_samples[] = { 1440, 1323, 960, 0 };
+static const uint8_t encoded_samples_mul[] = { 1, 2, 0, 0 };
+static const uint8_t encoded_quantization[] = { 16, 12, 0, 0 };
 static const uint8_t encoded_chans[] = { 2, 4, 0, 0 };
 static const enum AVCodecID encoded_codec[] = {
     AV_CODEC_ID_PCM_S16LE,
@@ -41,12 +42,26 @@ static int valid_frame(uint8_t *frame)
     uint8_t *mainid = subid+4;
     int chan_index = (mainid[0] >> 0) & 0x3;
     int rate_index = (mainid[0] >> 2) & 0x3;
+    int fmtid      = (mainid[0] >> 6) & 0x3;
+    int trackpitch = (mainid[1] >> 2) & 0x3;
     int enc_index  = (mainid[1] >> 6) & 0x3;
     int dataid     = (subid[0] >> 0) & 0xf;
-
-    if (dataid != 0 || encoded_codec[enc_index] == AV_CODEC_ID_NONE ||
+    int numpacks   = (subid[1] >> 0) & 0xf;
+    int pno1       = (subid[1] >> 4) & 0xf;
+    int pno3       = (subid[2] >> 0) & 0xf;
+    int pno2       = (subid[2] >> 4) & 0xf;
+    int pno = (pno1 << 8) | (pno2 << 4) | pno3;
+    int encoded_size = encoded_samples[rate_index] * encoded_samples_mul[trackpitch] * encoded_chans[chan_index] * encoded_quantization[enc_index] / 8;
+  
+    if (dataid != 0 ||
+        numpacks > 7 ||
+        pno == 0 ||
         encoded_chans[chan_index] == 0 ||
-        encoded_rate[rate_index] == 0)
+        encoded_samples[rate_index] == 0 ||
+        fmtid != 0 ||
+        encoded_samples_mul[trackpitch] == 0 ||
+        encoded_quantization[enc_index] == 0 ||
+        encoded_size > DAT_OFFSET)
         return 0;
 
     return 1;
@@ -62,7 +77,7 @@ static int read_probe(const AVProbeData *p)
 
         score += ret;
         if (ret == 0)
-            break;
+            return 0;
     }
 
     return FFMIN(score, AVPROBE_SCORE_MAX);
@@ -82,21 +97,29 @@ static int parse_frame(uint8_t *frame, AVCodecParameters *par)
     uint8_t *mainid = subid+4;
     int chan_index = (mainid[0] >> 0) & 0x3;
     int rate_index = (mainid[0] >> 2) & 0x3;
+    int fmtid      = (mainid[0] >> 6) & 0x3;
+    int trackpitch = (mainid[1] >> 2) & 0x3;
     int enc_index  = (mainid[1] >> 6) & 0x3;
     int dataid     = (subid[0] >> 0) & 0xf;
+    int encoded_size = 0;
+    int encoded_size = encoded_samples[rate_index] * encoded_samples_mul[trackpitch] * encoded_chans[chan_index] * encoded_quantization[enc_index] / 8;
 
     par->codec_type = AVMEDIA_TYPE_AUDIO;
     par->codec_id = encoded_codec[enc_index];
     av_channel_layout_default(&par->ch_layout, encoded_chans[chan_index]);
-    par->sample_rate = encoded_rate[rate_index];
-    par->bit_rate = (8LL * DAT_PACKET_SIZE * par->sample_rate) / FFMAX(1, av_get_audio_frame_duration2(par, encoded_size[rate_index]));
-
-    if (dataid != 0 || par->codec_id == AV_CODEC_ID_NONE ||
-        par->ch_layout.nb_channels <= 0 ||
-        par->sample_rate <= 0)
+    par->sample_rate = encoded_samples[rate_index] * 100 / 3;
+    par->bit_rate = (8LL * DAT_PACKET_SIZE * par->sample_rate) / FFMAX(1, av_get_audio_frame_duration2(par, encoded_size));
+
+    if (dataid != 0 ||
+        par->ch_layout.nb_channels == 0 ||
+        par->sample_rate == 0 ||
+        fmtid != 0 ||
+        encoded_samples_mul[trackpitch] == 0 ||
+        encoded_quantization[enc_index] == 0 ||
+        encoded_size > DAT_OFFSET)
         return AVERROR_INVALIDDATA;
 
-    return encoded_size[rate_index];
+    return encoded_size;
 }
 
 static int read_packet(AVFormatContext *s, AVPacket *pkt)
-- 
2.46.0.windows.1


[-- Attachment #3: Type: text/plain, Size: 251 bytes --]

_______________________________________________
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".