Re: [FFmpeg-devel] [PATCH] fix av_log format specifier
This patch does not meet the guidelines of commit messages.
On Thursday, April 25th, 2024 at 6:02 PM, Marcus B Spencer
wrote:
>
>
> Signed-off-by: Marcus B Spencer mar...@marcusspencer.xyz
>
> ---
> libavcodec/bsf/noise.c | 2 +-
> 1 file changed, 1 insertion(+), 1 deletion(-)
>
> diff --git a/libavcodec/bsf/noise.c b/libavcodec/bsf/noise.c
> index a622855717..d36be5fab4 100644
> --- a/libavcodec/bsf/noise.c
> +++ b/libavcodec/bsf/noise.c
> @@ -173,7 +173,7 @@ static int noise(AVBSFContext *ctx, AVPacket *pkt)
> drop = !(s->state % s->dropamount);
>
> }
>
> - av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %d pts %"PRId64" - amount %d
> drop %d\n",
> + av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %u pts %"PRId64" - amount %d
> drop %d\n",
> pkt->stream_index, (unsigned int)s->var_values[VAR_N], pkt->pts, amount,
> drop);
>
>
> if (drop) {
> --
> 2.44.0
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Re: [FFmpeg-devel] archive ffmpeg all objs into one lib, feasible?
No, it won't work because the libraries are for different purposes. as per your example, 4xm in the libavcodec directory is for decoding the file, it implements the actual algorithm. 4xm in the libavformat directory is for demuxing the codec from the 4xm container, you need both for it to work. On Wed, Sep 17, 2014 at 10:37 AM, Lynn Yu yuqilin1...@gmail.com wrote: I've build ffmpeg static libraries, and all necessary objs (*.o) generated in their source directory. I use 'ar' to archive all these objs into one libffmpeg.a. command: *ar rc libffmpeg.a *.o* I notice that, there are some objs have same name in different dirs, for example *libavformat/4xm.o and libavcodec/4xm.o*. by 'nm libffmpeg.a', it seems both objs are there? 'man ar' I find that, if obj with same name and Exactly with same function symbols, latter one will be replaced ,right? by this 'ar' way, will libffmpeg.a work correctly? because there so many symbols I can't test every one. thanks for help! ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Bug in flac demuxer?
I don't have access to my command line atm, but i was using the channel map feature to split a 24 bit 192khz 5.1 flac file into 6 mono wave files, and ffmpeg would downcovert it to 16 bit unless i added -acodec pcm_s24le after each -map flag, shouldn't ffmpeg read the bit depth of the input file and use that instead of assuming the file is 16 bit like it currently does? Originally i was splitting it to mono flacs but i couldn't set the bit depth without using wave so i had to change it. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] OPW Qualification Task: Validate MLP Bitstream
On Fri, Oct 31, 2014 at 10:09 AM, greeshma greeshmabalaba...@gmail.com wrote: https://docs.google.com/document/d/1oBy9AoGzJHR4UcvuogYa8sfFGUK96lcgF3qRF4HrIa4/edit?usp=sharing That's not going to work either, you need to use git to create a patch file, upload that file to dropbox or whatever, and paste a link to it, OR you can click on the paper clip icon in gmail and select the patch file for uploading, and that will attach it properly. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] Move ffmpeg to WinRT
As of Windows 10, WinRT is deprecated, so it's kind of a waste of time dude :/ On Mon, Nov 17, 2014 at 5:02 AM, Jesse Jiang jessejiang0...@outlook.com wrote: Hi All, I want to move ffmpeg to WinRT platform, like Windows Store and Windows Phone. As the GCC cannot compiler to ARM-COFF, so I convert the GNU-style assembly codes to ARM-style codes. Also the codes are open-sourced, here https://github.com/qyljcy/FFmpeg Now this project can be compiled, but I didn't know if the assembly codes work well. I want to know, if there is any test project to test the function like ff_ps_add_squares_neon, ect. As the WinRT platform is different from win32 or linux, so I need to test them one by one. I hope someone can help me, or work together. Thanks very much Best regards,Jesse ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] Errors We are facing
The SSR (Scalable Sample Rate) feature is not implemented, either add a patch to add that feature, or decode it with something else. On Tue, Nov 25, 2014 at 6:58 AM, Arpan Nag ar...@esolzmail.com wrote: Hello, During the conversion process we are seeing this error in an output log file --- which has not been implemented. [aac @ 0x41c1820] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org) --- [aac @ 0x41c1820] SSR is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0x41c1820] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org) Error while decoding stream #0:1: Not yet implemented in FFmpeg, patches welcome frame= 2138 fps=5.3 q=0.0 size=5240kB time=00:01:29.17 bitrate= 481.4kbits/s [aac @ 0x41c1820] SSR is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0x41c1820] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org) Error while decoding stream #0:1: Not yet implemented in FFmpeg, patches welcome --- Can you please help us out about it. Thank You Arpan Nag +919038706300 ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Git branch
I'm working on the DTS codec, and my code isn't complete yet, but a lot of it is done, I've committed a few patches to my own branch, and I need to update ffmpeg (there's a blocking change in the main tree so I can't currently) I was wondering if I could push the uncomplete code in it's own branch, or what I could do to remedy this situation? I'm a newb at git, and OSS development in general so I'm not really sure how all of this works. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] Git branch
it's not published anywhere, I just used the github app and clicked the new branch button, it's only on my computer right now On Sat, Dec 20, 2014 at 5:33 AM, Carl Eugen Hoyos ceho...@ag.or.at wrote: Marcus Johnson bumblebritches57 at gmail.com writes: I'm working on the DTS codec, and my code isn't complete yet, but a lot of it is done, I've committed a few patches to my own branch Where can we find this branch? Perhaps other developers will help you merging. Carl Eugen ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Duration/bitrate estimation
What functions are used to estimate the duration and bit rate of a file? I'd like to add support for these features for the DTS format, and would rather not dig around the source looking for it. thanks in advance. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] Read backwards with get_bits
I just tried using skip_bits with a negative number and it seems to have worked fine, thanks guys. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Read backwards with get_bits
Can this be done at all? ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH]Do not set bitrate for DTS-HD Master and High Resolution
This reminds me of another bug with DTS files, it estimates the file duration by counting each frame I assume, including the HD ones resulting in it being massively incorrect for example here's the ffmpeg output of a DTS-HD MA file that's actually 98 minutes long Log: ffmpeg -i /Users/Marcus/Desktop/DTS/ThePrincessBride.dtsma ffmpeg version N-68833-ge949e9f Copyright (c) 2000-2014 the FFmpeg developers built on Mar 18 2015 07:47:46 with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn) configuration: --disable-yasm --disable-asm --disable-inline-asm --disable-ffserver --disable-ffplay --disable-doc --disable-ffprobe libavutil 54. 16.100 / 54. 16.100 libavcodec 56. 19.100 / 56. 19.100 libavformat56. 16.102 / 56. 16.102 libavdevice56. 3.100 / 56. 3.100 libavfilter 5. 6.100 / 5. 6.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 [dts @ 0x7fe833822800] Estimating duration from bitrate, this may be inaccurate Input #0, dts, from '/Users/Marcus/Desktop/DTS/ThePrincessBride.dtsma': Duration: 04:06:57.61, start: 0.00, bitrate: 1535 kb/s Stream #0:0: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s Instead of it saying it's about 1 hours and 30 minutes, it says it's 4 hours, and 6 minutes. Maybe the parser should ignore the dts-hd frames, because they won't increase the duration at all, due to the differential nature of the codec. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH]Do not set bitrate for DTS-HD Master and High Resolution
I see, I thought it counted frames and not just multiplied the bitrate by the number of frames. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH]Do not set bitrate for DTS-HD Master and High Resolution
That's even worse, is there any way we can fix it? how much effort would it require? ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] DCA Decoder
I thought the patch on LibAV was completely removed? it was purged from the codebase like 9 months ago or something, I stumbled on that while trying to fix some of the issues with the white paper I was having. I haven't bothered with the Core decoder, but everything I've extracted so far is fixed point. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] DCA Decoder
I've been working on adding XLL for the last couple months, it's still not quite complete, basically I have to combine the Core and XLL samples before it's output, and I also have to finish the latter stages of decoding the XLL like channel decorreclation, and post processing. There are a few issues thought. 1: My code doesn't have the most recent chances from master, how do I fix this? 2: I read that you'd prefer for code to be small and self contained, but how can I do that when the later functions directly require earlier ones? 3: I've been using the variable names the white paper uses which doesn't match your guidelines, obviously they need to be renamed but I'm not sure what else you guys want me to do with that? 4: currently almost all of the variables are being stored in DCAContext and accessed in different functions this way, I assume this is looked down upon and you'll want me to change this style to use the variables in the disparate functions when it's called rather than passing it around the backend, is this correct? That said, I've rewritten the ExSS code, ExSS Asset Descriptor, XLL, and XLL ChSet functions are completely ready to go. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] FFmpeg 2.7
I like the John Nash idea. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] Off topic question about format specifiers in the C stdlib
No, I'm writing my own codec from the ground up; not an implementation of one that already exists based on a standardized codec like jpeg, but my own from scratch. I noticed that printf and scanf support format specifiers, so I was wondering if there was a function like getopt, but specifically for input and output, but I guess not. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Off topic question about format specifiers in the C stdlib
I'm writing my own Video codec, and I'm trying to take multiple frames as the input with a specifier like %03d, is there a stdlib function for this, or will I have to write it myself? ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] TrueHD Decoder fails on sample
The sample is exactly 1 minute long, 48000hz sample rate, 24 bits per sample, 8 channels. and all samples are set to 0xFF here's the thd sample, and here's one of the 8 mono wavs (they're all literally the same, I manually created the file in a hex editor and copied it for all the others.) the thd was created with Dolby Media Encoder Standalone Edition. Input.mlp is what ffmpeg decodes, to Output.wav. Original.wav is the file fed into Dolby Media Encode Standalone Edition Here are the files: Input.mlp: https://www.dropbox.com/s/fvh9lie1ofryws0/Input.mlp Output.wav: https://www.dropbox.com/s/vjfjgv1vu11uci4/Output.wav Original.wav: https://www.dropbox.com/s/8s5n5lv4qtvzysw/Original.wav ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Lossless SilenceRemove Filter
I'm not sure where to post this message, but this list seemed the most appropriate. It seems like a better way to go about removing non-digital silence would be to convert each sample to float or double, then compare that to the max dB considered silence, then store the starting offset and ending offsets based on those values from the original audio, and export just what's in between those. tl;dr output the original samples from an offset, instead of outputting the float samples used in the comparison. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] Job Request | Audio Codecs
Hello, I am interested in hiring someone to expand FFmpeg with a flag that changes the behavior of channel ordering for as many audio codecs as possible. When this flag is seen in the command line it has FFmpeg use slightly modified channel order and channel assignment (TYPESCE instead of TYPELFE) to allow more use cases that are outside of Surround Formats which can be very destructive to FFmpeg uses for newer audio formats such as spatial audio and audio for interactive installations or games. After this it would be fantastic to expand common audio codecs (pcm, aac, vorbis/ogg) to support +8 channels. Apologies if this is not the best way to contact everyone, this is my first email to the ffmpeg-devel chain. Best, Dylan -- 340.29 m / s MACH 1 A Sound Technology Company. New York New York The United States of America. LEGAL NOTICE The contents of this message may be privileged and confidential. Therefore, if this message has been received in error, please delete it without reading it. All contents of the message, including any attachments, are the copyright property of Q Department, L.L.C. This message cannot in any way bind Q Department, L.L.C. to any contract or other obligation. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] [PATCH] avutil/avcodec: expanded on channel configurations for spatial audio
---
Changelog | 1 +
libavcodec/aacenc.h| 16
libavutil/channel_layout.c | 8 ++
libavutil/channel_layout.h | 64 ++
4 files changed, 61 insertions(+), 28 deletions(-)
diff --git a/Changelog b/Changelog
index 2ccd2645fc..08918e95da 100644
--- a/Changelog
+++ b/Changelog
@@ -30,6 +30,7 @@ version :
- MPEG-H 3D Audio support in mp4
- thistogram filter
- freezeframes filter
+- Support for Mach1 Spatial audio channel configurations
version 4.2:
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index 5a015ca92e..a0f99680ee 100644
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -368,6 +368,22 @@ static const AACPCEInfo aac_pce_configs[] = {
.config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE,
TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15
},
},
+{
+.layout = AV_CH_LAYOUT_MACH1SPATIAL_8,
+.num_ele = { 4, 0, 4, 0 },
+.pairing = { { 0 }, { 0 }, { 0 } },
+.index = { { 0, 1, 4, 5 }, { 0 }, { 2, 3, 6, 7 } },
+.config_map = { 8, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE,
TYPE_SCE, TYPE_SCE, TYPE_SCE },
+.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+},
+{
+.layout = AV_CH_LAYOUT_MACH1SPATIAL_8_ST,
+.num_ele = { 5, 0, 4, 0 },
+.pairing = { { 0, 1 }, { 0 }, { 0 } },
+.index = { {0, 1, 4, 5, 0 }, { 0 }, { 2, 3, 6, 7 } },
+.config_map = { 9, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE,
TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
+.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 },
+},
};
/**
diff --git a/libavutil/channel_layout.c b/libavutil/channel_layout.c
index 3bd5ee29b7..37119fd023 100644
--- a/libavutil/channel_layout.c
+++ b/libavutil/channel_layout.c
@@ -62,6 +62,12 @@ static const struct channel_name channel_names[] = {
[33] = { "SDL", "surround direct left" },
[34] = { "SDR", "surround direct right" },
[35] = { "LFE2", "low frequency 2" },
+[36] = { "BFL", "bottom front left" },
+[37] = { "BFR", "bottom front right"},
+[38] = { "BBL", "bottom back left" },
+[39] = { "BBR", "bottom back right" },
+[40] = { "STL", "spatial stereo left" },
+[41] = { "STR", "spatial stereo right" },
};
static const char *get_channel_name(int channel_id)
@@ -104,6 +110,8 @@ static const struct {
{ "octagonal", 8, AV_CH_LAYOUT_OCTAGONAL },
{ "hexadecagonal", 16, AV_CH_LAYOUT_HEXADECAGONAL },
{ "downmix", 2, AV_CH_LAYOUT_STEREO_DOWNMIX, },
+{ "mach1spatial-8", 8, AV_CH_LAYOUT_MACH1SPATIAL_8 },
+{ "mach1spatial-8.2", 10, AV_CH_LAYOUT_MACH1SPATIAL_8_ST},
};
static uint64_t get_channel_layout_single(const char *name, int name_len)
diff --git a/libavutil/channel_layout.h b/libavutil/channel_layout.h
index 50bb8f03c5..5bb2415155 100644
--- a/libavutil/channel_layout.h
+++ b/libavutil/channel_layout.h
@@ -64,6 +64,10 @@
#define AV_CH_TOP_BACK_LEFT 0x8000
#define AV_CH_TOP_BACK_CENTER0x0001
#define AV_CH_TOP_BACK_RIGHT 0x0002
+#define AV_CH_BOTTOM_FRONT_LEFT 0x0010ULL
+#define AV_CH_BOTTOM_FRONT_RIGHT 0x0020ULL
+#define AV_CH_BOTTOM_BACK_LEFT 0x0040ULL
+#define AV_CH_BOTTOM_BACK_RIGHT 0x0080ULL
#define AV_CH_STEREO_LEFT0x2000 ///< Stereo downmix.
#define AV_CH_STEREO_RIGHT 0x4000 ///< See AV_CH_STEREO_LEFT.
#define AV_CH_WIDE_LEFT 0x8000ULL
@@ -71,6 +75,8 @@
#define AV_CH_SURROUND_DIRECT_LEFT 0x0002ULL
#define AV_CH_SURROUND_DIRECT_RIGHT 0x0004ULL
#define AV_CH_LOW_FREQUENCY_20x0008ULL
+#define AV_CH_SPATIAL_STEREO_LEFT0x0100ULL ///< Non diegetic
stereo for spatial channel formats
+#define AV_CH_SPATIAL_STEREO_RIGHT 0x0200ULL ///< See
AV_CH_SPATIAL_STEREO_LEFT
/** Channel mask value used for AVCodecContext.request_channel_layout
to indicate that the user requests the channel order of the decoder output
@@ -82,34 +88,36 @@
* @defgroup channel_mask_c Audio channel layouts
* @{
* */
-#define AV_CH_LAYOUT_MONO (AV_CH_FRONT_CENTER)
-#define AV_CH_LAYOUT_STEREO(AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT)
-#define AV_CH_LAYOUT_2POINT1
(AV_CH_LAYOUT_STEREO|AV_CH_LOW_FREQUENCY)
-#define AV_CH_LAYOUT_2_1 (AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER)
-#define AV_CH_LAYOUT_SURROUND (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER)
-#define AV_CH_LAYOUT_3POINT1
(AV_CH_LAYOUT_SURROUND|AV_CH_LOW_FREQUENCY)
-#define AV_CH_LAYOUT_4POINT0
(AV_CH_LAYOUT_SURROUND|AV_CH_BACK_CENTER)
-#define AV_CH_LAYOUT_4POINT1
Re: [FFmpeg-devel] [PATCH] avutil/avcodec: expanded on channel configurations for spatial audio
> On Jan 16, 2020, at 5:06 AM, Paul B Mahol wrote:
>
> Wait for custom channel layouts.
is there any information on this? I would love to review this as this is a
constant requirement for a growing number of projects utilizing ffmpeg and
would love to make sure this handles all use cases.
>
> On 1/16/20, Dylan Marcus wrote:
>> ---
>> Changelog | 1 +
>> libavcodec/aacenc.h| 16
>> libavutil/channel_layout.c | 8 ++
>> libavutil/channel_layout.h | 64
>> ++
>> 4 files changed, 61 insertions(+), 28 deletions(-)
>>
>> diff --git a/Changelog b/Changelog
>> index 2ccd2645fc..08918e95da 100644
>> --- a/Changelog
>> +++ b/Changelog
>> @@ -30,6 +30,7 @@ version :
>> - MPEG-H 3D Audio support in mp4
>> - thistogram filter
>> - freezeframes filter
>> +- Support for Mach1 Spatial audio channel configurations
>>
>>
>> version 4.2:
>> diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
>> index 5a015ca92e..a0f99680ee 100644
>> --- a/libavcodec/aacenc.h
>> +++ b/libavcodec/aacenc.h
>> @@ -368,6 +368,22 @@ static const AACPCEInfo aac_pce_configs[] = {
>> .config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE,
>> TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
>> .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
>> 15 },
>> },
>> +{
>> +.layout = AV_CH_LAYOUT_MACH1SPATIAL_8,
>> +.num_ele = { 4, 0, 4, 0 },
>> +.pairing = { { 0 }, { 0 }, { 0 } },
>> +.index = { { 0, 1, 4, 5 }, { 0 }, { 2, 3, 6, 7 } },
>> +.config_map = { 8, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE,
>> TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE },
>> +.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
>> +},
>> +{
>> +.layout = AV_CH_LAYOUT_MACH1SPATIAL_8_ST,
>> +.num_ele = { 5, 0, 4, 0 },
>> +.pairing = { { 0, 1 }, { 0 }, { 0 } },
>> +.index = { {0, 1, 4, 5, 0 }, { 0 }, { 2, 3, 6, 7 } },
>> +.config_map = { 9, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE,
>> TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
>> +.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 },
>> +},
>> };
>>
>> /**
>> diff --git a/libavutil/channel_layout.c b/libavutil/channel_layout.c
>> index 3bd5ee29b7..37119fd023 100644
>> --- a/libavutil/channel_layout.c
>> +++ b/libavutil/channel_layout.c
>> @@ -62,6 +62,12 @@ static const struct channel_name channel_names[] = {
>> [33] = { "SDL", "surround direct left" },
>> [34] = { "SDR", "surround direct right" },
>> [35] = { "LFE2", "low frequency 2" },
>> +[36] = { "BFL", "bottom front left" },
>> +[37] = { "BFR", "bottom front right"},
>> +[38] = { "BBL", "bottom back left" },
>> +[39] = { "BBR", "bottom back right" },
>> +[40] = { "STL", "spatial stereo left" },
>> +[41] = { "STR", "spatial stereo right" },
>> };
>>
>> static const char *get_channel_name(int channel_id)
>> @@ -104,6 +110,8 @@ static const struct {
>> { "octagonal", 8, AV_CH_LAYOUT_OCTAGONAL },
>> { "hexadecagonal", 16, AV_CH_LAYOUT_HEXADECAGONAL },
>> { "downmix", 2, AV_CH_LAYOUT_STEREO_DOWNMIX, },
>> +{ "mach1spatial-8", 8, AV_CH_LAYOUT_MACH1SPATIAL_8 },
>> +{ "mach1spatial-8.2", 10, AV_CH_LAYOUT_MACH1SPATIAL_8_ST},
>> };
>>
>> static uint64_t get_channel_layout_single(const char *name, int name_len)
>> diff --git a/libavutil/channel_layout.h b/libavutil/channel_layout.h
>> index 50bb8f03c5..5bb2415155 100644
>> --- a/libavutil/channel_layout.h
>> +++ b/libavutil/channel_layout.h
>> @@ -64,6 +64,10 @@
>> #define AV_CH_TOP_BACK_LEFT 0x8000
>> #define AV_CH_TOP_BACK_CENTER0x0001
>> #define AV_CH_TOP_BACK_RIGHT 0x0002
>> +#define AV_CH_BOTTOM_FRONT_LEFT 0x0010ULL
>> +#define AV_CH_BOTTOM_FRONT_RIGHT 0x0020ULL
>> +#define AV_CH_BOTTOM_BACK_LEFT 0x0040ULL
>> +#define AV_CH_BOTTOM_BACK_RIGHT 0x0080ULL
>> #define AV_CH_STEREO_LEFT0x2000 ///< Stereo downmix.
>> #define AV_CH_
[FFmpeg-devel] [PATCH] fix av_log format specifier
Signed-off-by: Marcus B Spencer
---
libavcodec/bsf/noise.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavcodec/bsf/noise.c b/libavcodec/bsf/noise.c
index a622855717..d36be5fab4 100644
--- a/libavcodec/bsf/noise.c
+++ b/libavcodec/bsf/noise.c
@@ -173,7 +173,7 @@ static int noise(AVBSFContext *ctx, AVPacket *pkt)
drop = !(s->state % s->dropamount);
}
-av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %d pts %"PRId64" - amount
%d drop %d\n",
+av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %u pts %"PRId64" - amount
%d drop %d\n",
pkt->stream_index, (unsigned int)s->var_values[VAR_N], pkt->pts,
amount, drop);
if (drop) {
--
2.44.0
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[FFmpeg-devel] [PATCH] avcodec/bsf/noise: fix av_log format specifier
Replace the "%d" specifier corresponding to the 2nd argument of av_log starting
on line 176 with "%u", due to the fact that the 2nd argument is an unsigned
int. Without this patch, if the second argument exceeded the maximum value of
an int, the behavior would be undefined.
Signed-off-by: Marcus B Spencer
---
libavcodec/bsf/noise.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavcodec/bsf/noise.c b/libavcodec/bsf/noise.c
index a622855717..d36be5fab4 100644
--- a/libavcodec/bsf/noise.c
+++ b/libavcodec/bsf/noise.c
@@ -173,7 +173,7 @@ static int noise(AVBSFContext *ctx, AVPacket *pkt)
drop = !(s->state % s->dropamount);
}
-av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %d pts %"PRId64" - amount
%d drop %d\n",
+av_log(ctx, AV_LOG_VERBOSE, "Stream #%d packet %u pts %"PRId64" - amount
%d drop %d\n",
pkt->stream_index, (unsigned int)s->var_values[VAR_N], pkt->pts,
amount, drop);
if (drop) {
--
2.44.0
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