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
From 90211198a936ca7087dbf04e5d636fd9992a8332 Mon Sep 17 00:00:00 2001
From: Jerome Martinez <jer...@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
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