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Git pushed a commit to branch release/8.1
in repository ffmpeg.
The following commit(s) were added to refs/heads/release/8.1 by this push:
new 1005b294ff avformat/iamf: backport assorted fixes
1005b294ff is described below
commit 1005b294ffdf1b4e75f58d7e98362f82462a6a61
Author: James Almer <[email protected]>
AuthorDate: Thu Sep 10 16:46:30 2026 -0300
Commit: James Almer <[email protected]>
CommitDate: Tue Sep 15 00:18:58 2026 -0300
avformat/iamf: backport assorted fixes
Too many merge conflics to attempt individual cherry-picks.
Signed-off-by: James Almer <[email protected]>
---
libavformat/iamf_parse.c | 5 +++++
libavformat/iamf_writer.c | 17 ++++++++++++-----
2 files changed, 17 insertions(+), 5 deletions(-)
diff --git a/libavformat/iamf_parse.c b/libavformat/iamf_parse.c
index 12d348a705..892145cd87 100644
--- a/libavformat/iamf_parse.c
+++ b/libavformat/iamf_parse.c
@@ -1011,6 +1011,11 @@ static int mix_presentation_obu(void *s, IAMFContext *c,
AVIOContext *pb, int le
mix_presentation->cmix = mix;
mix_presentation->count_label = ffio_read_leb(pbc);
+ if (mix_presentation->count_label > len - avio_tell(pbc)) {
+ mix_presentation->count_label = 0;
+ ret = AVERROR_INVALIDDATA;
+ goto fail;
+ }
mix_presentation->language_label = av_calloc(mix_presentation->count_label,
sizeof(*mix_presentation->language_label));
if (!mix_presentation->language_label) {
diff --git a/libavformat/iamf_writer.c b/libavformat/iamf_writer.c
index ef7cecab27..0c43b64f83 100644
--- a/libavformat/iamf_writer.c
+++ b/libavformat/iamf_writer.c
@@ -100,12 +100,18 @@ static int populate_audio_roll_distance(IAMFCodecConfig
*codec_config)
}
static int fill_codec_config(IAMFContext *iamf, const AVStreamGroup *stg,
- IAMFCodecConfig *codec_config)
+ IAMFCodecConfig *codec_config, void *log_ctx)
{
const AVStream *st = stg->streams[0];
IAMFCodecConfig **tmp;
int j, ret = 0;
+ if (!st->codecpar->frame_size) {
+ av_log(log_ctx, AV_LOG_ERROR, "frame_size is unset for stream id %d\n",
+ st->id);
+ return AVERROR(EINVAL);
+ }
+
codec_config->codec_id = st->codecpar->codec_id;
codec_config->codec_tag = st->codecpar->codec_tag;
switch (codec_config->codec_id) {
@@ -308,7 +314,7 @@ int ff_iamf_add_audio_element(IAMFContext *iamf, const
AVStreamGroup *stg, void
if (!codec_config)
return AVERROR(ENOMEM);
- ret = fill_codec_config(iamf, stg, codec_config);
+ ret = fill_codec_config(iamf, stg, codec_config, log_ctx);
if (ret < 0) {
av_free(codec_config);
return ret;
@@ -1239,9 +1245,10 @@ int ff_iamf_write_audio_frame(const IAMFContext *iamf,
AVIOContext *pb,
if (codec_config->codec_id == AV_CODEC_ID_OPUS) {
// IAMF's num_samples_to_trim_at_start is the same as Opus's pre-skip.
- skip_samples = pkt->dts < 0
- ? av_rescale(-pkt->dts, 48000, pkt->time_base.den)
- : 0;
+ if (!skip_samples)
+ skip_samples = pkt->dts < 0
+ ? av_rescale(-pkt->dts, 48000, pkt->time_base.den)
+ : 0;
discard_padding = av_rescale(discard_padding, 48000,
pkt->time_base.den);
}
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