[FFmpeg-devel] [PATCH 1/8] avformat/http: Return EIO for prematurely broken connection

[FFmpeg-devel] [PATCH 1/8] avformat/http: Return EIO for prematurely broken connection

[Tomas Härdin]

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Rebased and trimmed down Spotify patchset. Does not include the mfra or
ID3v2 stuff since those are turning out to be bigger tasks

/Tomas

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From c808ca473529ca952c42f00d12aa50ade38850d8 Mon Sep 17 00:00:00 2001
From: Ulrik <ulrikm@spotify.com>
Date: Mon, 27 Jul 2020 11:46:56 +0200
Subject: [PATCH 1/8] avformat/http: Return EIO for prematurely broken
 connection

Currently, a prematurely broken connection normally leads to the same
EOF, as a completed successful transfer. However, enabling reconnect
changes this logic, and leads to the return of EIO.

This patch unifies that logic, leading to the return of EIO for premature
disconnect, regardless of setting of "reconnect".
---
 libavformat/http.c | 13 ++++++++++---
 1 file changed, 10 insertions(+), 3 deletions(-)

diff --git a/libavformat/http.c b/libavformat/http.c
index ec60bc0b17..65ea5d993c 100644
--- a/libavformat/http.c
+++ b/libavformat/http.c
@@ -19,6 +19,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include <stdbool.h>
+
 #include "config.h"
 #include "config_components.h"
 
@@ -1747,6 +1749,7 @@ static int http_read_stream(URLContext *h, uint8_t *buf, int size)
     read_ret = http_buf_read(h, buf, size);
     while (read_ret < 0) {
         uint64_t target = h->is_streamed ? 0 : s->off;
+        bool is_premature = s->filesize > 0 && s->off < s->filesize;
 
         if (read_ret == AVERROR_EXIT)
             break;
@@ -1754,9 +1757,13 @@ static int http_read_stream(URLContext *h, uint8_t *buf, int size)
         if (h->is_streamed && !s->reconnect_streamed)
             break;
 
-        if (!(s->reconnect && s->filesize > 0 && s->off < s->filesize) &&
-            !(s->reconnect_at_eof && read_ret == AVERROR_EOF))
-            break;
+        if (!(s->reconnect && is_premature) &&
+            !(s->reconnect_at_eof && read_ret == AVERROR_EOF)) {
+            if (is_premature)
+                return AVERROR(EIO);
+            else
+                break;
+        }
 
         if (reconnect_delay > s->reconnect_delay_max || (s->reconnect_max_retries >= 0 && conn_attempts > s->reconnect_max_retries) ||
             reconnect_delay_total > s->reconnect_delay_total_max)
-- 
2.39.5


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[FFmpeg-devel] [PATCH 2/8] libavcodec/wmadec: Return AVERROR_INVALIDDATA on decoding errors

[Tomas Härdin]

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[-- Attachment #2: 0002-libavcodec-wmadec-Return-AVERROR_INVALIDDATA-on-deco.patch --]
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From 18a64198487582e9ef3246e7490d919ee19af595 Mon Sep 17 00:00:00 2001
From: Jonathan Murray <jonathanmurray@spotify.com>
Date: Wed, 9 Jun 2021 12:00:24 +0200
Subject: [PATCH 2/8] libavcodec/wmadec: Return AVERROR_INVALIDDATA on decoding
 errors

WMA files that fail to decode due to incoherent block lengths and
frame lengths currently result in a "Operation not permitted".
After this change, they will instead result in "Invalid data found
when processing input".

Several other error cases are also changed from returning -1.

As we change the error propagation logic in wma_decode_frame and
wma_decode_superframe, previous occurrences of returning
AVERROR_INVALIDDATA are also affected by this. This includes
"total_gain overread" and a "channel exponents_initialized" check.

---

Tomas: changed some -1's to AVERROR_INVALIDDATA
---
 libavcodec/wmadec.c | 40 ++++++++++++++++++++++++----------------
 1 file changed, 24 insertions(+), 16 deletions(-)

diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c
index 3427e482dc..c24fff5522 100644
--- a/libavcodec/wmadec.c
+++ b/libavcodec/wmadec.c
@@ -368,7 +368,7 @@ static int decode_exp_vlc(WMACodecContext *s, int ch)
         if ((unsigned) last_exp + 60 >= FF_ARRAY_ELEMS(pow_tab)) {
             av_log(s->avctx, AV_LOG_ERROR, "Exponent out of range: %d\n",
                    last_exp);
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
         v  = ptab[last_exp];
         iv = iptab[last_exp];
@@ -439,8 +439,10 @@ static void wma_window(WMACodecContext *s, float *out)
 }
 
 /**
- * @return 0 if OK. 1 if last block of frame. return -1 if
- * unrecoverable error.
+ * @return
+ * 0 if OK.
+ * 1 if last block of frame.
+ * AVERROR if unrecoverable error.
  */
 static int wma_decode_block(WMACodecContext *s)
 {
@@ -468,7 +470,7 @@ static int wma_decode_block(WMACodecContext *s)
                 av_log(s->avctx, AV_LOG_ERROR,
                        "prev_block_len_bits %d out of range\n",
                        s->frame_len_bits - v);
-                return -1;
+                return AVERROR_INVALIDDATA;
             }
             s->prev_block_len_bits = s->frame_len_bits - v;
             v                      = get_bits(&s->gb, n);
@@ -476,7 +478,7 @@ static int wma_decode_block(WMACodecContext *s)
                 av_log(s->avctx, AV_LOG_ERROR,
                        "block_len_bits %d out of range\n",
                        s->frame_len_bits - v);
-                return -1;
+                return AVERROR_INVALIDDATA;
             }
             s->block_len_bits = s->frame_len_bits - v;
         } else {
@@ -489,7 +491,7 @@ static int wma_decode_block(WMACodecContext *s)
             av_log(s->avctx, AV_LOG_ERROR,
                    "next_block_len_bits %d out of range\n",
                    s->frame_len_bits - v);
-            return -1;
+            return AVERROR_INVALIDDATA;
         }
         s->next_block_len_bits = s->frame_len_bits - v;
     } else {
@@ -501,14 +503,14 @@ static int wma_decode_block(WMACodecContext *s)
 
     if (s->frame_len_bits - s->block_len_bits >= s->nb_block_sizes){
         av_log(s->avctx, AV_LOG_ERROR, "block_len_bits not initialized to a valid value\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     /* now check if the block length is coherent with the frame length */
     s->block_len = 1 << s->block_len_bits;
     if ((s->block_pos + s->block_len) > s->frame_len) {
         av_log(s->avctx, AV_LOG_ERROR, "frame_len overflow\n");
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     if (channels == 2)
@@ -590,7 +592,7 @@ static int wma_decode_block(WMACodecContext *s)
             if (s->channel_coded[ch]) {
                 if (s->use_exp_vlc) {
                     if (decode_exp_vlc(s, ch) < 0)
-                        return -1;
+                        return AVERROR_INVALIDDATA;
                 } else {
                     decode_exp_lsp(s, ch);
                 }
@@ -802,7 +804,7 @@ static int wma_decode_frame(WMACodecContext *s, float **samples,
     for (;;) {
         ret = wma_decode_block(s);
         if (ret < 0)
-            return -1;
+            return ret;
         if (ret)
             break;
     }
@@ -879,8 +881,10 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
                 return AVERROR_INVALIDDATA;
 
             if ((s->last_superframe_len + buf_size - 1) >
-                MAX_CODED_SUPERFRAME_SIZE)
+                MAX_CODED_SUPERFRAME_SIZE) {
+                ret = AVERROR_INVALIDDATA;
                 goto fail;
+            }
 
             q   = s->last_superframe + s->last_superframe_len;
             len = buf_size - 1;
@@ -911,14 +915,17 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
             av_log(avctx, AV_LOG_ERROR,
                    "Invalid last frame bit offset %d > buf size %d (%d)\n",
                    bit_offset, get_bits_left(&s->gb), buf_size);
+            ret = AVERROR_INVALIDDATA;
             goto fail;
         }
 
         if (s->last_superframe_len > 0) {
             /* add bit_offset bits to last frame */
             if ((s->last_superframe_len + ((bit_offset + 7) >> 3)) >
-                MAX_CODED_SUPERFRAME_SIZE)
+                MAX_CODED_SUPERFRAME_SIZE) {
+                ret = AVERROR_INVALIDDATA;
                 goto fail;
+            }
             q   = s->last_superframe + s->last_superframe_len;
             len = bit_offset;
             while (len > 7) {
@@ -937,7 +944,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
                 skip_bits(&s->gb, s->last_bitoffset);
             /* this frame is stored in the last superframe and in the
              * current one */
-            if (wma_decode_frame(s, samples, samples_offset) < 0)
+            if ((ret = wma_decode_frame(s, samples, samples_offset)) < 0)
                 goto fail;
             samples_offset += s->frame_len;
             nb_frames--;
@@ -954,7 +961,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
 
         s->reset_block_lengths = 1;
         for (i = 0; i < nb_frames; i++) {
-            if (wma_decode_frame(s, samples, samples_offset) < 0)
+            if ((ret = wma_decode_frame(s, samples, samples_offset)) < 0)
                 goto fail;
             samples_offset += s->frame_len;
         }
@@ -967,13 +974,14 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
         len               = buf_size - pos;
         if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0) {
             av_log(s->avctx, AV_LOG_ERROR, "len %d invalid\n", len);
+            ret = AVERROR_INVALIDDATA;
             goto fail;
         }
         s->last_superframe_len = len;
         memcpy(s->last_superframe, buf + pos, len);
     } else {
         /* single frame decode */
-        if (wma_decode_frame(s, samples, samples_offset) < 0)
+        if ((ret = wma_decode_frame(s, samples, samples_offset)) < 0)
             goto fail;
         samples_offset += s->frame_len;
     }
@@ -989,7 +997,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, AVFrame *frame,
 fail:
     /* when error, we reset the bit reservoir */
     s->last_superframe_len = 0;
-    return -1;
+    return ret;
 }
 
 static av_cold void flush(AVCodecContext *avctx)
-- 
2.39.5


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[FFmpeg-devel] [PATCH 3/8] libavformat/flacdec: Export samples md5 as metadata

[Tomas Härdin]

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[-- Attachment #2: 0003-libavformat-flacdec-Export-samples-md5-as-metadata.patch --]
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From e1c4dfa4cc7a574f6fac76c11591547d3cd90ad2 Mon Sep 17 00:00:00 2001
From: Mattias Wadman <wader@spotify.com>
Date: Mon, 11 Oct 2021 15:38:13 +0200
Subject: [PATCH 3/8] libavformat/flacdec: Export samples md5 as metadata

Will be used by mal to compare metadat md5 with decoded samples md5.

Part of fixing https://jira.spotify.net/browse/GOL-681
---
 libavformat/flacdec.c                     | 6 ++++++
 tests/ref/fate/cover-art-aiff-id3v2-remux | 5 +++--
 tests/ref/fate/cover-art-flac-remux       | 1 +
 tests/ref/fate/id3v2-utf16-bom            | 5 +++--
 4 files changed, 13 insertions(+), 4 deletions(-)

diff --git a/libavformat/flacdec.c b/libavformat/flacdec.c
index 3c317acaee..9f65c25864 100644
--- a/libavformat/flacdec.c
+++ b/libavformat/flacdec.c
@@ -33,6 +33,7 @@
 #include "replaygain.h"
 
 #define SEEKPOINT_SIZE 18
+#define MD5_BYTE_SIZE 16
 
 typedef struct FLACDecContext {
     FFRawDemuxerContext rawctx;
@@ -109,6 +110,7 @@ static int flac_read_header(AVFormatContext *s)
         if (metadata_type == FLAC_METADATA_TYPE_STREAMINFO) {
             uint32_t samplerate;
             uint64_t samples;
+            char md5hex[MD5_BYTE_SIZE*2+1]; // hex representation plus null terminator
 
             /* STREAMINFO can only occur once */
             if (found_streaminfo) {
@@ -133,6 +135,10 @@ static int flac_read_header(AVFormatContext *s)
                 if (samples > 0)
                     st->duration = samples;
             }
+
+            ff_data_to_hex(md5hex, st->codecpar->extradata+18, MD5_BYTE_SIZE, 1 /* lowercase */);
+            md5hex[sizeof(md5hex)-1] = '\0';
+            av_dict_set(&s->metadata, "samples_md5", md5hex, 0);
         } else if (metadata_type == FLAC_METADATA_TYPE_CUESHEET) {
             uint8_t isrc[13];
             uint64_t start;
diff --git a/tests/ref/fate/cover-art-aiff-id3v2-remux b/tests/ref/fate/cover-art-aiff-id3v2-remux
index a59ba37c65..57c5a9f387 100644
--- a/tests/ref/fate/cover-art-aiff-id3v2-remux
+++ b/tests/ref/fate/cover-art-aiff-id3v2-remux
@@ -1,5 +1,5 @@
-330ad2bf538e91a31752b38024461df1 *tests/data/fate/cover-art-aiff-id3v2-remux.aiff
-608914 tests/data/fate/cover-art-aiff-id3v2-remux.aiff
+d8caddad3af3879f5957f71adcaedd23 *tests/data/fate/cover-art-aiff-id3v2-remux.aiff
+608970 tests/data/fate/cover-art-aiff-id3v2-remux.aiff
 #tb 0: 1/44100
 #media_type 0: audio
 #codec_id 0: pcm_s16be
@@ -68,6 +68,7 @@ TAG:comment=Composer
 [/STREAM]
 [FORMAT]
 TAG:artist=Мельница
+TAG:samples_md5=03462731b2dba8e6a9da639b07054121
 TAG:RATING=0
 TAG:album=Ангелофрения
 TAG:title=Дороги
diff --git a/tests/ref/fate/cover-art-flac-remux b/tests/ref/fate/cover-art-flac-remux
index fa91975881..109bf7395e 100644
--- a/tests/ref/fate/cover-art-flac-remux
+++ b/tests/ref/fate/cover-art-flac-remux
@@ -90,6 +90,7 @@ TAG:comment=Publisher/Studio logotype
 TAG:title=White King Granulated Soap
 [/STREAM]
 [FORMAT]
+TAG:samples_md5=496206705f222f9a63bf23dc874d9d71
 TAG:major_brand=M4A 
 TAG:minor_version=0
 TAG:compatible_brands=M4A mp42isom
diff --git a/tests/ref/fate/id3v2-utf16-bom b/tests/ref/fate/id3v2-utf16-bom
index dd2566de2b..a1adb4730d 100644
--- a/tests/ref/fate/id3v2-utf16-bom
+++ b/tests/ref/fate/id3v2-utf16-bom
@@ -1,5 +1,5 @@
-9b8bfdf87a8d3d089819ef9f6f264ec4 *tests/data/fate/id3v2-utf16-bom.aiff
-885482 tests/data/fate/id3v2-utf16-bom.aiff
+b712d97d4ab0cfd77d79235c312c438a *tests/data/fate/id3v2-utf16-bom.aiff
+885538 tests/data/fate/id3v2-utf16-bom.aiff
 #tb 0: 1/90000
 #media_type 0: video
 #codec_id 0: mjpeg
@@ -31,6 +31,7 @@ TAG:comment=Other
 [/STREAM]
 [FORMAT]
 TAG:artist=Мельница
+TAG:samples_md5=03462731b2dba8e6a9da639b07054121
 TAG:RATING=0
 TAG:album=Ангелофрения
 TAG:title=Дороги
-- 
2.39.5


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[FFmpeg-devel] [PATCH 4/8] avformat/flacdec: Return correct error-codes on read-failure

[Tomas Härdin]

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[-- Attachment #2: 0004-avformat-flacdec-Return-correct-error-codes-on-read-.patch --]
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From c81e350d5419cf02f029ce006d94f257bc18fb97 Mon Sep 17 00:00:00 2001
From: Ulrik <ulrikm@spotify.com>
Date: Thu, 26 Jan 2023 17:51:02 +0100
Subject: [PATCH 4/8] avformat/flacdec: Return correct error-codes on
 read-failure

Forward errors from `avio_read` directly. When `avio_read` sees EOF before
expected bytes can be read, consistently return `AVERROR_INVALIDDATA`

We used to return `AVERROR(AVERROR_INVALIDDATA)` when failing to read
metadata block headers. `AVERROR_INVALIDDATA` is already negative, so
wrapping in `AVERROR` leads to double-negation.

We used to return `AVERROR(EIO)` when failing to read extended metadata.
However, many times, the IO-layer is not at fault, the input data is simply
corrupted (truncated), so we return `AVERROR_INVALIDDATA` here as well.

---

Tomas: changed to use AVERROR_EOF
---
 libavformat/flacdec.c | 16 ++++++++++++----
 1 file changed, 12 insertions(+), 4 deletions(-)

diff --git a/libavformat/flacdec.c b/libavformat/flacdec.c
index 9f65c25864..77dcc620a4 100644
--- a/libavformat/flacdec.c
+++ b/libavformat/flacdec.c
@@ -81,8 +81,13 @@ static int flac_read_header(AVFormatContext *s)
 
     /* process metadata blocks */
     while (!avio_feof(s->pb) && !metadata_last) {
-        if (avio_read(s->pb, header, 4) != 4)
-            return AVERROR_INVALIDDATA;
+        ret = avio_read(s->pb, header, 4);
+        if (ret < 0) {
+            return ret;
+        } else if (ret != 4) {
+            return AVERROR_EOF;
+        }
+
         flac_parse_block_header(header, &metadata_last, &metadata_type,
                                    &metadata_size);
         switch (metadata_type) {
@@ -96,8 +101,11 @@ static int flac_read_header(AVFormatContext *s)
             if (!buffer) {
                 return AVERROR(ENOMEM);
             }
-            if (avio_read(s->pb, buffer, metadata_size) != metadata_size) {
-                RETURN_ERROR(AVERROR(EIO));
+            ret = avio_read(s->pb, buffer, metadata_size);
+            if (ret < 0) {
+                RETURN_ERROR(ret);
+            } else if (ret != metadata_size) {
+                RETURN_ERROR(AVERROR_EOF);
             }
             break;
         /* skip metadata block for unsupported types */
-- 
2.39.5


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[FFmpeg-devel] [PATCH 5/8] rtmp: Set correct message stream id when writing as server

[Tomas Härdin]

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This one is difficult to test. Any ideas?

/Tomas

[-- Attachment #2: 0005-rtmp-Set-correct-message-stream-id-when-writing-as-s.patch --]
[-- Type: text/x-patch, Size: 1761 bytes --]

From c8689abcbf9bf85e1f7775a347b6bc994679cb77 Mon Sep 17 00:00:00 2001
From: Jonathan Murray <jonathanmurray@spotify.com>
Date: Thu, 31 Mar 2022 16:23:17 +0200
Subject: [PATCH 5/8] rtmp: Set correct message stream id when writing as
 server

rtmp_write is used both for writing outputs as a server. The
rt->listen flag determines which mode we're running in.

Previously, when running as a server, the message stream id would
always be set to 0 for media/metadata messages. This is surprising
given that we have both responded to "createStream()" with a value
of 1 and sent a "Stream Begin 1" to the client. Furthermore, some
client libraries (Red5) seem to trip up on receiving
"@setDataFrame" on stream 0 (and may be correct to assume that
this message would be sent on stream 1).
---
 libavformat/rtmpproto.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 4095ae9421..846376e668 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -3049,7 +3049,12 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
                                              pkttype, ts, pktsize)) < 0)
                 return ret;
 
-            rt->out_pkt.extra = rt->stream_id;
+            // If rt->listen, then we're running as a a server and should
+            // use the ID that we've sent in Stream Begin and in the
+            // _result to createStream.
+            // Otherwise, we're running as a client and should use the ID
+            // that we've received in the createStream from the server.
+            rt->out_pkt.extra = (rt->listen) ? rt->nb_streamid : rt->stream_id;
             rt->flv_data = rt->out_pkt.data;
         }
 
-- 
2.39.5


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[FFmpeg-devel] [PATCH 6/8] GOL-1361: Remove invalid CTTS sample_offset check

[Tomas Härdin]

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I trust that ticket #385 has a FATE test these days

/Tomas

[-- Attachment #2: 0006-GOL-1361-Remove-invalid-CTTS-sample_offset-check.patch --]
[-- Type: text/x-patch, Size: 1176 bytes --]

From 8c105c5953c494402749eb27d2eb6a7a2393f855 Mon Sep 17 00:00:00 2001
From: ekir <ekir@spotify.com>
Date: Tue, 18 Apr 2023 17:31:43 +0200
Subject: [PATCH 6/8] GOL-1361: Remove invalid CTTS sample_offset check

We checked in this places:
* In 8.6.1.3 of ISO/IEC 14496-12 about the CTTS box
* In Apples MOV spec: https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-SW19
---
 libavformat/mov.c | 7 -------
 1 file changed, 7 deletions(-)

diff --git a/libavformat/mov.c b/libavformat/mov.c
index 2c8be51063..89bfb52d2e 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -3697,13 +3697,6 @@ static int mov_read_ctts(MOVContext *c, AVIOContext *pb, MOVAtom atom)
         av_log(c->fc, AV_LOG_TRACE, "count=%d, duration=%d\n",
                 count, duration);
 
-        if (FFNABS(duration) < -(1<<28) && i+2<entries) {
-            av_log(c->fc, AV_LOG_WARNING, "CTTS invalid\n");
-            av_freep(&sc->ctts_data);
-            sc->ctts_count = 0;
-            return 0;
-        }
-
         if (i+2<entries)
             mov_update_dts_shift(sc, duration, c->fc);
     }
-- 
2.39.5


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[FFmpeg-devel] [PATCH 7/8] avformat/mp3dec: Subtract known padding from duration

[Tomas Härdin]

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From 6dca5b958693588d74081e8fa29f05a5f257c841 Mon Sep 17 00:00:00 2001
From: Ulrik Mikaelsson <ulrik.mikaelsson@gmail.com>
Date: Tue, 22 Aug 2023 13:55:14 +0200
Subject: [PATCH 7/8] avformat/mp3dec: Subtract known padding from duration

When an Info-tag is present, marking initial and trailing samples as
padding, those samples should not be included in the calculation of track
duration.

This solves a surprising user experience where converting a WAV->MP3->WAV,
ffprobe will show the duration of the mp3 as slightly longer than both the
input and the output.

As a result, the estimated duration and imprecise seek-results of some
FATE-tests have been updated.
---
 libavformat/mp3dec.c                 | 20 ++++++++++++++------
 tests/ref/fate/gapless-mp3-side-data |  4 ++--
 tests/ref/seek/extra-mp3             |  8 ++++----
 3 files changed, 20 insertions(+), 12 deletions(-)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index dac91205d9..31eeb68ebb 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -52,6 +52,7 @@ typedef struct {
     int usetoc;
     unsigned frames; /* Total number of frames in file */
     unsigned header_filesize;   /* Total number of bytes in the stream */
+    unsigned frame_duration;   /* Frame duration in st->time_base */
     int is_cbr;
 } MP3DecContext;
 
@@ -341,6 +342,7 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
 
     mp3->frames = 0;
     mp3->header_filesize   = 0;
+    mp3->frame_duration = av_rescale_q(spf, (AVRational){1, c.sample_rate}, st->time_base);
 
     mp3_parse_info_tag(s, st, &c, spf);
     mp3_parse_vbri_tag(s, st, base);
@@ -351,11 +353,18 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
     /* Skip the vbr tag frame */
     avio_seek(s->pb, base + vbrtag_size, SEEK_SET);
 
-    if (mp3->frames)
-        st->duration = av_rescale_q(mp3->frames, (AVRational){spf, c.sample_rate},
+    if (mp3->frames) {
+        int64_t full_duration_samples;
+
+        full_duration_samples = mp3->frames * (int64_t)spf;
+        st->duration = av_rescale_q(full_duration_samples - mp3->start_pad - mp3->end_pad,
+                                    (AVRational){1, c.sample_rate},
                                     st->time_base);
-    if (mp3->header_filesize && mp3->frames && !mp3->is_cbr)
-        st->codecpar->bit_rate = av_rescale(mp3->header_filesize, 8 * c.sample_rate, mp3->frames * (int64_t)spf);
+
+        if (mp3->header_filesize && !mp3->is_cbr)
+            st->codecpar->bit_rate = av_rescale(mp3->header_filesize, 8 * c.sample_rate,
+                                                full_duration_samples);
+    }
 
     return 0;
 }
@@ -586,8 +595,7 @@ static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
         return best_pos;
 
     if (mp3->is_cbr && ie == &ie1 && mp3->frames && mp3->header_filesize > 0) {
-        int frame_duration = av_rescale(st->duration, 1, mp3->frames);
-        ie1.timestamp = frame_duration * av_rescale(best_pos - si->data_offset, mp3->frames, mp3->header_filesize);
+        ie1.timestamp = mp3->frame_duration * av_rescale(best_pos - si->data_offset, mp3->frames, mp3->header_filesize);
     }
 
     avpriv_update_cur_dts(s, st, ie->timestamp);
diff --git a/tests/ref/fate/gapless-mp3-side-data b/tests/ref/fate/gapless-mp3-side-data
index 49ebc32dc8..878425f092 100644
--- a/tests/ref/fate/gapless-mp3-side-data
+++ b/tests/ref/fate/gapless-mp3-side-data
@@ -593,5 +593,5 @@ packet|codec_type=audio|stream_index=0|pts=217866240|pts_time=15.438367|dts=2178
 packet|codec_type=audio|stream_index=0|pts=218234880|pts_time=15.464490|dts=218234880|dts_time=15.464490|duration=368640|duration_time=0.026122|size=418|pos=248882|flags=K__|data_hash=CRC32:fbc83c3c
 packet|codec_type=audio|stream_index=0|pts=218603520|pts_time=15.490612|dts=218603520|dts_time=15.490612|duration=368640|duration_time=0.026122|size=418|pos=249300|flags=K__|data_hash=CRC32:d5fb5f9c|side_datum/skip_samples:side_data_type=Skip Samples|side_datum/skip_samples:skip_samples=0|side_datum/skip_samples:discard_padding=303|side_datum/skip_samples:skip_reason=0|side_datum/skip_samples:discard_reason=0
 packet|codec_type=audio|stream_index=0|pts=218972160|pts_time=15.516735|dts=218972160|dts_time=15.516735|duration=368640|duration_time=0.026122|size=418|pos=249718|flags=K__|data_hash=CRC32:3789f3cf|side_datum/skip_samples:side_data_type=Skip Samples|side_datum/skip_samples:skip_samples=0|side_datum/skip_samples:discard_padding=1152|side_datum/skip_samples:skip_reason=0|side_datum/skip_samples:discard_reason=0
-stream|index=0|codec_name=mp3|profile=unknown|codec_type=audio|codec_tag_string=[0][0][0][0]|codec_tag=0x0000|sample_fmt=fltp|sample_rate=44100|channels=2|channel_layout=stereo|bits_per_sample=0|initial_padding=0|id=N/A|r_frame_rate=0/0|avg_frame_rate=0/0|time_base=1/14112000|start_pts=353600|start_time=0.025057|duration_ts=219340800|duration=15.542857|bit_rate=128000|max_bit_rate=N/A|bits_per_raw_sample=N/A|nb_frames=N/A|nb_read_frames=N/A|nb_read_packets=595|disposition:default=0|disposition:dub=0|disposition:original=0|disposition:comment=0|disposition:lyrics=0|disposition:karaoke=0|disposition:forced=0|disposition:hearing_impaired=0|disposition:visual_impaired=0|disposition:clean_effects=0|disposition:attached_pic=0|disposition:timed_thumbnails=0|disposition:non_diegetic=0|disposition:captions=0|disposition:descriptions=0|disposition:metadata=0|disposition:dependent=0|disposition:still_image=0|disposition:multilayer=0|tag:encoder=LAME3.93 
-format|filename=gapless.mp3|nb_streams=1|nb_programs=0|nb_stream_groups=0|format_name=mp3|start_time=0.025057|duration=15.542857|size=250264|bit_rate=128812|probe_score=51|tag:title=test
+stream|index=0|codec_name=mp3|profile=unknown|codec_type=audio|codec_tag_string=[0][0][0][0]|codec_tag=0x0000|sample_fmt=fltp|sample_rate=44100|channels=2|channel_layout=stereo|bits_per_sample=0|initial_padding=0|id=N/A|r_frame_rate=0/0|avg_frame_rate=0/0|time_base=1/14112000|start_pts=353600|start_time=0.025057|duration_ts=218521600|duration=15.484807|bit_rate=128000|max_bit_rate=N/A|bits_per_raw_sample=N/A|nb_frames=N/A|nb_read_frames=N/A|nb_read_packets=595|disposition:default=0|disposition:dub=0|disposition:original=0|disposition:comment=0|disposition:lyrics=0|disposition:karaoke=0|disposition:forced=0|disposition:hearing_impaired=0|disposition:visual_impaired=0|disposition:clean_effects=0|disposition:attached_pic=0|disposition:timed_thumbnails=0|disposition:non_diegetic=0|disposition:captions=0|disposition:descriptions=0|disposition:metadata=0|disposition:dependent=0|disposition:still_image=0|disposition:multilayer=0|tag:encoder=LAME3.93 
+format|filename=gapless.mp3|nb_streams=1|nb_programs=0|nb_stream_groups=0|format_name=mp3|start_time=0.025057|duration=15.484807|size=250264|bit_rate=129295|probe_score=51|tag:title=test
diff --git a/tests/ref/seek/extra-mp3 b/tests/ref/seek/extra-mp3
index fdfb1a7c66..ec5947a9d5 100644
--- a/tests/ref/seek/extra-mp3
+++ b/tests/ref/seek/extra-mp3
@@ -8,11 +8,11 @@ ret: 0         st: 0 flags:1 dts: 0.809796 pts: 0.809796 pos:  14407 size:   418
 ret: 0         st: 0 flags:1  ts:-0.317499
 ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:   1451 size:   417
 ret: 0         st:-1 flags:0  ts: 2.576668
-ret: 0         st: 0 flags:1 dts: 2.586122 pts: 2.586122 pos:  42828 size:   418
+ret: 0         st: 0 flags:1 dts: 2.612245 pts: 2.612245 pos:  43246 size:   418
 ret: 0         st:-1 flags:1  ts: 1.470835
 ret: 0         st: 0 flags:1 dts: 1.462857 pts: 1.462857 pos:  24856 size:   418
 ret: 0         st: 0 flags:0  ts: 0.365002
-ret: 0         st: 0 flags:1 dts: 0.365714 pts: 0.365714 pos:   7302 size:   418
+ret: 0         st: 0 flags:1 dts: 0.391837 pts: 0.391837 pos:   7720 size:   418
 ret: 0         st: 0 flags:1  ts:-0.740831
 ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:   1451 size:   417
 ret: 0         st:-1 flags:0  ts: 2.153336
@@ -22,11 +22,11 @@ ret: 0         st: 0 flags:1 dts: 1.044898 pts: 1.044898 pos:  18169 size:   418
 ret: 0         st: 0 flags:0  ts:-0.058330
 ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:   1451 size:   417
 ret: 0         st: 0 flags:1  ts: 2.835837
-ret: 0         st: 0 flags:1 dts: 2.821224 pts: 2.821224 pos:  46590 size:   418
+ret: 0         st: 0 flags:1 dts: 2.847347 pts: 2.847347 pos:  47008 size:   418
 ret: 0         st:-1 flags:0  ts: 1.730004
 ret: 0         st: 0 flags:1 dts: 1.750204 pts: 1.750204 pos:  29454 size:   418
 ret: 0         st:-1 flags:1  ts: 0.624171
-ret: 0         st: 0 flags:1 dts: 0.600816 pts: 0.600816 pos:  11064 size:   418
+ret: 0         st: 0 flags:1 dts: 0.626939 pts: 0.626939 pos:  11482 size:   418
 ret: 0         st: 0 flags:0  ts:-0.481662
 ret: 0         st: 0 flags:1 dts: 0.000000 pts: 0.000000 pos:   1451 size:   417
 ret: 0         st: 0 flags:1  ts: 2.412505
-- 
2.39.5


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[FFmpeg-devel] [PATCH 8/8] Make mime-type award a bonus probe score

[Tomas Härdin]

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Seems reasonable to me and passes FATE

/Tomas

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From ecc3459990f2871fd907f96fe66362b8fea41bd8 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Peter=20Zeb=C3=BChr?= <peterz@spotify.com>
Date: Tue, 21 Nov 2023 14:16:49 +0100
Subject: [PATCH 8/8] Make mime-type award a bonus probe score

This changes the default behaviour of ffmpeg where content-type headers
on an input gives an absolut probe score (of 75) to instead give a bonus
score (of 30). This gives the probe a better chance to arrive at the
correct format by (hopefully) giving a large enough bonus to push edge
cases in the right direction (MPEG-PS vs MP3, I am looking at you) while
also not adversly punishing clearer cases (raw ADTS marked as
"audio/mpeg" for example).

This patch was regression tested against 20 million recent podcast
submissions (after content-type propagation was added to
original-storage), and 50k Juno vodcasts submissions (dito). No adverse
effects observed (but the bonus may still need tweaking if other edge
cases are detected in production).
---
 libavformat/avformat.h   | 2 +-
 libavformat/format.c     | 8 ++++----
 libavformat/libopenmpt.c | 2 +-
 3 files changed, 6 insertions(+), 6 deletions(-)

diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 6abdb6d480..498c557a3c 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -459,7 +459,7 @@ typedef struct AVProbeData {
 #define AVPROBE_SCORE_STREAM_RETRY (AVPROBE_SCORE_MAX/4-1)
 
 #define AVPROBE_SCORE_EXTENSION  50 ///< score for file extension
-#define AVPROBE_SCORE_MIME       75 ///< score for file mime type
+#define AVPROBE_SCORE_MIME_BONUS 30 ///< score added for matching mime type
 #define AVPROBE_SCORE_MAX       100 ///< maximum score
 
 #define AVPROBE_PADDING_SIZE 32             ///< extra allocated bytes at the end of the probe buffer
diff --git a/libavformat/format.c b/libavformat/format.c
index e65a6fc05e..71018ea6ab 100644
--- a/libavformat/format.c
+++ b/libavformat/format.c
@@ -212,10 +212,10 @@ const AVInputFormat *av_probe_input_format3(const AVProbeData *pd,
                 score = AVPROBE_SCORE_EXTENSION;
         }
         if (av_match_name(lpd.mime_type, fmt1->mime_type)) {
-            if (AVPROBE_SCORE_MIME > score) {
-                av_log(NULL, AV_LOG_DEBUG, "Probing %s score:%d increased to %d due to MIME type\n", fmt1->name, score, AVPROBE_SCORE_MIME);
-                score = AVPROBE_SCORE_MIME;
-            }
+            int old_score = score;
+            score += AVPROBE_SCORE_MIME_BONUS;
+            if (score > AVPROBE_SCORE_MAX) score = AVPROBE_SCORE_MAX;
+            av_log(NULL, AV_LOG_DEBUG, "Probing %s score:%d increased to %d due to MIME type\n", fmt1->name, old_score, score);
         }
         if (score > score_max) {
             score_max = score;
diff --git a/libavformat/libopenmpt.c b/libavformat/libopenmpt.c
index 736af7caf2..dee975c9c2 100644
--- a/libavformat/libopenmpt.c
+++ b/libavformat/libopenmpt.c
@@ -244,7 +244,7 @@ static int read_probe_openmpt(const AVProbeData *p)
              * AVPROBE_SCORE_MAX in order to reduce the impact in the rare
              * cases of false positives.
              */
-            return AVPROBE_SCORE_MIME + 1;
+            return (AVPROBE_SCORE_MAX * 3) / 4 + 1;
         } else if (probe_result == OPENMPT_PROBE_FILE_HEADER_RESULT_WANTMOREDATA) {
             if (probe_openmpt_extension(p) > 0) {
                 return AVPROBE_SCORE_RETRY;
-- 
2.39.5


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Re: [FFmpeg-devel] [PATCH 2/8] libavcodec/wmadec: Return AVERROR_INVALIDDATA on decoding errors

[Marth64]

I know `if ((ret = av_do_something()) <0)` pattern has been frowned
upon recently but besides that it LGTM.
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