Signed-off-by: jack
---
libavformat/hlsenc.c | 5 +
1 file changed, 5 insertions(+)
diff --git a/libavformat/hlsenc.c b/libavformat/hlsenc.c
index a9fa5d8..77712d0 100644
--- a/libavformat/hlsenc.c
+++ b/libavformat/hlsenc.c
@@ -624,6 +624,11 @@ static int hls_start(AVFormatContext *s
Signed-off-by: jack
---
libavformat/tcp.c | 44
1 file changed, 44 insertions(+)
diff --git a/libavformat/tcp.c b/libavformat/tcp.c
index a11ccbb913..598d61067e 100644
--- a/libavformat/tcp.c
+++ b/libavformat/tcp.c
@@ -36,6 +36,8 @@ typedef struct
On Wed, Aug 7, 2019 at 10:22 AM Andriy Gelman
wrote:
> On Sun, 04. Aug 14:36, Andriy Gelman wrote:
> > Changes in v2:
> > 1. Replaced zmq_poll with zmq_msg_recv.
> > 2. Remove user timeout option as zmq_msg_recv(.., .., ZMQ_DONTWAIT) is
> a
> > non-blocking call.
> > 3. Updated docs.
> >
On Wed, Apr 8, 2020 at 3:34 PM pkv wrote:
>
> Le 08/04/2020 à 3:45 am, Limin Wang a écrit :
> > On Wed, Apr 08, 2020 at 02:16:23AM +0200, pkv wrote:
> >> Hi
> >>
> >> Le 08/04/2020 à 12:33 am, Limin Wang a écrit :
> >>> On Tue, Apr 07, 2020 at 08:25:44PM +0200, pkv wrote:
> Hi
>
>
On Sat, Mar 6, 2021 at 12:44 AM James Almer wrote:
> Signed-off-by: James Almer
> ---
> fftools/ffmpeg.c | 318 +++
> fftools/ffmpeg.h | 4 +
> fftools/ffmpeg_opt.c | 5 +-
> 3 files changed, 177 insertions(+), 150 deletions(-)
>
> diff --git
Dear:
In doc/example/resampling_audio.c
/* buffer is going to be directly written to a rawaudio file, no alignment
*/
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize,
dst_nb_channels,
x4->sei_size = nal[i].i_payload;
x4->sei = av_malloc(x4->sei_size);
if (!x4->sei)
On Mon, Jul 26, 2021 at 8:09 PM Jack Waller wrote:
> Dear:
>
> The libavcodec/libx264.c uses the wrong offset to obtain the SEI
>
> diff --gi
Dear:
The libavcodec/libx264.c uses the wrong offset to obtain the SEI
diff --git a/libavcodec/libx264.c b/libavcodec/libx264.c
index f78365a4f7..9afaf19547 100644
--- a/libavcodec/libx264.c
+++ b/libavcodec/libx264.c
@@ -936,7 +936,7 @@ static av_cold int X264_init(AVCodecContext *avctx)
Commit a500b975 removed NULL input handling from this function,
moving the check higher up the call tree in one branch. However,
there is another call to set_string_video_rate() which may pass
NULL, and future users of the function may not be clear that
a NULL check is required. This patch restores
tree to gain support for gcc 10, but there are concerns about the
removal of the null check. av_parse_video_rate() passes its argument
directly to strcmp, which would cause undefined behaviour if the argument
was NULL.
Thanks,
Jack
On Fri, Jul 31, 2020 at 8:31 PM Michael Niedermayer
wrote:
>
situations. If someone does pass NULL, as of a500b975 the response will be
undefined behaviour, where previously it would have been cleanly handled
with an explicit error code. I'd contend that this is a bad response.
Thanks
Jack
On Sun, Aug 2, 2020 at 11:28 PM Michael Niedermayer
wrote:
&g
d you explain why you
would not want to do that?
Thanks
Jack
On Mon, Aug 3, 2020 at 11:27 PM Michael Niedermayer
wrote:
> On Mon, Aug 03, 2020 at 02:07:36PM +0100, Jack Haughton wrote:
> > A NULL check in av_parse_video_rate() would certainly not be a bad idea.
> It
> > woul
; 0)
av_log(obj, AV_LOG_ERROR, "Unable to parse option value \"%s\" as
video rate\n", val);
return ret;
--
2.17.0.windows.1
On Wed, Aug 5, 2020 at 2:12 PM Jack Haughton
wrote:
> Fair enough, I agree an assert would be better - I'll send another patch.
>
>
much
certainly a segmentation fault, the documentation marks it as undefined
behaviour, which is, well, undefined - and therefore not to be relied upon
IMO.
Thanks
Jack
On Tue, Aug 4, 2020 at 2:58 PM Nicolas George wrote:
> Jack Haughton (12020-08-04):
> > Absolutely, he should fix his code.
Commit a500b975 removed NULL input handling from this function,
moving the check higher up the call tree in one branch. However,
there is another call to set_string_video_rate() which may pass
NULL, and future users of the function may not be clear that
a NULL check is required. This patch restores
Hi Andreas,
You're right - apologies. So in fact there was an exchange of one kind of
undefined behaviour for another.
Thanks
Jack
On Thu, Aug 6, 2020 at 1:52 PM Andreas Rheinhardt <
andreas.rheinha...@gmail.com> wrote:
> Jack Haughton:
> > A NULL check in av_pars
e-chains.ogg
It reports "Application provided invalid, non monotonically increasing
dts to muxer in stream 0", which appears to be because the initial
timestamp is not 0:00. This affects all raw muxers, including PCM.
Signed-off-by: Jack Bruienne
---
Changelog | 1 +
MAINT
ed patches fixing the issues you mentioned
promptly.
On 2/25/22 03:15, Paul B Mahol wrote:
On Fri, Feb 25, 2022 at 02:54:35AM -0500, Jack Bruienne wrote:
From the wiki page (https://wiki.vexatos.com/dfpwm):
DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
created by Ben “Gr
ize is constant with
respect to the input size.
Signed-off-by: Jack Bruienne
---
Changelog | 1 +
MAINTAINERS | 1 +
doc/general_contents.texi | 1 +
libavcodec/Makefile | 2 +
libavcodec/allcodecs.c| 2 +
libavcodec/codec_desc.c | 7 +
raw data.
Please see the previous patch for more information on DFPWM.
Signed-off-by: Jack Bruienne
---
Changelog | 2 +-
MAINTAINERS | 1 +
doc/general_contents.texi | 1 +
libavformat/Makefile | 2 +
libavformat/allformats.c | 2 +
libavformat
18:43, Jack Bruienne wrote:
From the wiki page (https://wiki.vexatos.com/dfpwm):
DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
created by Ben “GreaseMonkey” Russell in 2012, originally to be used
as a voice codec for asiekierka's pixmess, a C remake of 64pixels.
It is a 1-bit-
with them to make a few adjustments that should improve the
audio quality. I also made sure that the output matches the Java
codec exactly, so it should have the exact same quality as other codecs.
Signed-off-by: Jack Bruienne
---
Changelog | 1 +
MAINTAINERS
rove the
audio quality. I also made sure that the output matches the Java
codec exactly, so it should have the exact same quality as other codecs.
Signed-off-by: Jack Bruienne
---
Changelog | 1 +
MAINTAINERS | 1 +
doc/general_contents.texi | 1 +
libavcodec/M
raw data.
Please see the previous patch for more information on DFPWM.
Changes since v2/v3:
Removed unused MIME parsing code, and added channels option.
Signed-off-by: Jack Bruienne
---
Changelog | 2 +-
MAINTAINERS | 1 +
doc/general_contents.texi | 1
rked with them to make a few adjustments that should improve the
audio quality. I also made sure that the output matches the Java
codec exactly, so it should have the exact same quality as other codecs.
Signed-off-by: Jack Bruienne
---
Changelog | 1 +
MAINTAINERS
descriptions of formats.
Changes since v2/v3:
Removed unused MIME parsing code, and added channels option.
Signed-off-by: Jack Bruienne
---
Changelog | 2 +-
MAINTAINERS | 1 +
doc/general_contents.texi | 1 +
libavformat/Makefile | 2 +
libavformat
most often. I got
in contact with someone who knows DFPWM much better than I do, and I
worked with them to make a few adjustments that should improve the
audio quality. I also made sure that the output matches the Java
codec exactly, so it should have the exact same quality as other codecs.
On 3/7/22 06:03, Tomas Härdin wrote:
tor 2022-03-03 klockan 10:44 -0500 skrev Jack Bruienne:
From the wiki page (https://wiki.vexatos.com/dfpwm):
DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
created by Ben “GreaseMonkey” Russell in 2012, originally to be
used
as a voice
most often. I got
in contact with someone who knows DFPWM much better than I do, and I
worked with them to make a few adjustments that should improve the
audio quality. I also made sure that the output matches the Java
codec exactly, so it should have the exact same quality as other codecs.
raw data.
Please see the previous patch for more information on DFPWM.
Changes since v4:
Fixed descriptions of formats.
Changes since v2/v3:
Removed unused MIME parsing code, and added channels option.
Signed-off-by: Jack Bruienne
---
Changelog | 2 +-
MAINTAINERS
to being an external parameter as in the raw format.
This format is already supported in my AUKit library, which is the CC
analog to libav (albeit much smaller). Support in other applications is TBD.
Signed-off-by: Jack Bruienne
---
libavformat/riff.c| 3 +++
libavformat/riffenc.c | 4 ++--
2
Please check the code:line 209
if (c == 0 || fabs(buf[index + c]) > max_peak)
max_peak = fabs(buf[index + c]);
'max_peak' is initialized.
On Sat, Aug 20, 2022 at 5:39 PM jagad hariseno
wrote:
> Hi All,
>
> at af_loudnorm.c in line number 188:
> doub
This adds the exact bits per sample for DFPWM to
av_get_exact_bits_per_sample.
Previously, the DTS and PTS were set to 0 because the codec never
reported them, but adding this allows libavformat to automatically
set DTS and PTS from the byte position of the stream.
Signed-off-by: Jack
ling issues and ensure correct video duration.
Signed-off-by: Jack Lau
---
doc/examples/transcoding.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/doc/examples/transcoding.c b/doc/examples/transcoding.c
index 013f89fc7d..847bdb7e1a 100644
--- a/doc/examples/transcod
--- Begin Message ---
From bfd5500a5448ad468d32994816e8a55c0d4a2428 Mon Sep 17 00:00:00 2001
From: Jack Lau
Date: Tue, 4 Feb 2025 21:39:20 +0800
Subject: [PATCH] examples/transcoding: Fix time_base handling
X-Unsent: 1
To: ffmpeg-devel@ffmpeg.org
The `dec_ctx->time_base` was incorrectly defa
fix-time-base-handling.patch
Description: Binary data
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>
> AVCodecContext.time_base is not used for decoding.
Thank you for your reply. I understand that time_base is not used during
decoding, but the transcoding code calls av_packet_rescale_ts twice, once
before decoding and once after encoding, as shown below:
540 if (filter_ctx[stream_
>
> AVCodecContext.time_base is not used for decoding.
Thank you for your reply. I understand that time_base is not used during
decoding, but the transcoding code calls av_packet_rescale_ts twice, once
before decoding and once after encoding, as shown below:
540 if (filter_ctx[stream_
sues and ensure correct video duration.
Signed-off-by: Jack Lau
---
doc/examples/transcoding.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/doc/examples/transcoding.c b/doc/examples/transcoding.c
index 013f89fc7d..847bdb7e1a 100644
--- a/doc/examples/transcoding.c
+++
The `dec_ctx->time_base` was incorrectly default set to 0/60, while
`enc_ctx->time_base` was derived from `dec_ctx->framerate`. This mismatch could
cause incorrect video duration in the output.
This patch aligns `enc_ctx->time_base` with `dec_ctx->time_base` to prevent
rescaling issues and ensu
tb: 60
output pkt in_tb: 30 out_tb: 15360
so i get one 20s duration and 15fps video(audio duration is normal because
the input's audio sample ratio is 44100)
So i think the problem is that the enc_ctx->time_base shouldn't set to
av_inv_q(dec_ctx->frame
.
Could any one give me some advice please so that i can fix it?
On Wed, Feb 5, 2025 at 11:25 AM Jack Lau wrote:
> To be clear, i want to give an example, i use a 10s duration, 30fps video.
> The ifmt_ctx->streams[stream_index]->time_base is same as
> ofmt_ctx->streams[stream
rtant patches.
Best wishes
Jack
> On Feb 10, 2025, at 08:54, Soft Works
> wrote:
>
>
>
>> -Original Message-
>> From: ffmpeg-devel > <mailto:ffmpeg-devel-boun...@ffmpeg.org>> On Behalf Of
>> Jack Lau via ffmpeg-devel
>> Sent: Monday,
English and grammar correction may
> not come instinctively.
>
> Thank you for the contribution Jack. Unless there is any objection on
> grounds of maintaining commit history, I'll adjust the grammar of
> "need to setting" -> "we need to set" and push t
> On Feb 10, 2025, at 09:36, Soft Works
> wrote:
>
>
>
>> -Original Message-
>> From: ffmpeg-devel > <mailto:ffmpeg-devel-boun...@ffmpeg.org>> On Behalf Of
>> Jack Lau
>> Sent: Monday, February 10, 2025 2:13 AM
>> To: FFmp
> I have been absent from the list for a few years, so I would appreciate it
> if someone could catch me up a bit. I am needing to extract and embed scc
> files with 608 captions. I am pleased to see that transcoding without
> frame rate changes now preserves 608 intact, and there appear to be
>
> Hi Jack,
>
> "paying attention next time"? That's not the right answer.
>
> Please make sure that there won't be a next time.
>
> The big evil with LLVMs is not the fact they are making mistakes but the
> extreme level of confidence at whi
>
>
> Hi Zack,
>
> that message from "Jack" had confused me for a moment, but on re-reading it
> appears to be an AI response.
> The content is total nonsense. There is no "SCC" encoder in ffmpeg, and if
> there was one, it wouldn't help much b
ago, I submitted a patch titled "[PATCH] examples/transcoding: Fix
time_base handling," but it seems that no one has reviewed it yet. I understand
everyone is busy, but I was wondering if there has been any feedback or if
there are any concerns regarding the patch.
Jack
&g
like general_avoption which has a longer life cycle
than ordinary avoption, so that ordinary avoption can still retain the feature
of being detected whether it is used.
Anyone feel free to give me some advice, I really want to make some
contribution for this issue.
Thanks
Jack
> On Mar 2, 2025, at 15:47, Jack Lau via ffmpeg-devel
> wrote:
>
> fix ticket: 10786
> parse the SPS from extradata and get profile_compatibility, tier, constraints
> which was been hard code before.
>
> HEVC CODECS Attribute reference to: ISO/IEC14496-15
&
> On Mar 3, 2025, at 09:08, Steven Liu wrote:
>
> Jack Lau via ffmpeg-devel <mailto:ffmpeg-devel@ffmpeg.org>> 于2025年3月2日周日 21:31写道：
>>
>> fix ticket: 10786
>> parse the SPS from extradata and get profile_compatibility, tier,
>> constraints which was
> On Mar 24, 2025, at 08:11, Steven Liu wrote:
>
> Jack Lau via ffmpeg-devel <mailto:ffmpeg-devel@ffmpeg.org>> 于2025年3月24日周一 07:13写道：
>>
>> The previous code sets the bitrate to be calculated only when duration>0.5,
>> which is obviously not general e
> On Mar 12, 2025, at 15:06, Jack Lau via ffmpeg-devel
> wrote:
>
> fix ticket: 11316
> add set_hevc_codec_str function refer to hlsenc.c but do some necessary
> changes
> Signed-off-by: Jack Lau
> ---
> libavformat/dashenc.c | 81 +
culation
to account for `ticks_per_frame`, ensuring that the time base is consistent
between the decoder and encoder contexts.
--- Begin Message ---
From 6a02fbaf6c6068040640ff105ad70115fb81b5d2 Mon Sep 17 00:00:00 2001
From: Jack Lau
Date: Tue, 4 Feb 2025 21:39:20 +0800
Subject: [PATCH] examples/tran
From: Jack Lau
The `dec_ctx->time_base` was incorrectly default set by avcodec_open2(), while
`enc_ctx->time_base` was derived from `dec_ctx->framerate`. This mismatch could
cause incorrect video duration in the output.
This patch corrects the issue by adjusting the `enc_ctx-&
---
libavformat/hls.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/hls.c b/libavformat/hls.c
index 3bdc1bc848..c2130bb883 100644
--- a/libavformat/hls.c
+++ b/libavformat/hls.c
@@ -1993,7 +1993,7 @@ static int hls_read_header(AVFormatContext *s)
return r
Increase the default buffer size to match more modern encoding scenarios.
---
libavformat/mpegenc.c | 5 ++---
1 file changed, 2 insertions(+), 3 deletions(-)
diff --git a/libavformat/mpegenc.c b/libavformat/mpegenc.c
index 6b6763c30f..2b3b98b894 100644
--- a/libavformat/mpegenc.c
+++ b/libavform
mpty, and ff_hls_write_stream_info cannot write stream info normally,
causing master_pl to be unavailable.
Signed-off-by: Jack Lau
---
libavformat/hlsenc.c | 4 +---
1 file changed, 1 insertion(+), 3 deletions(-)
diff --git a/libavformat/hlsenc.c b/libavformat/hlsenc.c
index c6ffdb99e5..223c516103 100
Should fix ticket 11543
if input url is relative path, the seg-url would make absolute url twice in
get_content_url and open_input function
but it doesn't need make absolute url in open_input since we set it already
Signed-off-by: Jack Lau
---
libavformat/dashdec.c | 2 +-
1 file chang
patch add it
Signed-off-by: Jack Lau
---
libavformat/rtpdec_mpeg4.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/libavformat/rtpdec_mpeg4.c b/libavformat/rtpdec_mpeg4.c
index 6531632b2d..c506bcbed1 100644
--- a/libavformat/rtpdec_mpeg4.c
+++ b/libavformat/rtpdec_mpeg4.c
@@ -363,6 +363,7
fix ticket: 10786
parse the SPS from extradata and get profile_compatibility, tier, constraints
which was been hard code before.
HEVC CODECS Attribute reference to: ISO/IEC14496-15
Signed-off-by: Jack Lau
---
libavformat/hlsenc.c | 41 ++---
1 file changed
fix ticket: 10786
parse the SPS from extradata and get profile_compatibility, tier, constraints
which was been hard code before.
HEVC CODECS Attribute reference to: ISO/IEC14496-15
Signed-off-by: Jack Lau
---
libavformat/hlsenc.c | 38 +++---
1 file changed, 35
fix ticket: 10786
parse the SPS from extradata and get profile_compatibility, tier, constraints
which was been hard code before.
HEVC CODECS Attribute reference to: ISO/IEC14496-15
Signed-off-by: Jack Lau
---
libavformat/hlsenc.c | 37 ++---
1 file changed, 34
fix ticket: 10786
parse the SPS from extradata and get profile_compatibility, tier, constraints
which was been hard code before.
HEVC CODECS Attribute reference to: ISO/IEC14496-15
Signed-off-by: Jack Lau
---
libavformat/hlsenc.c | 38 +++---
1 file changed, 35
fix ticket: 11316
add set_hevc_codec_str function refer to hlsenc.c but do some necessary changes
Signed-off-by: Jack Lau
---
libavformat/dashenc.c | 81 +++
1 file changed, 81 insertions(+)
diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
index
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