[FFmpeg-cvslog] Changelog: Add Support hevc,vp9,av1 codec in enhanced flv format

2023-08-14 Thread Steven Liu 
ffmpeg | branch: master | Steven Liu  | Mon Aug 14 
15:20:00 2023 +0800| [a1928dff2c498ab9439d997bdc93fc98d2862cb0] | committer: 
Steven Liu

Changelog: Add Support hevc,vp9,av1 codec in enhanced flv format

Signed-off-by: Steven Liu 

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a1928dff2c498ab9439d997bdc93fc98d2862cb0
---

 Changelog | 1 +
 1 file changed, 1 insertion(+)

diff --git a/Changelog b/Changelog
index fa48d2811f..3d20acc44a 100644
--- a/Changelog
+++ b/Changelog
@@ -28,6 +28,7 @@ version :
 - scale_vt filter for videotoolbox
 - transpose_vt filter for videotoolbox
 - support for the P_SKIP hinting to speed up libx264 encoding
+- Support HEVC,VP9,AV1 codec in enhanced flv format
 
 version 6.0:
 - Radiance HDR image support

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[FFmpeg-cvslog] avfilter: add apsnr filter

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Sun Aug 13 02:57:57 
2023 +0200| [951def850abe9dc77311e5afd4b581defa1575bb] | committer: Paul B Mahol

avfilter: add apsnr filter

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=951def850abe9dc77311e5afd4b581defa1575bb
---

 doc/filters.texi |  7 +
 libavfilter/Makefile |  1 +
 libavfilter/af_asdr.c| 67 +---
 libavfilter/allfilters.c |  1 +
 4 files changed, 73 insertions(+), 3 deletions(-)

diff --git a/doc/filters.texi b/doc/filters.texi
index 43e9c037b9..5d1f10f95d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2836,6 +2836,13 @@ Default value is 8.
 
 This filter supports the all above options as @ref{commands}.
 
+@section apsnr
+Measure Audio Peak Signal-to-Noise Ratio.
+
+This filter takes two audio streams for input, and outputs first
+audio stream.
+Results are in dB per channel at end of either input.
+
 @section apsyclip
 Apply Psychoacoustic clipper to input audio stream.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 30a0e22ef8..9cd1407250 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -84,6 +84,7 @@ OBJS-$(CONFIG_APAD_FILTER)   += af_apad.o
 OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
 OBJS-$(CONFIG_APHASER_FILTER)+= af_aphaser.o 
generate_wave_table.o
 OBJS-$(CONFIG_APHASESHIFT_FILTER)+= af_afreqshift.o
+OBJS-$(CONFIG_APSNR_FILTER)  += af_asdr.o
 OBJS-$(CONFIG_APSYCLIP_FILTER)   += af_apsyclip.o
 OBJS-$(CONFIG_APULSATOR_FILTER)  += af_apulsator.o
 OBJS-$(CONFIG_AREALTIME_FILTER)  += f_realtime.o
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index 7d778b7f6b..b0401804f6 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -18,6 +18,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include 
+
 #include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 
@@ -27,6 +29,8 @@
 
 typedef struct AudioSDRContext {
 int channels;
+uint64_t nb_samples;
+double max;
 double *sum_u;
 double *sum_uv;
 
@@ -67,6 +71,34 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int 
jobnr, int nb_jobs)\
 SDR_FILTER(fltp, float)
 SDR_FILTER(dblp, double)
 
+#define PSNR_FILTER(name, type)   \
+static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
+{ \
+AudioSDRContext *s = ctx->priv;   \
+AVFrame *u = s->cache[0]; \
+AVFrame *v = s->cache[1]; \
+const int channels = u->ch_layout.nb_channels;\
+const int start = (channels * jobnr) / nb_jobs;   \
+const int end = (channels * (jobnr+1)) / nb_jobs; \
+const int nb_samples = u->nb_samples; \
+  \
+for (int ch = start; ch < end; ch++) {\
+const type *const us = (type *)u->extended_data[ch];  \
+const type *const vs = (type *)v->extended_data[ch];  \
+double sum_uv = 0.;   \
+  \
+for (int n = 0; n < nb_samples; n++)  \
+sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]);  \
+  \
+s->sum_uv[ch] += sum_uv;  \
+} \
+  \
+return 0; \
+}
+
+PSNR_FILTER(fltp, float)
+PSNR_FILTER(dblp, double)
+
 static int activate(AVFilterContext *ctx)
 {
 AudioSDRContext *s = ctx->priv;
@@ -97,6 +129,7 @@ static int activate(AVFilterContext *ctx)
 out = s->cache[0];
 s->cache[0] = NULL;
 
+s->nb_samples += available;
 return ff_filter_frame(outlink, out);
 }
 
@@ -126,7 +159,12 @@ static int config_output(AVFilterLink *outlink)
 AudioSDRContext *s = ctx->priv;
 
 s->channels = inlink->ch_layout.nb_channels;
-s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
+
+if (!strcmp(ctx->filter->name, "asdr"))
+s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
+else
+s->filter = inlink->format == AV_SAM

[FFmpeg-cvslog] avfilter/af_asdr: use single structure for sums

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Sun Aug 13 05:03:00 
2023 +0200| [10110a30b54c161fd267bb53b424b12aabdfcf70] | committer: Paul B Mahol

avfilter/af_asdr: use single structure for sums

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=10110a30b54c161fd267bb53b424b12aabdfcf70
---

 libavfilter/af_asdr.c | 44 
 1 file changed, 24 insertions(+), 20 deletions(-)

diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index 53069427bf..dbbb7e3419 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -27,13 +27,18 @@
 #include "filters.h"
 #include "internal.h"
 
+typedef struct ChanStats {
+double u;
+double v;
+double uv;
+} ChanStats;
+
 typedef struct AudioSDRContext {
 int channels;
 uint64_t nb_samples;
 double max;
-double *sum_u;
-double *sum_v;
-double *sum_uv;
+
+ChanStats *chs;
 
 AVFrame *cache[2];
 
@@ -52,6 +57,7 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int 
jobnr, int nb_jobs)\
 const int nb_samples = u->nb_samples; \
   \
 for (int ch = start; ch < end; ch++) {\
+ChanStats *chs = &s->chs[ch]; \
 const type *const us = (type *)u->extended_data[ch];  \
 const type *const vs = (type *)v->extended_data[ch];  \
 double sum_uv = 0.;   \
@@ -62,8 +68,8 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int 
jobnr, int nb_jobs)\
 sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]);  \
 } \
   \
-s->sum_uv[ch] += sum_uv;  \
-s->sum_u[ch]  += sum_u;   \
+chs->uv += sum_uv;\
+chs->u  += sum_u; \
 } \
   \
 return 0; \
@@ -84,6 +90,7 @@ static int sisdr_##name(AVFilterContext *ctx, void *arg,int 
jobnr,int nb_jobs)\
 const int nb_samples = u->nb_samples; \
   \
 for (int ch = start; ch < end; ch++) {\
+ChanStats *chs = &s->chs[ch]; \
 const type *const us = (type *)u->extended_data[ch];  \
 const type *const vs = (type *)v->extended_data[ch];  \
 double sum_uv = 0.;   \
@@ -96,9 +103,9 @@ static int sisdr_##name(AVFilterContext *ctx, void *arg,int 
jobnr,int nb_jobs)\
 sum_uv += us[n] * vs[n];  \
 } \
   \
-s->sum_uv[ch] += sum_uv;  \
-s->sum_u[ch]  += sum_u;   \
-s->sum_v[ch]  += sum_v;   \
+chs->uv += sum_uv;\
+chs->u  += sum_u; \
+chs->v  += sum_v; \
 } \
   \
 return 0; \
@@ -119,6 +126,7 @@ static int psnr_##name(AVFilterContext *ctx, void *arg, int 
jobnr,int nb_jobs)\
 const int nb_samples = u->nb_samples; \
   \
 for (int ch = start; ch < end; ch++) {\
+ChanStats *chs = &s->chs[ch]; \
 const type *const us = (type *)u->extended_data[ch];  \
 const type *const vs = (type *)v->extended_data[ch];  \
 double sum_uv = 0.;   \
@@ -126,7 +134,7 @@ static int psnr_##name(AVFilterContext *ctx, void *arg, i

[FFmpeg-cvslog] avfilter: add asisdr filter

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Sun Aug 13 04:19:08 
2023 +0200| [e41d52216cfe3537d7eadca863fb25838edd18c6] | committer: Paul B Mahol

avfilter: add asisdr filter

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e41d52216cfe3537d7eadca863fb25838edd18c6
---

 doc/filters.texi |  7 ++
 libavfilter/Makefile |  1 +
 libavfilter/af_asdr.c| 64 +++-
 libavfilter/allfilters.c |  1 +
 4 files changed, 72 insertions(+), 1 deletion(-)

diff --git a/doc/filters.texi b/doc/filters.texi
index 5d1f10f95d..cac1ee4381 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3141,6 +3141,13 @@ audio, the data is treated as if all the planes were 
concatenated.
 A list of Adler-32 checksums for each data plane.
 @end table
 
+@section asisdr
+Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.
+
+This filter takes two audio streams for input, and outputs first
+audio stream.
+Results are in dB per channel at end of either input.
+
 @section asoftclip
 Apply audio soft clipping.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 9cd1407250..2fe0033b21 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -102,6 +102,7 @@ OBJS-$(CONFIG_ASETRATE_FILTER)   += 
af_asetrate.o
 OBJS-$(CONFIG_ASETTB_FILTER) += settb.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)  += af_ashowinfo.o
 OBJS-$(CONFIG_ASIDEDATA_FILTER)  += f_sidedata.o
+OBJS-$(CONFIG_ASISDR_FILTER) += af_asdr.o
 OBJS-$(CONFIG_ASOFTCLIP_FILTER)  += af_asoftclip.o
 OBJS-$(CONFIG_ASPECTRALSTATS_FILTER) += af_aspectralstats.o
 OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index b0401804f6..53069427bf 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -32,6 +32,7 @@ typedef struct AudioSDRContext {
 uint64_t nb_samples;
 double max;
 double *sum_u;
+double *sum_v;
 double *sum_uv;
 
 AVFrame *cache[2];
@@ -71,6 +72,41 @@ static int sdr_##name(AVFilterContext *ctx, void *arg, int 
jobnr, int nb_jobs)\
 SDR_FILTER(fltp, float)
 SDR_FILTER(dblp, double)
 
+#define SISDR_FILTER(name, type)  \
+static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int nb_jobs)\
+{ \
+AudioSDRContext *s = ctx->priv;   \
+AVFrame *u = s->cache[0]; \
+AVFrame *v = s->cache[1]; \
+const int channels = u->ch_layout.nb_channels;\
+const int start = (channels * jobnr) / nb_jobs;   \
+const int end = (channels * (jobnr+1)) / nb_jobs; \
+const int nb_samples = u->nb_samples; \
+  \
+for (int ch = start; ch < end; ch++) {\
+const type *const us = (type *)u->extended_data[ch];  \
+const type *const vs = (type *)v->extended_data[ch];  \
+double sum_uv = 0.;   \
+double sum_u = 0.;\
+double sum_v = 0.;\
+  \
+for (int n = 0; n < nb_samples; n++) {\
+sum_u  += us[n] * us[n];  \
+sum_v  += vs[n] * vs[n];  \
+sum_uv += us[n] * vs[n];  \
+} \
+  \
+s->sum_uv[ch] += sum_uv;  \
+s->sum_u[ch]  += sum_u;   \
+s->sum_v[ch]  += sum_v;   \
+} \
+  \
+return 0; \
+}
+
+SISDR_FILTER(fltp, float)
+SISDR_FILTER(dblp, double)
+
 #define PSNR_FILTER(name, type)   \
 static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
 { \
@@ -162,13 +198,16 @@ static int config_output(A

[FFmpeg-cvslog] avfilter/af_asdr: remove wrong scaling from sdr, and fix sisdr formula

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Mon Aug 14 01:36:34 
2023 +0200| [24c013369dee69b87b517d6a7d71644d094a4e46] | committer: Paul B Mahol

avfilter/af_asdr: remove wrong scaling from sdr, and fix sisdr formula

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=24c013369dee69b87b517d6a7d71644d094a4e46
---

 libavfilter/af_asdr.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
index dbbb7e3419..3942f48053 100644
--- a/libavfilter/af_asdr.c
+++ b/libavfilter/af_asdr.c
@@ -225,11 +225,11 @@ static av_cold void uninit(AVFilterContext *ctx)
 
 if (!strcmp(ctx->filter->name, "asdr")) {
 for (int ch = 0; ch < s->channels; ch++)
-av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * 
log10(s->chs[ch].u / s->chs[ch].uv));
+av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 10. * 
log10(s->chs[ch].u / s->chs[ch].uv));
 } else if (!strcmp(ctx->filter->name, "asisdr")) {
 for (int ch = 0; ch < s->channels; ch++) {
 double scale = s->chs[ch].uv / s->chs[ch].v;
-double sisdr = s->chs[ch].u / fmax(0., s->chs[ch].u + 
scale*scale*s->chs[ch].v - 2.0*scale*s->chs[ch].uv);
+double sisdr = scale * scale * s->chs[ch].v / fmax(0., 
s->chs[ch].u + scale*scale*s->chs[ch].v - 2.0*scale*s->chs[ch].uv);
 
 av_log(ctx, AV_LOG_INFO, "SI-SDR ch%d: %g dB\n", ch, 10. * 
log10(sisdr));
 }

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[FFmpeg-cvslog] Bump minor libavfilter version and add Changelog entry

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Mon Aug 14 11:24:12 
2023 +0200| [c7049013247ac6c4851cf1b4ad6e22f0461a775a] | committer: Paul B Mahol

Bump minor libavfilter version and add Changelog entry

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c7049013247ac6c4851cf1b4ad6e22f0461a775a
---

 Changelog | 2 ++
 libavfilter/version.h | 2 +-
 2 files changed, 3 insertions(+), 1 deletion(-)

diff --git a/Changelog b/Changelog
index 3d20acc44a..c010e86159 100644
--- a/Changelog
+++ b/Changelog
@@ -29,6 +29,8 @@ version :
 - transpose_vt filter for videotoolbox
 - support for the P_SKIP hinting to speed up libx264 encoding
 - Support HEVC,VP9,AV1 codec in enhanced flv format
+- apsnr and asisdr audio filters
+
 
 version 6.0:
 - Radiance HDR image support
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 4a69d6be98..8f4a7a9cd3 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -31,7 +31,7 @@
 
 #include "version_major.h"
 
-#define LIBAVFILTER_VERSION_MINOR  10
+#define LIBAVFILTER_VERSION_MINOR  11
 #define LIBAVFILTER_VERSION_MICRO 100
 
 

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[FFmpeg-cvslog] configure: Set WIN32_LEAN_AND_MEAN at configure time

2023-08-14 Thread L . E . Segovia 
ffmpeg | branch: master | L. E. Segovia  | Mon Aug 14 
08:53:37 2023 -0300| [ddc1cd5cdd2570bf3d6ab807ee0ecfacdf09431d] | committer: 
Martin Storsjö

configure: Set WIN32_LEAN_AND_MEAN at configure time

Including winsock2.h or windows.h without WIN32_LEAN_AND_MEAN cause
bzlib.h to parse as nonsense, due to an instance of #define char small
in rpcndr.h.

See:

https://stackoverflow.com/a/27794577

Signed-off-by: L. E. Segovia 
Signed-off-by: Martin Storsjö 

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ddc1cd5cdd2570bf3d6ab807ee0ecfacdf09431d
---

 compat/atomics/win32/stdatomic.h | 1 -
 compat/w32pthreads.h | 1 -
 configure| 3 +++
 libavdevice/dshow_capture.h  | 1 -
 libavdevice/opengl_enc.c | 1 -
 libavfilter/vsrc_ddagrab.c   | 1 -
 libavformat/os_support.c | 6 ++
 libavutil/wchar_filename.h   | 1 -
 libswscale/utils.c   | 1 -
 9 files changed, 5 insertions(+), 11 deletions(-)

diff --git a/compat/atomics/win32/stdatomic.h b/compat/atomics/win32/stdatomic.h
index 28a627bfd3..4f8ac2bb60 100644
--- a/compat/atomics/win32/stdatomic.h
+++ b/compat/atomics/win32/stdatomic.h
@@ -19,7 +19,6 @@
 #ifndef COMPAT_ATOMICS_WIN32_STDATOMIC_H
 #define COMPAT_ATOMICS_WIN32_STDATOMIC_H
 
-#define WIN32_LEAN_AND_MEAN
 #include 
 #include 
 #include 
diff --git a/compat/w32pthreads.h b/compat/w32pthreads.h
index dae8d9420d..2ff9735227 100644
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -35,7 +35,6 @@
  * As most functions here are used without checking return values,
  * only implement return values as necessary. */
 
-#define WIN32_LEAN_AND_MEAN
 #include 
 #include 
 #include 
diff --git a/configure b/configure
index 932998b8d6..ff6a5c3600 100755
--- a/configure
+++ b/configure
@@ -5666,6 +5666,7 @@ case $target_os in
 fi
 ! enabled small && test_cmd $windres --version && enable gnu_windres
 enabled x86_32 && check_ldflags -Wl,--large-address-aware
+add_cppflags -DWIN32_LEAN_AND_MEAN
 shlibdir_default="$bindir_default"
 SLIBPREF=""
 SLIBSUF=".dll"
@@ -5716,6 +5717,7 @@ case $target_os in
 fi
 ! enabled small && test_cmd $windres --version && enable gnu_windres
 enabled x86_32 && check_ldflags -LARGEADDRESSAWARE
+add_cppflags -DWIN32_LEAN_AND_MEAN
 shlibdir_default="$bindir_default"
 SLIBPREF=""
 SLIBSUF=".dll"
@@ -5745,6 +5747,7 @@ case $target_os in
 enabled x86_64 && objformat="win64" || objformat="win32"
 enable dos_paths
 ! enabled small && test_cmd $windres --version && enable gnu_windres
+add_cppflags -DWIN32_LEAN_AND_MEAN
 add_cppflags -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600
 ;;
 *-dos|freedos|opendos)
diff --git a/libavdevice/dshow_capture.h b/libavdevice/dshow_capture.h
index b548cd7afc..81e684b9be 100644
--- a/libavdevice/dshow_capture.h
+++ b/libavdevice/dshow_capture.h
@@ -27,7 +27,6 @@
 #include "avdevice.h"
 
 #define COBJMACROS
-#define WIN32_LEAN_AND_MEAN
 #include 
 #define NO_DSHOW_STRSAFE
 #include 
diff --git a/libavdevice/opengl_enc.c b/libavdevice/opengl_enc.c
index f9dc846bf1..1b0cf5aa8f 100644
--- a/libavdevice/opengl_enc.c
+++ b/libavdevice/opengl_enc.c
@@ -30,7 +30,6 @@
 #include "config.h"
 
 #if HAVE_WINDOWS_H
-#define WIN32_LEAN_AND_MEAN
 #include 
 #endif
 #if HAVE_OPENGL_GL3_H
diff --git a/libavfilter/vsrc_ddagrab.c b/libavfilter/vsrc_ddagrab.c
index b65261f685..c3107e11b4 100644
--- a/libavfilter/vsrc_ddagrab.c
+++ b/libavfilter/vsrc_ddagrab.c
@@ -22,7 +22,6 @@
 #undef _WIN32_WINNT
 #define _WIN32_WINNT 0x0A00
 #endif
-#define WIN32_LEAN_AND_MEAN
 
 #include 
 
diff --git a/libavformat/os_support.c b/libavformat/os_support.c
index 15cea7fa5b..2de6a7c3d9 100644
--- a/libavformat/os_support.c
+++ b/libavformat/os_support.c
@@ -34,11 +34,9 @@
 #if HAVE_SYS_TIME_H
 #include 
 #endif /* HAVE_SYS_TIME_H */
-#if HAVE_WINSOCK2_H
-#include 
-#elif HAVE_SYS_SELECT_H
+#if HAVE_SYS_SELECT_H
 #include 
-#endif /* HAVE_WINSOCK2_H */
+#endif /* HAVE_SYS_SELECT_H */
 #endif /* !HAVE_POLL_H */
 
 #include "network.h"
diff --git a/libavutil/wchar_filename.h b/libavutil/wchar_filename.h
index fbc0a55146..4f87e6f88b 100644
--- a/libavutil/wchar_filename.h
+++ b/libavutil/wchar_filename.h
@@ -21,7 +21,6 @@
 
 #ifdef _WIN32
 
-#define WIN32_LEAN_AND_MEAN
 #include 
 #include "mem.h"
 
diff --git a/libswscale/utils.c b/libswscale/utils.c
index 8fe34a3437..8e74c6603e 100644
--- a/libswscale/utils.c
+++ b/libswscale/utils.c
@@ -34,7 +34,6 @@
 #endif
 #endif
 #if HAVE_VIRTUALALLOC
-#define WIN32_LEAN_AND_MEAN
 #include 
 #endif
 

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[FFmpeg-cvslog] avcodec/wavpack: remove hard limitation on number of supported channels

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Tue Aug 15 00:12:52 
2023 +0200| [46412a8935e4632b2460988bfce4152c7dccce22] | committer: Paul B Mahol

avcodec/wavpack: remove hard limitation on number of supported channels

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=46412a8935e4632b2460988bfce4152c7dccce22
---

 libavcodec/wavpack.c | 20 +---
 1 file changed, 5 insertions(+), 15 deletions(-)

diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c
index 4346304f54..71e7d40c81 100644
--- a/libavcodec/wavpack.c
+++ b/libavcodec/wavpack.c
@@ -96,12 +96,10 @@ typedef struct WavpackFrameContext {
 uint8_t *value_lookup[MAX_HISTORY_BINS];
 } WavpackFrameContext;
 
-#define WV_MAX_FRAME_DECODERS 14
-
 typedef struct WavpackContext {
 AVCodecContext *avctx;
 
-WavpackFrameContext *fdec[WV_MAX_FRAME_DECODERS];
+WavpackFrameContext **fdec;
 int fdec_num;
 
 int block;
@@ -971,7 +969,8 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, 
GetBitContext *gb,
 
 static av_cold int wv_alloc_frame_context(WavpackContext *c)
 {
-if (c->fdec_num == WV_MAX_FRAME_DECODERS)
+c->fdec = av_realloc_f(c->fdec, c->fdec_num + 1, sizeof(*c->fdec));
+if (!c->fdec)
 return -1;
 
 c->fdec[c->fdec_num] = av_mallocz(sizeof(**c->fdec));
@@ -1064,6 +1063,7 @@ static av_cold int wavpack_decode_end(AVCodecContext 
*avctx)
 
 for (int i = 0; i < s->fdec_num; i++)
 av_freep(&s->fdec[i]);
+av_freep(&s->fdec);
 s->fdec_num = 0;
 
 ff_thread_release_ext_buffer(avctx, &s->curr_frame);
@@ -1415,18 +1415,12 @@ static int wavpack_decode_block(AVCodecContext *avctx, 
int block_no,
 size = bytestream2_get_byte(&gb);
 chan  |= (bytestream2_get_byte(&gb) & 0xF) << 8;
 chan  += 1;
-if (avctx->ch_layout.nb_channels != chan)
-av_log(avctx, AV_LOG_WARNING, "%i channels signalled"
-   " instead of %i.\n", chan, 
avctx->ch_layout.nb_channels);
 chmask = bytestream2_get_le24(&gb);
 break;
 case 5:
 size = bytestream2_get_byte(&gb);
 chan  |= (bytestream2_get_byte(&gb) & 0xF) << 8;
 chan  += 1;
-if (avctx->ch_layout.nb_channels != chan)
-av_log(avctx, AV_LOG_WARNING, "%i channels signalled"
-   " instead of %i.\n", chan, 
avctx->ch_layout.nb_channels);
 chmask = bytestream2_get_le32(&gb);
 break;
 default:
@@ -1519,11 +1513,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, 
int block_no,
 return AVERROR_INVALIDDATA;
 }
 } else {
-ret = av_channel_layout_copy(&new_ch_layout, 
&avctx->ch_layout);
-if (ret < 0) {
-av_log(avctx, AV_LOG_ERROR, "Error copying channel 
layout\n");
-return ret;
-}
+av_channel_layout_default(&new_ch_layout, chan);
 }
 } else {
 av_channel_layout_default(&new_ch_layout, s->stereo + 1);

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[FFmpeg-cvslog] avcodec/wavpackenc: add support for encoding unspec channel layouts

2023-08-14 Thread Paul B Mahol 
ffmpeg | branch: master | Paul B Mahol  | Tue Aug 15 00:27:40 
2023 +0200| [a3b434e1515ecb0de0c4b92c6b7659e510b980c2] | committer: Paul B Mahol

avcodec/wavpackenc: add support for encoding unspec channel layouts

Also write 0 for chmask if mask have bits outside of first 32 bits,
as wavpack does not support more bits than 32 for channel layouts.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a3b434e1515ecb0de0c4b92c6b7659e510b980c2
---

 libavcodec/wavpackenc.c | 11 ++-
 1 file changed, 10 insertions(+), 1 deletion(-)

diff --git a/libavcodec/wavpackenc.c b/libavcodec/wavpackenc.c
index 9271e87990..33a5dfcc89 100644
--- a/libavcodec/wavpackenc.c
+++ b/libavcodec/wavpackenc.c
@@ -2592,7 +2592,16 @@ static int wavpack_encode_block(WavPackEncodeContext *s,
 s->avctx->ch_layout.u.mask != AV_CH_LAYOUT_STEREO) {
 put_metadata_block(&pb, WP_ID_CHANINFO, 5);
 bytestream2_put_byte(&pb, s->avctx->ch_layout.nb_channels);
-bytestream2_put_le32(&pb, s->avctx->ch_layout.u.mask);
+if (s->avctx->ch_layout.u.mask >> 32)
+bytestream2_put_le32(&pb, 0);
+else
+bytestream2_put_le32(&pb, s->avctx->ch_layout.u.mask);
+bytestream2_put_byte(&pb, 0);
+} else if (s->flags & WV_INITIAL_BLOCK &&
+   s->avctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) {
+put_metadata_block(&pb, WP_ID_CHANINFO, 5);
+bytestream2_put_byte(&pb, s->avctx->ch_layout.nb_channels);
+bytestream2_put_le32(&pb, 0);
 bytestream2_put_byte(&pb, 0);
 }
 

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