[FFmpeg-cvslog] avcodec/avpacket: Don't write into non-writable buffer
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Feb 12 13:27:17 2020 +0100| [8f51a89d66aacd9dc5896bac22e62cbd566e7a71] |
committer: Andreas Rheinhardt
avcodec/avpacket: Don't write into non-writable buffer
The data of an AVPacket may be a part of the data of an AVBufferRef;
Therefore av_grow_packet() doesn't reallocate if the available space in
the actual buffer is sufficient for the enlargement. But given that it
also zeroes the padding it also needs to make sure that the buffer is
actually writable; this commit implements this.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8f51a89d66aacd9dc5896bac22e62cbd566e7a71
---
libavcodec/avpacket.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/libavcodec/avpacket.c b/libavcodec/avpacket.c
index 55b509108e..ee51c0799c 100644
--- a/libavcodec/avpacket.c
+++ b/libavcodec/avpacket.c
@@ -128,7 +128,8 @@ int av_grow_packet(AVPacket *pkt, int grow_by)
return AVERROR(ENOMEM);
}
-if (new_size + data_offset > pkt->buf->size) {
+if (new_size + data_offset > pkt->buf->size ||
+!av_buffer_is_writable(pkt->buf)) {
int ret = av_buffer_realloc(&pkt->buf, new_size + data_offset);
if (ret < 0) {
pkt->data = old_data;
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[FFmpeg-cvslog] tools: fix const specifier for AVInputFormat
ffmpeg | branch: master | Josh de Kock | Wed Apr 29 11:45:49
2020 +0100| [d817b57d36795b22dc493ef66e3ed14a583cae49] | committer: Josh de Kock
tools: fix const specifier for AVInputFormat
Signed-off-by: Josh de Kock
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d817b57d36795b22dc493ef66e3ed14a583cae49
---
tools/probetest.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/tools/probetest.c b/tools/probetest.c
index cfa309cabd..6f0e002b74 100644
--- a/tools/probetest.c
+++ b/tools/probetest.c
@@ -66,7 +66,7 @@ static void probe(AVProbeData *pd, int type, int p, int size)
static void print_times(void)
{
int i = 0;
-AVInputFormat *fmt = NULL;
+const AVInputFormat *fmt = NULL;
void *fmt_opaque = NULL;
while ((fmt = av_demuxer_iterate(&fmt_opaque))) {
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[FFmpeg-cvslog] avfilter/af_crossfeed: make slope configurable
ffmpeg | branch: master | Paul B Mahol | Thu Apr 30 11:58:04
2020 +0200| [35d6001815e1282c5babf71c71f1fd04b59e5440] | committer: Paul B Mahol
avfilter/af_crossfeed: make slope configurable
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=35d6001815e1282c5babf71c71f1fd04b59e5440
---
doc/filters.texi | 4
libavfilter/af_crossfeed.c | 4 +++-
2 files changed, 7 insertions(+), 1 deletion(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index be224bd5b9..8ff32f49bf 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3267,6 +3267,10 @@ Set soundstage wideness. Default is 0.5. Allowed range
is from 0 to 1.
This sets cut off frequency of low shelf filter. Default is cut off near
1550 Hz. With range set to 1 cut off frequency is set to 2100 Hz.
+@item slope
+Set curve slope of low shelf filter. Default is 0.5.
+Allowed range is from 0.01 to 1.
+
@item level_in
Set input gain. Default is 0.9.
diff --git a/libavfilter/af_crossfeed.c b/libavfilter/af_crossfeed.c
index f1bbb357ba..cc2db70e62 100644
--- a/libavfilter/af_crossfeed.c
+++ b/libavfilter/af_crossfeed.c
@@ -28,6 +28,7 @@ typedef struct CrossfeedContext {
double range;
double strength;
+double slope;
double level_in;
double level_out;
@@ -62,7 +63,7 @@ static int config_input(AVFilterLink *inlink)
double w0 = 2 * M_PI * (1. - s->range) * 2100 / inlink->sample_rate;
double alpha;
-alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / 0.5 - 1) + 2);
+alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / s->slope - 1) + 2);
s->a0 = (A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha;
s->a1 =-2 * ((A - 1) + (A + 1) * cos(w0));
@@ -139,6 +140,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVOption crossfeed_options[] = {
{ "strength", "set crossfeed strength", OFFSET(strength),
AV_OPT_TYPE_DOUBLE, {.dbl=.2}, 0, 1, FLAGS },
{ "range", "set soundstage wideness", OFFSET(range),
AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
+{ "slope", "set curve slope", OFFSET(slope),
AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .01, 1, FLAGS },
{ "level_in", "set level in",OFFSET(level_in),
AV_OPT_TYPE_DOUBLE, {.dbl=.9}, 0, 1, FLAGS },
{ "level_out", "set level out", OFFSET(level_out),
AV_OPT_TYPE_DOUBLE, {.dbl=1.}, 0, 1, FLAGS },
{ NULL }
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[FFmpeg-cvslog] avfilter/af_crossfeed: fix calculation of alpha parameter
ffmpeg | branch: master | Paul B Mahol | Thu Apr 30 11:44:48 2020 +0200| [0ec61fcff71d212b616660a6e93d856ef1849466] | committer: Paul B Mahol avfilter/af_crossfeed: fix calculation of alpha parameter Use A in calculation. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0ec61fcff71d212b616660a6e93d856ef1849466 --- libavfilter/af_crossfeed.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavfilter/af_crossfeed.c b/libavfilter/af_crossfeed.c index c819ca59a7..f1bbb357ba 100644 --- a/libavfilter/af_crossfeed.c +++ b/libavfilter/af_crossfeed.c @@ -62,7 +62,7 @@ static int config_input(AVFilterLink *inlink) double w0 = 2 * M_PI * (1. - s->range) * 2100 / inlink->sample_rate; double alpha; -alpha = sin(w0) / 2 * sqrt(2 * (1 / 0.5 - 1) + 2); +alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / 0.5 - 1) + 2); s->a0 = (A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha; s->a1 =-2 * ((A - 1) + (A + 1) * cos(w0)); ___ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email ffmpeg-cvslog-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-cvslog] avfilter: add asubboost filter
ffmpeg | branch: master | Paul B Mahol | Wed Apr 22 15:29:16
2020 +0200| [c7d8082357965bfd02bc3a8ba6867b36119d2895] | committer: Paul B Mahol
avfilter: add asubboost filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c7d8082357965bfd02bc3a8ba6867b36119d2895
---
Changelog | 1 +
doc/filters.texi | 39
libavfilter/Makefile | 1 +
libavfilter/af_asubboost.c | 232 +
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 275 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 83b8a4a46e..9b3e34560f 100644
--- a/Changelog
+++ b/Changelog
@@ -63,6 +63,7 @@ version :
- maskedthreshold filter
- Support for muxing pcm and pgs in m2ts
- Cunning Developments ADPCM decoder
+- asubboost filter
version 4.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index f8dc02f00d..be224bd5b9 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2454,6 +2454,45 @@ Number of points where the waveform crosses the zero
level axis.
Rate of Zero crossings and number of audio samples.
@end table
+@section asubboost
+Boost subwoofer frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item dry
+Set dry gain, how much of original signal is kept. Allowed range is from 0 to
1.
+Default value is 0.5.
+
+@item wet
+Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to
1.
+Default value is 0.8.
+
+@item decay
+Set delay line decay gain value. Allowed range is from 0 to 1.
+Default value is 0.7.
+
+@item feedback
+Set delay line feedback gain value. Allowed range is from 0 to 1.
+Default value is 0.5.
+
+@item cutoff
+Set cutoff frequency in herz. Allowed range is 50 to 900.
+Default value is 100.
+
+@item slope
+Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1.
+Default value is 0.5.
+
+@item delay
+Set delay. Allowed range is from 1 to 100.
+Default value is 20.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section atempo
Adjust audio tempo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e1205eb063..f982afe15f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -86,6 +86,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASR_FILTER)+= af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
+OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER)+= af_axcorrelate.o
diff --git a/libavfilter/af_asubboost.c b/libavfilter/af_asubboost.c
new file mode 100644
index 00..f8369fd818
--- /dev/null
+++ b/libavfilter/af_asubboost.c
@@ -0,0 +1,232 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct ASubBoostContext {
+const AVClass *class;
+
+double dry_gain;
+double wet_gain;
+double feedback;
+double decay;
+double delay;
+double cutoff;
+double slope;
+
+double a0, a1, a2;
+double b0, b1, b2;
+
+int write_pos;
+int buffer_samples;
+
+AVFrame *i, *o;
+AVFrame *buffer;
+} ASubBoostContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+AVFilterFormats *formats = NULL;
+AVFilterChannelLayouts *layouts = NULL;
+static const enum AVSampleFormat sample_fmts[] = {
+AV_SAMPLE_FMT_DBLP,
+AV_SAMPLE_FMT_NONE
+};
+int ret;
+
+formats = ff_make_format_list(sample_fmts);
+if (!formats)
+return AVERROR(ENOMEM);
+ret = ff_set_common_formats(ctx, formats);
+if (ret < 0)
+return ret;
+
+layouts = ff_all_channel_counts();
+if (!layouts)
+return AVERROR(ENOMEM);
+
+ret = ff_set_common_channel_layouts(ctx, layouts);
+if (ret < 0)
+return ret;
+
+formats = ff_all_sam
[FFmpeg-cvslog] avfilter/af_crossfeed: make options runtime configurable
ffmpeg | branch: master | Paul B Mahol | Thu Apr 30 12:04:27
2020 +0200| [0a181c2cb1fe4038f900d3e1e7c8106e0326f784] | committer: Paul B Mahol
avfilter/af_crossfeed: make options runtime configurable
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0a181c2cb1fe4038f900d3e1e7c8106e0326f784
---
doc/filters.texi | 4
libavfilter/af_crossfeed.c | 15 ++-
2 files changed, 18 insertions(+), 1 deletion(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 8ff32f49bf..d19fd346ae 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3278,6 +3278,10 @@ Set input gain. Default is 0.9.
Set output gain. Default is 1.
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section crystalizer
Simple algorithm to expand audio dynamic range.
diff --git a/libavfilter/af_crossfeed.c b/libavfilter/af_crossfeed.c
index cc2db70e62..70dd26eb28 100644
--- a/libavfilter/af_crossfeed.c
+++ b/libavfilter/af_crossfeed.c
@@ -134,8 +134,20 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
return ff_filter_frame(outlink, out);
}
+static int process_command(AVFilterContext *ctx, const char *cmd, const char
*args,
+ char *res, int res_len, int flags)
+{
+int ret;
+
+ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+if (ret < 0)
+return ret;
+
+return config_input(ctx->inputs[0]);
+}
+
#define OFFSET(x) offsetof(CrossfeedContext, x)
-#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define FLAGS
AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption crossfeed_options[] = {
{ "strength", "set crossfeed strength", OFFSET(strength),
AV_OPT_TYPE_DOUBLE, {.dbl=.2}, 0, 1, FLAGS },
@@ -175,4 +187,5 @@ AVFilter ff_af_crossfeed = {
.inputs = inputs,
.outputs= outputs,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+.process_command = process_command,
};
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[FFmpeg-cvslog] avfoundation: Fix building on iOS
ffmpeg | branch: master | Martin Storsjö | Thu Apr 30
13:34:13 2020 +0300| [c810a9502cebe32e1dd08ee3d0d17053dde44aa9] | committer:
Thilo Borgmann
avfoundation: Fix building on iOS
Apparently the changes from 3c9185bf3a83395d12a987f626dbdb985eac4320
aren't enough; even with that in place, I got errors like this
when trying to build for iOS:
src/libavdevice/avfoundation.m:135:5: error:
'AVCaptureDeviceTransportControlsPlaybackMode' is unavailable: not
available on iOS
AVCaptureDeviceTransportControlsPlaybackMode observed_mode;
^
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c810a9502cebe32e1dd08ee3d0d17053dde44aa9
---
libavdevice/avfoundation.m | 4
1 file changed, 4 insertions(+)
diff --git a/libavdevice/avfoundation.m b/libavdevice/avfoundation.m
index 0b1888693f..59d5b0af4f 100644
--- a/libavdevice/avfoundation.m
+++ b/libavdevice/avfoundation.m
@@ -132,7 +132,9 @@ typedef struct
CMSampleBufferRef current_audio_frame;
AVCaptureDevice *observed_device;
+#if !TARGET_OS_IPHONE && __MAC_OS_X_VERSION_MIN_REQUIRED >= 1070
AVCaptureDeviceTransportControlsPlaybackMode observed_mode;
+#endif
int observed_quit;
} AVFContext;
@@ -200,6 +202,7 @@ static void unlock_frames(AVFContext* ctx)
change:(NSDictionary *)change
context:(void *)context {
if (context == _context) {
+#if !TARGET_OS_IPHONE && __MAC_OS_X_VERSION_MIN_REQUIRED >= 1070
AVCaptureDeviceTransportControlsPlaybackMode mode =
[change[NSKeyValueChangeNewKey] integerValue];
@@ -209,6 +212,7 @@ static void unlock_frames(AVFContext* ctx)
}
_context->observed_mode = mode;
}
+#endif
} else {
[super observeValueForKeyPath: keyPath
ofObject: object
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[FFmpeg-cvslog] avcodec/v4l2_context: Log warning when all capture buffers are in userspace
ffmpeg | branch: master | Andriy Gelman | Thu Apr 30
10:58:37 2020 -0400| [2a9d62356152d4ef079416101664f26d2562c681] | committer:
Andriy Gelman
avcodec/v4l2_context: Log warning when all capture buffers are in userspace
v4l2_m2m uses device memory mapped buffers to store dequeued
frames/packets (reference counted by AVBufferRef). When the reference
count drops to zero, the buffer ownership is returned back to the
device, so that they can re-filled with frames/packets.
There are some cases when all the capture buffers are in userspace
(i.e. due to internal buffering in ffmpeg). On the s5p-mfc this causes
an infinite wait when polling to dequeue the buffers, which can be
prevented by increasing the total number of buffers. This commit adds a
warning when all the capture buffers are dequeued.
Reviewed-by: Ming Qian
Signed-off-by: Andriy Gelman
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2a9d62356152d4ef079416101664f26d2562c681
---
libavcodec/v4l2_context.c | 12
1 file changed, 12 insertions(+)
diff --git a/libavcodec/v4l2_context.c b/libavcodec/v4l2_context.c
index 6c2db5c849..f70e151ec8 100644
--- a/libavcodec/v4l2_context.c
+++ b/libavcodec/v4l2_context.c
@@ -291,6 +291,18 @@ static V4L2Buffer* v4l2_dequeue_v4l2buf(V4L2Context *ctx,
int timeout)
};
int i, ret;
+if (!V4L2_TYPE_IS_OUTPUT(ctx->type) && ctx->buffers) {
+for (i = 0; i < ctx->num_buffers; i++) {
+if (ctx->buffers[i].status == V4L2BUF_IN_DRIVER)
+break;
+}
+if (i == ctx->num_buffers)
+av_log(logger(ctx), AV_LOG_WARNING, "All capture buffers returned
to "
+"userspace. Increase
num_capture_buffers "
+"to prevent device deadlock or
dropped "
+"packets/frames.\n");
+}
+
/* if we are draining and there are no more capture buffers queued in the
driver we are done */
if (!V4L2_TYPE_IS_OUTPUT(ctx->type) && ctx_to_m2mctx(ctx)->draining) {
for (i = 0; i < ctx->num_buffers; i++) {
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[FFmpeg-cvslog] avcodec/cbs_h265: fix writing extension_data bits
ffmpeg | branch: master | James Almer | Mon Apr 20 15:25:58 2020 -0300| [38d1815cc65dd447de80760895ee008cfc9a0091] | committer: James Almer avcodec/cbs_h265: fix writing extension_data bits We only care about the right most bit. Signed-off-by: James Almer > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=38d1815cc65dd447de80760895ee008cfc9a0091 --- libavcodec/cbs_h265_syntax_template.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavcodec/cbs_h265_syntax_template.c b/libavcodec/cbs_h265_syntax_template.c index 5f5531944c..0696eeeb9e 100644 --- a/libavcodec/cbs_h265_syntax_template.c +++ b/libavcodec/cbs_h265_syntax_template.c @@ -80,7 +80,7 @@ static int FUNC(extension_data)(CodedBitstreamContext *ctx, RWContext *rw, } #else for (k = 0; k < current->bit_length; k++) -xu(1, extension_data, current->data[k / 8] >> (7 - k % 8), 0, 1, 0); +xu(1, extension_data, current->data[k / 8] >> (7 - k % 8) & 1, 0, 1, 0); #endif return 0; } ___ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email ffmpeg-cvslog-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-cvslog] avformat/utils: change the duration to int64_t for update_initial_durations
ffmpeg | branch: master | Limin Wang | Wed Apr 29
07:49:24 2020 +0800| [8afa03a5f682cec9925b302a092d66a0532ae328] | committer:
Limin Wang
avformat/utils: change the duration to int64_t for update_initial_durations
Reviewed-by: Michael Niedermayer
Signed-off-by: Limin Wang
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8afa03a5f682cec9925b302a092d66a0532ae328
---
libavformat/utils.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 3b53f97bee..44109d866d 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -1174,7 +1174,7 @@ static void update_initial_timestamps(AVFormatContext *s,
int stream_index,
}
static void update_initial_durations(AVFormatContext *s, AVStream *st,
- int stream_index, int duration)
+ int stream_index, int64_t duration)
{
AVPacketList *pktl = s->internal->packet_buffer ?
s->internal->packet_buffer : s->internal->parse_queue;
int64_t cur_dts= RELATIVE_TS_BASE;
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[FFmpeg-cvslog] oggdec: add support for proper demuxing of chained Opus files and streams
ffmpeg | branch: master | Lynne | Tue Apr 28 12:25:46 2020
+0100| [8296443a70f052a6f5c9a867d28b83a5eb7d304d] | committer: Lynne
oggdec: add support for proper demuxing of chained Opus files and streams
Part of this patch is based on Paul B Mahol's patch from last year.
This also allows for single-stream parameter/codec changes.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8296443a70f052a6f5c9a867d28b83a5eb7d304d
---
libavformat/oggdec.c | 45 +
libavformat/oggdec.h | 1 +
libavformat/oggparseopus.c | 1 +
3 files changed, 27 insertions(+), 20 deletions(-)
diff --git a/libavformat/oggdec.c b/libavformat/oggdec.c
index 92dcafe2ed..c591bafddd 100644
--- a/libavformat/oggdec.c
+++ b/libavformat/oggdec.c
@@ -178,6 +178,7 @@ static int ogg_reset(AVFormatContext *s)
if (start_pos <= s->internal->data_offset) {
os->lastpts = 0;
}
+os->start_trimming = 0;
os->end_trimming = 0;
av_freep(&os->new_metadata);
os->new_metadata_size = 0;
@@ -206,7 +207,8 @@ static const struct ogg_codec *ogg_find_codec(uint8_t *buf,
int size)
* situation where a new audio stream spawn (identified with a new serial) and
* must replace the previous one (track switch).
*/
-static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, char *magic)
+static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, char *magic,
+ int probing)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *os;
@@ -220,24 +222,25 @@ static int ogg_replace_stream(AVFormatContext *s,
uint32_t serial, char *magic)
/* Check for codecs */
codec = ogg_find_codec(magic, 8);
-if (!codec) {
+if (!codec && !probing) {
av_log(s, AV_LOG_ERROR, "Cannot identify new stream\n");
return AVERROR_INVALIDDATA;
}
-/* If the codec matches, then we assume its a replacement */
-for (i = 0; i < ogg->nstreams; i++) {
-if (ogg->streams[i].codec == codec)
-break;
-}
-
-/* Otherwise, create a new stream */
-if (i >= ogg->nstreams)
-return ogg_new_stream(s, serial);
-
-os = &ogg->streams[i];
-os->serial = serial;
-os->codec = codec;
+/* We only have a single stream anyway, so if there's a new stream with
+ * a different codec just replace it */
+os = &ogg->streams[0];
+os->serial = serial;
+os->codec = codec;
+os->serial = serial;
+os->lastpts = 0;
+os->lastdts = 0;
+os->start_trimming = 0;
+os->end_trimming = 0;
+
+/* Chained files have extradata as a new packet */
+if (codec == &ff_opus_codec)
+os->header = -1;
return i;
}
@@ -294,7 +297,7 @@ static int data_packets_seen(const struct ogg *ogg)
return 0;
}
-static int ogg_read_page(AVFormatContext *s, int *sid)
+static int ogg_read_page(AVFormatContext *s, int *sid, int probing)
{
AVIOContext *bc = s->pb;
struct ogg *ogg = s->priv_data;
@@ -417,7 +420,7 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
/* CRC is correct so we can be 99% sure there's an actual change here */
if (idx < 0) {
if (data_packets_seen(ogg))
-idx = ogg_replace_stream(s, serial, readout_buf);
+idx = ogg_replace_stream(s, serial, readout_buf, probing);
else
idx = ogg_new_stream(s, serial);
@@ -492,7 +495,7 @@ static int ogg_packet(AVFormatContext *s, int *sid, int
*dstart, int *dsize,
idx = ogg->curidx;
while (idx < 0) {
-ret = ogg_read_page(s, &idx);
+ret = ogg_read_page(s, &idx, 0);
if (ret < 0)
return ret;
}
@@ -643,7 +646,7 @@ static int ogg_get_length(AVFormatContext *s)
avio_seek(s->pb, end, SEEK_SET);
ogg->page_pos = -1;
-while (!ogg_read_page(s, &i)) {
+while (!ogg_read_page(s, &i, 1)) {
if (ogg->streams[i].granule != -1 && ogg->streams[i].granule != 0 &&
ogg->streams[i].codec) {
s->streams[i]->duration =
@@ -847,13 +850,15 @@ retry:
pkt->duration = os->pduration;
pkt->pos = fpos;
-if (os->end_trimming) {
+if (os->start_trimming || os->end_trimming) {
uint8_t *side_data = av_packet_new_side_data(pkt,
AV_PKT_DATA_SKIP_SAMPLES,
10);
if(!side_data)
return AVERROR(ENOMEM);
+ AV_WL32(side_data + 0, os->start_trimming);
AV_WL32(side_data + 4, os->end_trimming);
+os->start_trimming = 0;
os->end_trimming = 0;
}
diff --git a/libavformat/oggdec.h b/libavformat/oggdec.h
index 4a2b6ddee8..e2057c46f6 100644
--- a/libavformat/oggdec.h
+++ b/libavformat/oggdec.h
@@ -84,6 +84,7 @@ struct ogg_stream {
int got_start;
int got_data; ///< 1 if the stream got some
[FFmpeg-cvslog] oggdec: use ffio_ensure_seekback() to seek back on incorrect data
ffmpeg | branch: master | Lynne | Tue Apr 28 12:55:17 2020
+0100| [e983197cbc93420b67aa7e811be47d7278c2c8a2] | committer: Lynne
oggdec: use ffio_ensure_seekback() to seek back on incorrect data
This cleans up the code and simplifies it.
It also speeds up parsing since the old pb position was incorrect.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e983197cbc93420b67aa7e811be47d7278c2c8a2
---
libavformat/oggdec.c | 68 +---
1 file changed, 27 insertions(+), 41 deletions(-)
diff --git a/libavformat/oggdec.c b/libavformat/oggdec.c
index e0188c7c59..92dcafe2ed 100644
--- a/libavformat/oggdec.c
+++ b/libavformat/oggdec.c
@@ -206,59 +206,40 @@ static const struct ogg_codec *ogg_find_codec(uint8_t
*buf, int size)
* situation where a new audio stream spawn (identified with a new serial) and
* must replace the previous one (track switch).
*/
-static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, int size)
+static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, char *magic)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *os;
const struct ogg_codec *codec;
int i = 0;
-if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) {
-uint8_t magic[8];
-avio_seek(s->pb, -size, SEEK_CUR);
-if (avio_read(s->pb, magic, sizeof(magic)) != sizeof(magic))
-return AVERROR_INVALIDDATA;
-avio_seek(s->pb, size - sizeof(magic), SEEK_CUR);
-codec = ogg_find_codec(magic, sizeof(magic));
-if (!codec) {
-av_log(s, AV_LOG_ERROR, "Cannot identify new stream\n");
-return AVERROR_INVALIDDATA;
-}
-for (i = 0; i < ogg->nstreams; i++) {
-if (ogg->streams[i].codec == codec)
-break;
-}
-if (i >= ogg->nstreams)
-return ogg_new_stream(s, serial);
-} else if (ogg->nstreams != 1) {
+if (ogg->nstreams != 1) {
avpriv_report_missing_feature(s, "Changing stream parameters in
multistream ogg");
return AVERROR_PATCHWELCOME;
}
-os = &ogg->streams[i];
-
-os->serial = serial;
-return i;
+/* Check for codecs */
+codec = ogg_find_codec(magic, 8);
+if (!codec) {
+av_log(s, AV_LOG_ERROR, "Cannot identify new stream\n");
+return AVERROR_INVALIDDATA;
+}
-#if 0
-buf = os->buf;
-bufsize = os->bufsize;
-codec = os->codec;
+/* If the codec matches, then we assume its a replacement */
+for (i = 0; i < ogg->nstreams; i++) {
+if (ogg->streams[i].codec == codec)
+break;
+}
-if (!ogg->state || ogg->state->streams[i].private != os->private)
-av_freep(&ogg->streams[i].private);
+/* Otherwise, create a new stream */
+if (i >= ogg->nstreams)
+return ogg_new_stream(s, serial);
-/* Set Ogg stream settings similar to what is done in ogg_new_stream(). We
- * also re-use the ogg_stream allocated buffer */
-memset(os, 0, sizeof(*os));
-os->serial = serial;
-os->bufsize = bufsize;
-os->buf = buf;
-os->header = -1;
-os->codec = codec;
+os = &ogg->streams[i];
+os->serial = serial;
+os->codec = codec;
return i;
-#endif
}
static int ogg_new_stream(AVFormatContext *s, uint32_t serial)
@@ -325,6 +306,7 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
uint32_t crc, crc_tmp;
int size = 0, idx;
int64_t version, page_pos;
+int64_t start_pos;
uint8_t sync[4];
uint8_t segments[255];
uint8_t *readout_buf;
@@ -364,6 +346,10 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
/* 0x4fa9b05f = av_crc(AV_CRC_32_IEEE, 0x0, "OggS", 4) */
ffio_init_checksum(bc, ff_crc04C11DB7_update, 0x4fa9b05f);
+/* To rewind if checksum is bad/check magic on switches - this is the max
packet size */
+ffio_ensure_seekback(bc, MAX_PAGE_SIZE);
+start_pos = avio_tell(bc);
+
version = avio_r8(bc);
flags = avio_r8(bc);
gp = avio_rl64(bc);
@@ -414,7 +400,7 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
av_log(s, AV_LOG_ERROR, "CRC mismatch!\n");
if (idx < 0)
av_free(readout_buf);
-avio_seek(bc, -size, SEEK_CUR);
+avio_seek(bc, start_pos, SEEK_SET);
return 0;
}
@@ -424,14 +410,14 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
av_log(s, AV_LOG_ERROR, "Invalid Ogg vers!\n");
if (idx < 0)
av_free(readout_buf);
-avio_seek(bc, -size, SEEK_CUR);
+avio_seek(bc, start_pos, SEEK_SET);
return 0;
}
/* CRC is correct so we can be 99% sure there's an actual change here */
if (idx < 0) {
if (data_packets_seen(ogg))
-idx = ogg_replace_stream(s, serial, size);
+idx = ogg_replace_stream(s, serial, readout_buf);
else
idx = ogg_new_stream(s, serial);
[FFmpeg-cvslog] oggdec: eliminate copies and extra buffers
ffmpeg | branch: master | Lynne | Tue Apr 28 12:41:34 2020
+0100| [f619e1ec66b89215582eff4404b681b760540b4f] | committer: Lynne
oggdec: eliminate copies and extra buffers
This also makes implementing CRC checking far simpler and more robust.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f619e1ec66b89215582eff4404b681b760540b4f
---
libavformat/oggdec.c | 127 +++
1 file changed, 58 insertions(+), 69 deletions(-)
diff --git a/libavformat/oggdec.c b/libavformat/oggdec.c
index 95190589ab..7db26840b2 100644
--- a/libavformat/oggdec.c
+++ b/libavformat/oggdec.c
@@ -205,7 +205,7 @@ static const struct ogg_codec *ogg_find_codec(uint8_t *buf,
int size)
* situation where a new audio stream spawn (identified with a new serial) and
* must replace the previous one (track switch).
*/
-static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, int nsegs)
+static int ogg_replace_stream(AVFormatContext *s, uint32_t serial, int size)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *os;
@@ -214,11 +214,10 @@ static int ogg_replace_stream(AVFormatContext *s,
uint32_t serial, int nsegs)
if (s->pb->seekable & AVIO_SEEKABLE_NORMAL) {
uint8_t magic[8];
-int64_t pos = avio_tell(s->pb);
-avio_skip(s->pb, nsegs);
+avio_seek(s->pb, -size, SEEK_CUR);
if (avio_read(s->pb, magic, sizeof(magic)) != sizeof(magic))
return AVERROR_INVALIDDATA;
-avio_seek(s->pb, pos, SEEK_SET);
+avio_seek(s->pb, size - sizeof(magic), SEEK_CUR);
codec = ogg_find_codec(magic, sizeof(magic));
if (!codec) {
av_log(s, AV_LOG_ERROR, "Cannot identify new stream\n");
@@ -303,27 +302,6 @@ static int ogg_new_stream(AVFormatContext *s, uint32_t
serial)
return idx;
}
-static int ogg_new_buf(struct ogg *ogg, int idx)
-{
-struct ogg_stream *os = ogg->streams + idx;
-uint8_t *nb = av_malloc(os->bufsize + AV_INPUT_BUFFER_PADDING_SIZE);
-int size = os->bufpos - os->pstart;
-
-if (!nb)
-return AVERROR(ENOMEM);
-
-if (os->buf) {
-memcpy(nb, os->buf + os->pstart, size);
-av_free(os->buf);
-}
-
-os->buf= nb;
-os->bufpos = size;
-os->pstart = 0;
-
-return 0;
-}
-
static int data_packets_seen(const struct ogg *ogg)
{
int i;
@@ -343,8 +321,11 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
int flags, nsegs;
uint64_t gp;
uint32_t serial;
-int size, idx;
+int size = 0, idx;
+int64_t page_pos;
uint8_t sync[4];
+uint8_t segments[255];
+uint8_t *readout_buf;
int sp = 0;
ret = avio_read(bc, sync, 4);
@@ -387,47 +368,73 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
gp = avio_rl64(bc);
serial = avio_rl32(bc);
avio_skip(bc, 8); /* seq, crc */
-nsegs = avio_r8(bc);
+
+nsegs= avio_r8(bc);
+page_pos = avio_tell(bc) - 27;
+
+ret = avio_read(bc, segments, nsegs);
+if (ret < nsegs)
+return ret < 0 ? ret : AVERROR_EOF;
if (avio_feof(bc))
return AVERROR_EOF;
+for (i = 0; i < nsegs; i++)
+size += segments[i];
+
idx = ogg_find_stream(ogg, serial);
+if (idx >= 0) {
+os = ogg->streams + idx;
+
+/* Even if invalid guarantee there's enough memory to read the page */
+if (os->bufsize - os->bufpos < size) {
+uint8_t *nb = av_realloc(os->buf, 2*os->bufsize +
AV_INPUT_BUFFER_PADDING_SIZE);
+if (!nb)
+return AVERROR(ENOMEM);
+os->buf = nb;
+os->bufsize *= 2;
+}
+
+readout_buf = os->buf + os->bufpos;
+} else {
+readout_buf = av_malloc(size);
+}
+
+ret = avio_read(bc, readout_buf, size);
+if (ret < size) {
+if (idx < 0)
+av_free(readout_buf);
+return ret < 0 ? ret : AVERROR_EOF;
+}
+
if (idx < 0) {
if (data_packets_seen(ogg))
-idx = ogg_replace_stream(s, serial, nsegs);
+idx = ogg_replace_stream(s, serial, size);
else
idx = ogg_new_stream(s, serial);
if (idx < 0) {
av_log(s, AV_LOG_ERROR, "failed to create or replace stream\n");
+av_free(readout_buf);
return idx;
}
-}
-os = ogg->streams + idx;
-ogg->page_pos =
-os->page_pos = avio_tell(bc) - 27;
+os = ogg->streams + idx;
-if (os->psize > 0) {
-ret = ogg_new_buf(ogg, idx);
-if (ret < 0)
-return ret;
+memcpy(os->buf + os->bufpos, readout_buf, size);
+av_free(readout_buf);
}
-ret = avio_read(bc, os->segments, nsegs);
-if (ret < nsegs)
-return ret < 0 ? ret : AVERROR_EOF;
-
-os->nsegs = nsegs;
-os->segp = 0;
-
-size = 0;
-for (i = 0; i < nsegs; i++)
-size += os->segments[i];
-
-if (!(flags & OGG_FLAG_BOS))
-
[FFmpeg-cvslog] oggdec: verify page checksum
ffmpeg | branch: master | Lynne | Tue Apr 28 12:52:11 2020
+0100| [9ad47762c17d2c6d06595aa17b88112baa91b72c] | committer: Lynne
oggdec: verify page checksum
This makes decoding far more robust, since OggS, the ogg magic,
can be commonly found randomly in streams, which previously made
the demuxer think there's a new stream or a change in such.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9ad47762c17d2c6d06595aa17b88112baa91b72c
---
libavformat/oggdec.c | 46 ++
1 file changed, 34 insertions(+), 12 deletions(-)
diff --git a/libavformat/oggdec.c b/libavformat/oggdec.c
index 7db26840b2..e0188c7c59 100644
--- a/libavformat/oggdec.c
+++ b/libavformat/oggdec.c
@@ -31,6 +31,7 @@
#include
#include "libavutil/avassert.h"
#include "libavutil/intreadwrite.h"
+#include "avio_internal.h"
#include "oggdec.h"
#include "avformat.h"
#include "internal.h"
@@ -321,8 +322,9 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
int flags, nsegs;
uint64_t gp;
uint32_t serial;
+uint32_t crc, crc_tmp;
int size = 0, idx;
-int64_t page_pos;
+int64_t version, page_pos;
uint8_t sync[4];
uint8_t segments[255];
uint8_t *readout_buf;
@@ -359,15 +361,19 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
return AVERROR_INVALIDDATA;
}
-if (avio_r8(bc) != 0) { /* version */
-av_log (s, AV_LOG_ERROR, "ogg page, unsupported version\n");
-return AVERROR_INVALIDDATA;
-}
+/* 0x4fa9b05f = av_crc(AV_CRC_32_IEEE, 0x0, "OggS", 4) */
+ffio_init_checksum(bc, ff_crc04C11DB7_update, 0x4fa9b05f);
-flags = avio_r8(bc);
-gp = avio_rl64(bc);
-serial = avio_rl32(bc);
-avio_skip(bc, 8); /* seq, crc */
+version = avio_r8(bc);
+flags = avio_r8(bc);
+gp = avio_rl64(bc);
+serial = avio_rl32(bc);
+avio_skip(bc, 4); /* seq */
+
+crc_tmp = ffio_get_checksum(bc);
+crc = avio_rb32(bc);
+crc_tmp = ff_crc04C11DB7_update(crc_tmp, (uint8_t[4]){0}, 4);
+ffio_init_checksum(bc, ff_crc04C11DB7_update, crc_tmp);
nsegs= avio_r8(bc);
page_pos = avio_tell(bc) - 27;
@@ -376,9 +382,6 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
if (ret < nsegs)
return ret < 0 ? ret : AVERROR_EOF;
-if (avio_feof(bc))
-return AVERROR_EOF;
-
for (i = 0; i < nsegs; i++)
size += segments[i];
@@ -407,6 +410,25 @@ static int ogg_read_page(AVFormatContext *s, int *sid)
return ret < 0 ? ret : AVERROR_EOF;
}
+if (crc ^ ffio_get_checksum(bc)) {
+av_log(s, AV_LOG_ERROR, "CRC mismatch!\n");
+if (idx < 0)
+av_free(readout_buf);
+avio_seek(bc, -size, SEEK_CUR);
+return 0;
+}
+
+/* Since we're almost sure its a valid packet, checking the version after
+ * the checksum lets the demuxer be more tolerant */
+if (version) {
+av_log(s, AV_LOG_ERROR, "Invalid Ogg vers!\n");
+if (idx < 0)
+av_free(readout_buf);
+avio_seek(bc, -size, SEEK_CUR);
+return 0;
+}
+
+/* CRC is correct so we can be 99% sure there's an actual change here */
if (idx < 0) {
if (data_packets_seen(ogg))
idx = ogg_replace_stream(s, serial, size);
___
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[FFmpeg-cvslog] avfilter/vf_mix: Check sscanf() return value
ffmpeg | branch: master | Limin Wang | Sun Mar 29
09:08:13 2020 +0800| [ee5d6d2ef88a3a7fd480435cf931dd6165bbb688] | committer:
Limin Wang
avfilter/vf_mix: Check sscanf() return value
Reviewed-by: Nicolas George
Signed-off-by: Limin Wang
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ee5d6d2ef88a3a7fd480435cf931dd6165bbb688
---
libavfilter/vf_mix.c | 5 -
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/libavfilter/vf_mix.c b/libavfilter/vf_mix.c
index f418c6fba8..bdb67d8e5a 100644
--- a/libavfilter/vf_mix.c
+++ b/libavfilter/vf_mix.c
@@ -108,7 +108,10 @@ static av_cold int init(AVFilterContext *ctx)
break;
p = NULL;
-av_sscanf(arg, "%f", &s->weights[i]);
+if (av_sscanf(arg, "%f", &s->weights[i]) != 1) {
+av_log(ctx, AV_LOG_ERROR, "Invalid syntax for weights[%d].\n", i);
+return AVERROR(EINVAL);
+}
s->wfactor += s->weights[i];
last = i;
}
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[FFmpeg-cvslog] avfilter/af_adelay: Check sscanf() return value
ffmpeg | branch: master | Limin Wang | Sun Mar 29
08:08:05 2020 +0800| [f441fadbcffa32b71470220b522d4971a774e760] | committer:
Limin Wang
avfilter/af_adelay: Check sscanf() return value
Reviewed-by: Michael Niedermayer
Signed-off-by: Limin Wang
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f441fadbcffa32b71470220b522d4971a774e760
---
libavfilter/af_adelay.c | 5 -
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/libavfilter/af_adelay.c b/libavfilter/af_adelay.c
index c9647771f2..6ac81c2a3e 100644
--- a/libavfilter/af_adelay.c
+++ b/libavfilter/af_adelay.c
@@ -155,7 +155,10 @@ static int config_input(AVFilterLink *inlink)
ret = av_sscanf(arg, "%d%c", &d->delay, &type);
if (ret != 2 || type != 'S') {
div = type == 's' ? 1.0 : 1000.0;
-av_sscanf(arg, "%f", &delay);
+if (av_sscanf(arg, "%f", &delay) != 1) {
+av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");
+return AVERROR(EINVAL);
+}
d->delay = delay * inlink->sample_rate / div;
}
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[FFmpeg-cvslog] fate/oggopus-demux: fix fate failure
ffmpeg | branch: master | Lynne | Fri May 1 01:36:43 2020 +0100| [4fd0559b77cfdd38d37a94a45cc830d90454194e] | committer: Lynne fate/oggopus-demux: fix fate failure Failure was due to the extra comment printed by libavcodec/utils.c since side data is used to signal the skipped samples. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4fd0559b77cfdd38d37a94a45cc830d90454194e --- tests/ref/fate/oggopus-demux | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tests/ref/fate/oggopus-demux b/tests/ref/fate/oggopus-demux index 9192760700..580758c0dc 100644 --- a/tests/ref/fate/oggopus-demux +++ b/tests/ref/fate/oggopus-demux @@ -5,7 +5,7 @@ #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo -0, -356, -356, 960, 402, 0x89b1c40f +0, -356, -356, 960, 402, 0x89b1c40f, S=1, 10, 0x03f10065 0,604,604, 960, 216, 0x7bf97146 0, 1564, 1564, 960, 215, 0x6cb86d8b 0, 2524, 2524, 960, 218, 0x9cfd691c ___ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email ffmpeg-cvslog-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-cvslog] avformat/mpegts: fix av_freep for dovi pointer
ffmpeg | branch: master | Limin Wang | Thu Apr 30
21:20:28 2020 +0800| [9dd2587f60015a211f3120233e44e829ffd66c6f] | committer:
Jun Zhao
avformat/mpegts: fix av_freep for dovi pointer
Reviewed-by: Andreas Rheinhardt
Signed-off-by: Limin Wang
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9dd2587f60015a211f3120233e44e829ffd66c6f
---
libavformat/mpegts.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index ff3898c3a5..0833d62ea5 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -2180,7 +2180,7 @@ int ff_parse_mpeg2_descriptor(AVFormatContext *fc,
AVStream *st, int stream_type
ret = av_stream_add_side_data(st, AV_PKT_DATA_DOVI_CONF,
(uint8_t *)dovi, dovi_size);
if (ret < 0) {
-av_freep(dovi);
+av_free(dovi);
return ret;
}
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[FFmpeg-cvslog] avformat/mov: fix av_freep for dovi pointer
ffmpeg | branch: master | Limin Wang | Thu Apr 30
21:19:11 2020 +0800| [1112823962ab2e881359e5d60d706ba322ea8551] | committer:
Jun Zhao
avformat/mov: fix av_freep for dovi pointer
Reviewed-by: Andreas Rheinhardt
Signed-off-by: Limin Wang
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1112823962ab2e881359e5d60d706ba322ea8551
---
libavformat/mov.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/mov.c b/libavformat/mov.c
index 3d6fef685d..ad718cdaa2 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -6827,7 +6827,7 @@ static int mov_read_dvcc_dvvc(MOVContext *c, AVIOContext
*pb, MOVAtom atom)
ret = av_stream_add_side_data(st, AV_PKT_DATA_DOVI_CONF,
(uint8_t *)dovi, dovi_size);
if (ret < 0) {
-av_freep(dovi);
+av_free(dovi);
return ret;
}
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[FFmpeg-cvslog] avformat/matroskadec: Reject sipr flavor > 3
ffmpeg | branch: master | Andreas Rheinhardt |
Mon Dec 2 10:41:12 2019 +0100| [8287c201536e52f2765cfa9a70551814a6f36ebb] |
committer: Andreas Rheinhardt
avformat/matroskadec: Reject sipr flavor > 3
Only flavors 0..3 seem to exist. E.g. rmdec.c treats any flavor > 3
as invalid data. Furthermore, we do not know how big the packets to
create ought to be given that for sipr these values are not read from
the bitstream, but from a table.
Furthermore, flavor is only used for sipr, so only check it for sipr;
rmdec.c does the same. (The old check for flavor being < 0 was
always wrong given that flavor is an int that is read via avio_rb16(),
so it has been removed completely.)
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8287c201536e52f2765cfa9a70551814a6f36ebb
---
libavformat/matroskadec.c | 16 +---
1 file changed, 9 insertions(+), 7 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 8e1326abf6..8c65e98e77 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2606,28 +2606,30 @@ static int matroska_parse_tracks(AVFormatContext *s)
track->audio.sub_packet_h= avio_rb16(&b);
track->audio.frame_size = avio_rb16(&b);
track->audio.sub_packet_size = avio_rb16(&b);
-if (flavor< 0 ||
-track->audio.coded_framesize <= 0 ||
+if (track->audio.coded_framesize <= 0 ||
track->audio.sub_packet_h<= 0 ||
track->audio.frame_size <= 0 ||
track->audio.sub_packet_size <= 0 && codec_id !=
AV_CODEC_ID_SIPR)
return AVERROR_INVALIDDATA;
-track->audio.buf = av_malloc_array(track->audio.sub_packet_h,
- track->audio.frame_size);
-if (!track->audio.buf)
-return AVERROR(ENOMEM);
+
if (codec_id == AV_CODEC_ID_RA_288) {
st->codecpar->block_align = track->audio.coded_framesize;
track->codec_priv.size = 0;
} else {
-if (codec_id == AV_CODEC_ID_SIPR && flavor < 4) {
+if (codec_id == AV_CODEC_ID_SIPR) {
static const int sipr_bit_rate[4] = { 6504, 8496, 5000,
16000 };
+if (flavor > 3)
+return AVERROR_INVALIDDATA;
track->audio.sub_packet_size = ff_sipr_subpk_size[flavor];
st->codecpar->bit_rate = sipr_bit_rate[flavor];
}
st->codecpar->block_align = track->audio.sub_packet_size;
extradata_offset = 78;
}
+track->audio.buf = av_malloc_array(track->audio.sub_packet_h,
+ track->audio.frame_size);
+if (!track->audio.buf)
+return AVERROR(ENOMEM);
} else if (codec_id == AV_CODEC_ID_FLAC && track->codec_priv.size) {
ret = matroska_parse_flac(s, track, &extradata_offset);
if (ret < 0)
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[FFmpeg-cvslog] avformat/matroskadec: Simplify check for RealAudio
ffmpeg | branch: master | Andreas Rheinhardt |
Mon Dec 2 11:04:55 2019 +0100| [c6f60b90f00da7d942a7dbeb513ed284953f6f20] |
committer: Andreas Rheinhardt
avformat/matroskadec: Simplify check for RealAudio
They need a special parsing mode and in order to find out whether this
mode is in use, several checks have to be performed. They can all be
combined into one: If the buffer that is only used to assemble their
packets has been allocated, use the RealAudio parsing mode.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c6f60b90f00da7d942a7dbeb513ed284953f6f20
---
libavformat/matroskadec.c | 6 +-
1 file changed, 1 insertion(+), 5 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 8c65e98e77..668323ba08 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -3617,11 +3617,7 @@ static int matroska_parse_block(MatroskaDemuxContext
*matroska, AVBufferRef *buf
for (n = 0; n < laces; n++) {
int64_t lace_duration = block_duration*(n+1) / laces -
block_duration*n / laces;
-if ((st->codecpar->codec_id == AV_CODEC_ID_RA_288 ||
- st->codecpar->codec_id == AV_CODEC_ID_COOK ||
- st->codecpar->codec_id == AV_CODEC_ID_SIPR ||
- st->codecpar->codec_id == AV_CODEC_ID_ATRAC3) &&
-st->codecpar->block_align && track->audio.sub_packet_size) {
+if (track->audio.buf) {
res = matroska_parse_rm_audio(matroska, track, st, data,
lace_size[n],
timecode, pos);
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[FFmpeg-cvslog] avformat/matroskadec: Fix demuxing RealAudio 28.8
ffmpeg | branch: master | Andreas Rheinhardt |
Mon Apr 20 07:47:06 2020 +0200| [c91e3690d9c6667123b116c9fd3becf5f4f4530e] |
committer: Andreas Rheinhardt
avformat/matroskadec: Fix demuxing RealAudio 28.8
RealAudio 28.8 does not need or use sub_packet_size for its demuxing
and this field is therefore commonly set to zero. But since 18ca491b
the Real Audio specific demuxing is no longer applied if sub_packet_size
is zero because the codepath for cook and ATRAC3 divide by it; this made
these files undecodable.
Furthermore, since 569d18aa (merged in 2c8d876d) sub_packet_size being
zero is used as an indicator for invalid data, so that a file containing
such a track was completely skipped.
This commit fixes this by not checking sub_packet_size for RealAudio
28.8 at all.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c91e3690d9c6667123b116c9fd3becf5f4f4530e
---
libavformat/matroskadec.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 668323ba08..844f96cd52 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2608,8 +2608,7 @@ static int matroska_parse_tracks(AVFormatContext *s)
track->audio.sub_packet_size = avio_rb16(&b);
if (track->audio.coded_framesize <= 0 ||
track->audio.sub_packet_h<= 0 ||
-track->audio.frame_size <= 0 ||
-track->audio.sub_packet_size <= 0 && codec_id !=
AV_CODEC_ID_SIPR)
+track->audio.frame_size <= 0)
return AVERROR_INVALIDDATA;
if (codec_id == AV_CODEC_ID_RA_288) {
@@ -2622,7 +2621,8 @@ static int matroska_parse_tracks(AVFormatContext *s)
return AVERROR_INVALIDDATA;
track->audio.sub_packet_size = ff_sipr_subpk_size[flavor];
st->codecpar->bit_rate = sipr_bit_rate[flavor];
-}
+} else if (track->audio.sub_packet_size <= 0)
+return AVERROR_INVALIDDATA;
st->codecpar->block_align = track->audio.sub_packet_size;
extradata_offset = 78;
}
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[FFmpeg-cvslog] avformat/matroskadec: Fix buffer overflow when demuxing RealAudio 28.8
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Apr 22 00:15:54 2020 +0200| [4f5c6c1b0ec2407dfd42fcfa3441067de1962a54] |
committer: Andreas Rheinhardt
avformat/matroskadec: Fix buffer overflow when demuxing RealAudio 28.8
RealAudio 28.8 (like other RealAudio codecs) uses a special demuxing
mode in which the data of the existing Matroska Blocks is not simply
forwarded as-is. Instead data from several Blocks is recombined
together to output several packets. The parameters governing this
process are parsed from the CodecPrivate: Coded framesize (cfs), frame
size (w) and sub_packet_h (h).
During demuxing, h/2 pieces of data of size cfs each are read from every
Matroska (Simple)Block and put at offset m * 2 * w + n * cfs of a buffer
of size h * w, where m ranges from 0 to h/2 - 1 for each Block while n
is initially zero and incremented after a Block has been parsed until it
is h, at which poin the assembled packets are output and n reset.
The highest offset is given by (h/2 - 1) * 2 * w + (h - 1) * cfs + cfs
while the destination buffer's size is given by h * w. For even h, this
leads to a buffer overflow (and potential segfault) if h * cfs > 2 * w;
for odd h, the condition is h * cfs > 3 * w.
This commit adds a check to rule this out.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4f5c6c1b0ec2407dfd42fcfa3441067de1962a54
---
libavformat/matroskadec.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 844f96cd52..951695b5b5 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2612,6 +2612,9 @@ static int matroska_parse_tracks(AVFormatContext *s)
return AVERROR_INVALIDDATA;
if (codec_id == AV_CODEC_ID_RA_288) {
+if ((int64_t)track->audio.sub_packet_h *
track->audio.coded_framesize
+> (2 + (track->audio.sub_packet_h & 1)) *
track->audio.frame_size)
+return AVERROR_INVALIDDATA;
st->codecpar->block_align = track->audio.coded_framesize;
track->codec_priv.size = 0;
} else {
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[FFmpeg-cvslog] avformat/matroskadec: Don't output uninitialized data for RealAudio 28.8
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Apr 22 15:17:41 2020 +0200| [bdaa98dd4aac08b8f23f959cbb5a80db2dacd14a] |
committer: Andreas Rheinhardt
avformat/matroskadec: Don't output uninitialized data for RealAudio 28.8
The Matroska demuxer splits every sequence of h Matroska Blocks into
h * w / cfs packets of size cfs; here h (sub_packet_h), w (frame_size)
and cfs (coded_framesize) are parameters from the track's CodecPrivate.
It does this by splitting the Block's data in h/2 pieces of size cfs each
and putting them into a buffer at offset m * 2 * w + n * cfs where
m (range 0..(h/2 - 1)) indicates the index of the current piece in the
current Block and n (range 0..(h - 1)) is the index of the current Block
in the current sequence of Blocks. The data in this buffer is then used
for the output packets.
The problem is that there is currently no check to actually guarantee
that no uninitialized data will be output. One instance where this is
trivially so is if h == 1; another is if cfs * h is so small that the
input pieces do not cover everything that is output. In order to
preclude this, rmdec.c checks for h * cfs == 2 * w and h >= 2. The
former requirement certainly makes much sense, as it means that for
every given m the input pieces (corresponding to the h different values
of n) form a nonoverlapping partition of the two adjacent frames of size w
corresponding to m. But precluding h == 1 is not enough, other odd
values can cause problems, too. That is because the assumption behind
the code is that h frames of size w contain data to be output, although
the real number is h/2 * 2. E.g. for h = 3, cfs = 2 and w = 3 the
current code would output four (== h * w / cfs) packets. although only
data for three (== h/2 * h) packets has been read.
(Notice that if h * cfs == 2 * w, h being even is equivalent to
cfs dividing w; the latter condition also seems very reasonable:
It means that the subframes are a partition of the frames.)
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bdaa98dd4aac08b8f23f959cbb5a80db2dacd14a
---
libavformat/matroskadec.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 951695b5b5..1dc0b77962 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2612,8 +2612,8 @@ static int matroska_parse_tracks(AVFormatContext *s)
return AVERROR_INVALIDDATA;
if (codec_id == AV_CODEC_ID_RA_288) {
-if ((int64_t)track->audio.sub_packet_h *
track->audio.coded_framesize
-> (2 + (track->audio.sub_packet_h & 1)) *
track->audio.frame_size)
+if (track->audio.sub_packet_h & 1 || 2 *
track->audio.frame_size
+!= (int64_t)track->audio.sub_packet_h *
track->audio.coded_framesize)
return AVERROR_INVALIDDATA;
st->codecpar->block_align = track->audio.coded_framesize;
track->codec_priv.size = 0;
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[FFmpeg-cvslog] avformat/matroskadec: Simplify checks for cook and ATRAC3
ffmpeg | branch: master | Andreas Rheinhardt |
Mon Apr 20 08:54:23 2020 +0200| [4b1c19a054432b4a5bc3e90b8c186dc25a416ebd] |
committer: Andreas Rheinhardt
avformat/matroskadec: Simplify checks for cook and ATRAC3
Some conditions which don't change and which can therefore be checked
in read_header() were instead rechecked upon parsing each block. This
has been changed.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4b1c19a054432b4a5bc3e90b8c186dc25a416ebd
---
libavformat/matroskadec.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 1dc0b77962..5643e15a20 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2624,7 +2624,8 @@ static int matroska_parse_tracks(AVFormatContext *s)
return AVERROR_INVALIDDATA;
track->audio.sub_packet_size = ff_sipr_subpk_size[flavor];
st->codecpar->bit_rate = sipr_bit_rate[flavor];
-} else if (track->audio.sub_packet_size <= 0)
+} else if (track->audio.sub_packet_size <= 0 ||
+ track->audio.frame_size %
track->audio.sub_packet_size)
return AVERROR_INVALIDDATA;
st->codecpar->block_align = track->audio.sub_packet_size;
extradata_offset = 78;
@@ -3138,7 +3139,7 @@ static int matroska_parse_rm_audio(MatroskaDemuxContext
*matroska,
}
memcpy(track->audio.buf + y * w, data, w);
} else {
-if (size < sps * w / sps || h<=0 || w%sps) {
+if (size < w) {
av_log(matroska->ctx, AV_LOG_ERROR,
"Corrupt generic RM-style audio packet size\n");
return AVERROR_INVALIDDATA;
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[FFmpeg-cvslog] avformat/matroskadec: Don't discard valid packets
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Mar 25 06:00:53 2020 +0100| [e471faf96230076f67e393df9d1a90a08c22a055] |
committer: Andreas Rheinhardt
avformat/matroskadec: Don't discard valid packets
A Block (meaning both a Block in a BlockGroup as well as a SimpleBlock)
must have at least three bytes after the field containing the encoded
TrackNumber. So if there are <= 3 bytes, the Matroska demuxer would
skip this block, believing it to be an empty, but valid Block.
This might discard valid nonempty Blocks, namely if the track uses header
stripping. And certain definitely spec-incompliant Blocks don't raise
errors: Those with two or less bytes left after the encoded TrackNumber
and those with three bytes left, but with flags indicating that the Block
uses lacing as then there has to be further data describing the lacing.
Furthermore, zero-sized packets were still possible because only the
size of the last entry of a lace was checked.
This commit fixes this. All spec-compliant Blocks that contain data
(even if side data only) are now returned to the caller; spec-compliant
Blocks that don't contain anything are not returned.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e471faf96230076f67e393df9d1a90a08c22a055
---
libavformat/matroskadec.c | 18 --
1 file changed, 12 insertions(+), 6 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 5643e15a20..3b1b447d8a 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -3020,7 +3020,9 @@ static int matroska_parse_laces(MatroskaDemuxContext
*matroska, uint8_t **buf,
return 0;
}
-av_assert0(size > 0);
+if (size <= 0)
+return AVERROR_INVALIDDATA;
+
*laces= *data + 1;
data += 1;
size -= 1;
@@ -3046,7 +3048,7 @@ static int matroska_parse_laces(MatroskaDemuxContext
*matroska, uint8_t **buf,
break;
}
}
-if (size <= total) {
+if (size < total) {
return AVERROR_INVALIDDATA;
}
@@ -3093,7 +3095,7 @@ static int matroska_parse_laces(MatroskaDemuxContext
*matroska, uint8_t **buf,
}
data += offset;
size -= offset;
-if (size <= total) {
+if (size < total) {
return AVERROR_INVALIDDATA;
}
lace_size[*laces - 1] = size - total;
@@ -3413,7 +3415,7 @@ static int matroska_parse_frame(MatroskaDemuxContext
*matroska,
{
MatroskaTrackEncoding *encodings = track->encodings.elem;
uint8_t *pkt_data = data;
-int res;
+int res = 0;
AVPacket pktl, *pkt = &pktl;
if (encodings && !encodings->type && encodings->scope & 1) {
@@ -3449,6 +3451,9 @@ static int matroska_parse_frame(MatroskaDemuxContext
*matroska,
pkt_data = pr_data;
}
+if (!pkt_size && !additional_size)
+goto no_output;
+
av_init_packet(pkt);
if (pkt_data != data)
pkt->buf = av_buffer_create(pkt_data, pkt_size +
AV_INPUT_BUFFER_PADDING_SIZE,
@@ -3519,6 +3524,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
return 0;
+no_output:
fail:
if (pkt_data != data)
av_freep(&pkt_data);
@@ -3554,8 +3560,8 @@ static int matroska_parse_block(MatroskaDemuxContext
*matroska, AVBufferRef *buf
av_log(matroska->ctx, AV_LOG_INFO,
"Invalid stream %"PRIu64"\n", num);
return AVERROR_INVALIDDATA;
-} else if (size <= 3)
-return 0;
+} else if (size < 3)
+return AVERROR_INVALIDDATA;
st = track->stream;
if (st->discard >= AVDISCARD_ALL)
return res;
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[FFmpeg-cvslog] avformat/matroskadec: Improve forward compability
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Mar 25 06:52:09 2020 +0100| [b577968cabae4a0927adcf5d7c24fca5a7a8385d] |
committer: Andreas Rheinhardt
avformat/matroskadec: Improve forward compability
Matroska is built around the principle that a reader does not need to
understand everything in a file in order to be able to make use of it;
it just needs to ignore the data it doesn't know about.
Our demuxer typically follows this principle, but there is one important
instance where it does not: A Block belonging to a TrackEntry with no
associated stream is treated as invalid data (i.e. the demuxer will try
to resync to the next level 1 element because it takes this as a sign
that it has lost sync). Given that we do not create streams if we don't
know or don't support the type of the TrackEntry, this impairs this
demuxer's forward compability.
Furthermore, ignoring Blocks belonging to a TrackEntry without
corresponding stream can (in future commits) also be used to ignore
TrackEntries with obviously bogus entries without affecting the other
TrackEntries (by not creating a stream for said TrackEntry).
Finally, given that matroska_find_track_by_num() already emits its own
error message in case there is no TrackEntry with a given TrackNumber,
the error message (with level AV_LOG_INFO) for this can be removed.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b577968cabae4a0927adcf5d7c24fca5a7a8385d
---
libavformat/matroskadec.c | 15 +--
1 file changed, 9 insertions(+), 6 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 3b1b447d8a..088eeabf2d 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -3556,13 +3556,16 @@ static int matroska_parse_block(MatroskaDemuxContext
*matroska, AVBufferRef *buf
size -= n;
track = matroska_find_track_by_num(matroska, num);
-if (!track || !track->stream) {
-av_log(matroska->ctx, AV_LOG_INFO,
- "Invalid stream %"PRIu64"\n", num);
+if (!track || size < 3)
return AVERROR_INVALIDDATA;
-} else if (size < 3)
-return AVERROR_INVALIDDATA;
-st = track->stream;
+
+if (!(st = track->stream)) {
+av_log(matroska->ctx, AV_LOG_VERBOSE,
+ "No stream associated to TrackNumber %"PRIu64". "
+ "Ignoring Block with this TrackNumber.\n", num);
+return 0;
+}
+
if (st->discard >= AVDISCARD_ALL)
return res;
av_assert1(block_duration != AV_NOPTS_VALUE);
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[FFmpeg-cvslog] avformat/matroskadec: Cache whether a track needs to be decoded
ffmpeg | branch: master | Andreas Rheinhardt |
Wed Dec 4 17:54:45 2019 +0100| [96012d17a9f5003f2695e137c4876485e2fdb03a] |
committer: Andreas Rheinhardt
avformat/matroskadec: Cache whether a track needs to be decoded
There is no need to recheck this for every frame.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=96012d17a9f5003f2695e137c4876485e2fdb03a
---
libavformat/matroskadec.c | 9 +++--
1 file changed, 7 insertions(+), 2 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 088eeabf2d..9e3e98e9c0 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -249,6 +249,7 @@ typedef struct MatroskaTrack {
AVStream *stream;
int64_t end_timecode;
int ms_compat;
+int needs_decoding;
uint64_t max_block_additional_id;
uint32_t palette[AVPALETTE_COUNT];
@@ -2405,6 +2406,11 @@ static int matroska_parse_tracks(AVFormatContext *s)
}
}
}
+track->needs_decoding = encodings && !encodings[0].type &&
+encodings[0].scope & 1 &&
+(encodings[0].compression.algo !=
+ MATROSKA_TRACK_ENCODING_COMP_HEADERSTRIP ||
+ encodings[0].compression.settings.size);
for (j = 0; ff_mkv_codec_tags[j].id != AV_CODEC_ID_NONE; j++) {
if (!strncmp(ff_mkv_codec_tags[j].str, track->codec_id,
@@ -3413,12 +3419,11 @@ static int matroska_parse_frame(MatroskaDemuxContext
*matroska,
uint8_t *additional, uint64_t additional_id,
int additional_size,
int64_t discard_padding)
{
-MatroskaTrackEncoding *encodings = track->encodings.elem;
uint8_t *pkt_data = data;
int res = 0;
AVPacket pktl, *pkt = &pktl;
-if (encodings && !encodings->type && encodings->scope & 1) {
+if (track->needs_decoding) {
res = matroska_decode_buffer(&pkt_data, &pkt_size, track);
if (res < 0)
return res;
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[FFmpeg-cvslog] avformat/matroskadec: Support ContentCompression for all codecs
ffmpeg | branch: master | Andreas Rheinhardt |
Fri Dec 6 09:53:34 2019 +0100| [979b5b89594c7628bd846c63198cb64ef9d81d16] |
committer: Andreas Rheinhardt
avformat/matroskadec: Support ContentCompression for all codecs
The Matroska demuxer has three functions for creating packets out of
the data read: One for certain RealAudio codecs (ATRAC3, cook, sipr,
RealAudio 28.8), one for WebVTT (actually, the WebM flavour of it) and
one for all the others. Only the last function supported Matroska's
ContentCompression (e.g. it reversed zlib compression or added the
removed headers to the packets). But in Matroska, all tracks are allowed
to be compressed. This commit adds support for this.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=979b5b89594c7628bd846c63198cb64ef9d81d16
---
libavformat/matroskadec.c | 79 ++-
1 file changed, 44 insertions(+), 35 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 9e3e98e9c0..c16e18cb22 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -3192,10 +3192,11 @@ static int matroska_parse_rm_audio(MatroskaDemuxContext
*matroska,
}
/* reconstruct full wavpack blocks from mangled matroska ones */
-static int matroska_parse_wavpack(MatroskaTrack *track, uint8_t *src,
- uint8_t **pdst, int *size)
+static int matroska_parse_wavpack(MatroskaTrack *track,
+ uint8_t **data, int *size)
{
uint8_t *dst = NULL;
+uint8_t *src = *data;
int dstlen = 0;
int srclen = *size;
uint32_t samples;
@@ -3265,7 +3266,7 @@ static int matroska_parse_wavpack(MatroskaTrack *track,
uint8_t *src,
memset(dst + dstlen, 0, AV_INPUT_BUFFER_PADDING_SIZE);
-*pdst = dst;
+*data = dst;
*size = dstlen;
return 0;
@@ -3275,8 +3276,8 @@ fail:
return ret;
}
-static int matroska_parse_prores(MatroskaTrack *track, uint8_t *src,
- uint8_t **pdst, int *size)
+static int matroska_parse_prores(MatroskaTrack *track,
+ uint8_t **data, int *size)
{
uint8_t *dst;
int dstlen = *size + 8;
@@ -3287,10 +3288,10 @@ static int matroska_parse_prores(MatroskaTrack *track,
uint8_t *src,
AV_WB32(dst, dstlen);
AV_WB32(dst + 4, MKBETAG('i', 'c', 'p', 'f'));
-memcpy(dst + 8, src, dstlen - 8);
+memcpy(dst + 8, *data, dstlen - 8);
memset(dst + dstlen, 0, AV_INPUT_BUFFER_PADDING_SIZE);
-*pdst = dst;
+*data = dst;
*size = dstlen;
return 0;
@@ -3413,54 +3414,46 @@ static int matroska_parse_webvtt(MatroskaDemuxContext
*matroska,
static int matroska_parse_frame(MatroskaDemuxContext *matroska,
MatroskaTrack *track, AVStream *st,
-AVBufferRef *buf, uint8_t *data, int pkt_size,
+AVBufferRef *buf, uint8_t **data, int pkt_size,
uint64_t timecode, uint64_t lace_duration,
int64_t pos, int is_keyframe,
uint8_t *additional, uint64_t additional_id,
int additional_size,
int64_t discard_padding)
{
-uint8_t *pkt_data = data;
+uint8_t *pkt_data = *data;
int res = 0;
AVPacket pktl, *pkt = &pktl;
-if (track->needs_decoding) {
-res = matroska_decode_buffer(&pkt_data, &pkt_size, track);
-if (res < 0)
-return res;
-}
-
if (st->codecpar->codec_id == AV_CODEC_ID_WAVPACK) {
-uint8_t *wv_data;
-res = matroska_parse_wavpack(track, pkt_data, &wv_data, &pkt_size);
+res = matroska_parse_wavpack(track, &pkt_data, &pkt_size);
if (res < 0) {
av_log(matroska->ctx, AV_LOG_ERROR,
"Error parsing a wavpack block.\n");
goto fail;
}
-if (pkt_data != data)
-av_freep(&pkt_data);
-pkt_data = wv_data;
+if (!buf)
+av_freep(data);
+buf = NULL;
}
if (st->codecpar->codec_id == AV_CODEC_ID_PRORES &&
AV_RB32(pkt_data + 4) != MKBETAG('i', 'c', 'p', 'f')) {
-uint8_t *pr_data;
-res = matroska_parse_prores(track, pkt_data, &pr_data, &pkt_size);
+res = matroska_parse_prores(track, &pkt_data, &pkt_size);
if (res < 0) {
av_log(matroska->ctx, AV_LOG_ERROR,
"Error parsing a prores block.\n");
goto fail;
}
-if (pkt_data != data)
-av_freep(&pkt_data);
-pkt_data = pr_data;
+if (!buf)
+av_freep(data);
+buf = NULL;
}
if (!pkt_size && !additional_size)
goto no_output;
av_init_packet(pkt);
-if (pkt_data != data)
+if (!buf)
pkt->buf = av_buffer_create
[FFmpeg-cvslog] avformat/matroskadec: Cosmetics
ffmpeg | branch: master | Andreas Rheinhardt |
Sat Dec 7 00:11:01 2019 +0100| [39fb1e968a5a8bd62bb454a5291d877a6ddd726c] |
committer: Andreas Rheinhardt
avformat/matroskadec: Cosmetics
Reindentation as well as marking several variables used for demuxing
RealAudio as const to clearly see that they don't change during
demuxing.
Signed-off-by: Andreas Rheinhardt
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=39fb1e968a5a8bd62bb454a5291d877a6ddd726c
---
libavformat/matroskadec.c | 24
1 file changed, 12 insertions(+), 12 deletions(-)
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index c16e18cb22..917c106258 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -3119,12 +3119,12 @@ static int matroska_parse_rm_audio(MatroskaDemuxContext
*matroska,
uint8_t *data, int size, uint64_t timecode,
int64_t pos)
{
-int a = st->codecpar->block_align;
-int sps = track->audio.sub_packet_size;
-int cfs = track->audio.coded_framesize;
-int h = track->audio.sub_packet_h;
+const int a = st->codecpar->block_align;
+const int sps = track->audio.sub_packet_size;
+const int cfs = track->audio.coded_framesize;
+const int h = track->audio.sub_packet_h;
+const int w = track->audio.frame_size;
int y = track->audio.sub_packet_cnt;
-int w = track->audio.frame_size;
int x;
if (!track->audio.pkt_cnt) {
@@ -3282,14 +3282,14 @@ static int matroska_parse_prores(MatroskaTrack *track,
uint8_t *dst;
int dstlen = *size + 8;
-dst = av_malloc(dstlen + AV_INPUT_BUFFER_PADDING_SIZE);
-if (!dst)
-return AVERROR(ENOMEM);
+dst = av_malloc(dstlen + AV_INPUT_BUFFER_PADDING_SIZE);
+if (!dst)
+return AVERROR(ENOMEM);
-AV_WB32(dst, dstlen);
-AV_WB32(dst + 4, MKBETAG('i', 'c', 'p', 'f'));
-memcpy(dst + 8, *data, dstlen - 8);
-memset(dst + dstlen, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+AV_WB32(dst, dstlen);
+AV_WB32(dst + 4, MKBETAG('i', 'c', 'p', 'f'));
+memcpy(dst + 8, *data, dstlen - 8);
+memset(dst + dstlen, 0, AV_INPUT_BUFFER_PADDING_SIZE);
*data = dst;
*size = dstlen;
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