Re: [FFmpeg-devel] [PATCH] lavformat/utils: Fix a memleak that st->codec->hw_frames_ctx
On Fri, 20 Jan 2017 10:06:50 +0800 "Huang, Zhengxu" wrote: > From 9ceb2ac6a89246f2e686eb3ad3448fbaff5328f7 Mon Sep 17 00:00:00 2001 > From: Zhengxu > Date: Fri, 13 Jan 2017 10:33:05 +0800 > Subject: [PATCH] lavformat/utils: Fix a memleak that st->codec->hw_frames_ctx > is not released. > > Signed-off-by: ChaoX A Liu > Signed-off-by: Huang, Zhengxu > Signed-off-by: Andrew, Zhang > --- > libavformat/utils.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/libavformat/utils.c b/libavformat/utils.c > index d5dfca7..cadec15 100644 > --- a/libavformat/utils.c > +++ b/libavformat/utils.c > @@ -4127,6 +4127,7 @@ static void free_stream(AVStream **pst) > FF_DISABLE_DEPRECATION_WARNINGS > av_freep(&st->codec->extradata); > av_freep(&st->codec->subtitle_header); > +av_buffer_unref(&st->codec->hw_frames_ctx); > av_freep(&st->codec); > FF_ENABLE_DEPRECATION_WARNINGS > #endif What triggers this? In a sane world this should never ever happen. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] dcaenc: use Huffman codes for Bit Allocation Index
Hi. My previous patch introduced Huffman for quantized data, this patch allows to use Huffman for bit allocation indexes (abits). This is a less significant improvement but still noticeable in case of limited bitrate. In case of DIN_45403 input sample I got: before: Best PSNR is 44.47 for shift 0 after: Best PSNR is 46.64 for shift 0 Thank you. -- Daniil Cherednik 0001-dcaenc-Use-Huffman-codes-for-Bit-Allocation-Index.patch Description: Binary data ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] lavformat/utils: Fix a memleak that st->codec->hw_frames_ctx
Have you ever used valgrind? Please just run the command below:
valgrind --leak-check=full --log-file=out.log ffmpeg -hwaccel qsv
-qsv_device /dev/dri/renderD128 -c:v h264_qsv -i a.h264 -c:v h264_qsv -b:v
2M -y out.h264
See line 3323 of ffmpeg.c,
ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
and see what have been done in avcodec_copy_context:
if (src->hw_frames_ctx) {
dest->hw_frames_ctx = av_buffer_ref(src->hw_frames_ctx);
if (!dest->hw_frames_ctx)
goto fail;
}
However, that is not freed when calling avformat_free_context.
On Fri, Jan 20, 2017 at 4:47 PM, wm4 wrote:
> On Fri, 20 Jan 2017 10:06:50 +0800
> "Huang, Zhengxu" wrote:
>
> > From 9ceb2ac6a89246f2e686eb3ad3448fbaff5328f7 Mon Sep 17 00:00:00 2001
> > From: Zhengxu
> > Date: Fri, 13 Jan 2017 10:33:05 +0800
> > Subject: [PATCH] lavformat/utils: Fix a memleak that
> st->codec->hw_frames_ctx
> > is not released.
> >
> > Signed-off-by: ChaoX A Liu
> > Signed-off-by: Huang, Zhengxu
> > Signed-off-by: Andrew, Zhang
> > ---
> > libavformat/utils.c | 1 +
> > 1 file changed, 1 insertion(+)
> >
> > diff --git a/libavformat/utils.c b/libavformat/utils.c
> > index d5dfca7..cadec15 100644
> > --- a/libavformat/utils.c
> > +++ b/libavformat/utils.c
> > @@ -4127,6 +4127,7 @@ static void free_stream(AVStream **pst)
> > FF_DISABLE_DEPRECATION_WARNINGS
> > av_freep(&st->codec->extradata);
> > av_freep(&st->codec->subtitle_header);
> > +av_buffer_unref(&st->codec->hw_frames_ctx);
> > av_freep(&st->codec);
> > FF_ENABLE_DEPRECATION_WARNINGS
> > #endif
>
> What triggers this? In a sane world this should never ever happen.
> ___
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> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
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[FFmpeg-devel] [PATCH] ffmpeg_qsv.c: Init an hwframes_context for decoder instead of encoder
From 2149f87637ab941be14828f7ae2c224908784c7d Mon Sep 17 00:00:00 2001
From: Zhengxu
Date: Wed, 4 Jan 2017 16:43:43 +0800
Subject: [PATCH] ffmpeg_qsv.c: Init an hwframes_context for decoder instead of
encoder.
We consider that encoder is the last stage in the pipeline. Thinking
about filters, they get hwframes_context from their inputs. Likewise,
encoder should get hwframes_context from its input instead creating a
new faker one. Encoder can get acuurate parameters by doing so.
Signed-off-by: ChaoX A Liu
Signed-off-by: Huang, Zhengxu
Signed-off-by: Andrew, Zhang
---
ffmpeg_qsv.c | 95 +++-
1 file changed, 43 insertions(+), 52 deletions(-)
diff --git a/ffmpeg_qsv.c b/ffmpeg_qsv.c
index 86824b6..8cedb7e 100644
--- a/ffmpeg_qsv.c
+++ b/ffmpeg_qsv.c
@@ -81,25 +81,26 @@ int qsv_init(AVCodecContext *s)
return ret;
}
-av_buffer_unref(&ist->hw_frames_ctx);
-ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
-if (!ist->hw_frames_ctx)
-return AVERROR(ENOMEM);
-
-frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
-frames_hwctx = frames_ctx->hwctx;
-
-frames_ctx->width = FFALIGN(s->coded_width, 32);
-frames_ctx->height= FFALIGN(s->coded_height, 32);
-frames_ctx->format= AV_PIX_FMT_QSV;
-frames_ctx->sw_format = s->sw_pix_fmt;
-frames_ctx->initial_pool_size = 64;
-frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
-
-ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
-if (ret < 0) {
-av_log(NULL, AV_LOG_ERROR, "Error initializing a QSV frame pool\n");
-return ret;
+if(!ist->hw_frames_ctx) {
+ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
+if (!ist->hw_frames_ctx)
+return AVERROR(ENOMEM);
+
+frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
+frames_hwctx = frames_ctx->hwctx;
+
+frames_ctx->width = FFALIGN(s->coded_width, 32);
+frames_ctx->height= FFALIGN(s->coded_height, 32);
+frames_ctx->format= AV_PIX_FMT_QSV;
+frames_ctx->sw_format = s->sw_pix_fmt;
+frames_ctx->initial_pool_size = 64;
+frames_hwctx->frame_type =
MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
+
+ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
+if (ret < 0) {
+av_log(NULL, AV_LOG_ERROR, "Error initializing a QSV frame
pool\n");
+return ret;
+}
}
ist->hwaccel_get_buffer = qsv_get_buffer;
@@ -114,9 +115,8 @@ int qsv_transcode_init(OutputStream *ost)
const enum AVPixelFormat *pix_fmt;
int err, i;
-AVBufferRef *encode_frames_ref = NULL;
-AVHWFramesContext *encode_frames;
-AVQSVFramesContext *qsv_frames;
+AVHWFramesContext *frames_ctx;
+AVQSVFramesContext *frames_hwctx;
/* check if the encoder supports QSV */
if (!ost->enc->pix_fmts)
@@ -150,42 +150,33 @@ int qsv_transcode_init(OutputStream *ost)
if (!hw_device_ctx) {
err = qsv_device_init(ist);
if (err < 0)
-goto fail;
-}
-
-// This creates a dummy hw_frames_ctx for the encoder to be
-// suitably initialised. It only contains one real frame, so
-// hopefully doesn't waste too much memory.
-
-encode_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx);
-if (!encode_frames_ref) {
-err = AVERROR(ENOMEM);
-goto fail;
+return err;
}
-encode_frames = (AVHWFramesContext*)encode_frames_ref->data;
-qsv_frames = encode_frames->hwctx;
-encode_frames->width = FFALIGN(ist->resample_width, 32);
-encode_frames->height= FFALIGN(ist->resample_height, 32);
-encode_frames->format= AV_PIX_FMT_QSV;
-encode_frames->sw_format = AV_PIX_FMT_NV12;
-encode_frames->initial_pool_size = 1;
+ist->dec_ctx->pix_fmt = AV_PIX_FMT_QSV;
+ist->resample_pix_fmt = AV_PIX_FMT_QSV;
+ist->hw_frames_ctx= av_hwframe_ctx_alloc(hw_device_ctx);
+if (!ist->hw_frames_ctx)
+return AVERROR(ENOMEM);
-qsv_frames->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
+frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
+frames_hwctx = frames_ctx->hwctx;
-err = av_hwframe_ctx_init(encode_frames_ref);
-if (err < 0)
-goto fail;
+frames_ctx->width = FFALIGN(ist->resample_width, 32);
+frames_ctx->height= FFALIGN(ist->resample_height, 32);
+frames_ctx->format= AV_PIX_FMT_QSV;
+frames_ctx->sw_format = AV_PIX_FMT_NV12;
+frames_ctx->initial_pool_size = 64;
+frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
-ist->dec_ctx->pix_fmt = AV_PIX_FMT_QSV;
-ist->resample_pix_fmt = AV_PIX_FMT_QSV;
+err = av_hwframe_ctx_init(ist->hw_frames_ctx);
+if (err < 0) {
+
Re: [FFmpeg-devel] [PATCH] libavcodec/nvenc.c Reduce initialization time for gpu id > 0
> Yes, this is a simpler logic. Works for me locally. Can you please apply this > patch to ffmpeg master. Thanks. Applied We also discovered a case where this patch is necessary to successfully use multiple GPUs at all. A user asked in #ffmpeg about nvenc getting stuck when one GPU is at full load, even when ffmpeg was instructed to use a different GPU. Turned out that when GPU #0 is under full video engine load, initializing CUDA/NVENC on it just hangs indefinitely until the load drops. With the old logic this meant then when trying to use GPU #1, ffmpeg would still initialize GPU #0 to test its capabilities and get stuck while doing so. Not sure if this is intended behavior, I'd expect CUDA/NVENC to just fail when no resources are left instead of blocking indefinitely on init. The primary problem of being unable to use GPU #1 has been fixed with this patch though. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] lavformat/utils: Fix a memleak that st->codec->hw_frames_ctx
On Fri, Jan 20, 2017 at 8:35 PM, Chao Liu wrote:
> Have you ever used valgrind? Please just run the command below:
> valgrind --leak-check=full --log-file=out.log ffmpeg -hwaccel qsv
> -qsv_device /dev/dri/renderD128 -c:v h264_qsv -i a.h264 -c:v h264_qsv -b:v
> 2M -y out.h264
>
> See line 3323 of ffmpeg.c,
> ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
> and see what have been done in avcodec_copy_context:
> if (src->hw_frames_ctx) {
> dest->hw_frames_ctx = av_buffer_ref(src->hw_frames_ctx);
> if (!dest->hw_frames_ctx)
> goto fail;
> }
> However, that is not freed when calling avformat_free_context.
>
avcodec_copy_context is deprecated and should generally not be used
anymore. It would be more appropriate to resolve whatever issue
remains in ffmpeg that it needs to call it.
Otherwise, hw_frames_ctx is supplied by the caller to libavcodec, so
it also should manage its lifetime, I would think, and not be blindly
free'ed by avcodec.
PS:
Please don't top post.
- Hendrik
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Re: [FFmpeg-devel] [PATCH] lavformat/utils: Fix a memleak that st->codec->hw_frames_ctx
On Fri, 20 Jan 2017 17:35:33 +0800
Chao Liu wrote:
> Have you ever used valgrind? Please just run the command below:
> valgrind --leak-check=full --log-file=out.log ffmpeg -hwaccel qsv
> -qsv_device /dev/dri/renderD128 -c:v h264_qsv -i a.h264 -c:v h264_qsv -b:v
> 2M -y out.h264
>
> See line 3323 of ffmpeg.c,
> ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
> and see what have been done in avcodec_copy_context:
> if (src->hw_frames_ctx) {
> dest->hw_frames_ctx = av_buffer_ref(src->hw_frames_ctx);
> if (!dest->hw_frames_ctx)
> goto fail;
> }
> However, that is not freed when calling avformat_free_context.
>
The hardware decoder should never be created in the demuxer.
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Re: [FFmpeg-devel] [PATCH] ffmpeg_qsv.c: Init an hwframes_context for decoder instead of encoder
On Fri, 20 Jan 2017 17:41:01 +0800
"Huang, Zhengxu" wrote:
> From 2149f87637ab941be14828f7ae2c224908784c7d Mon Sep 17 00:00:00 2001
> From: Zhengxu
> Date: Wed, 4 Jan 2017 16:43:43 +0800
> Subject: [PATCH] ffmpeg_qsv.c: Init an hwframes_context for decoder instead of
> encoder.
>
> We consider that encoder is the last stage in the pipeline. Thinking
> about filters, they get hwframes_context from their inputs. Likewise,
> encoder should get hwframes_context from its input instead creating a
> new faker one. Encoder can get acuurate parameters by doing so.
the idea is right. What ffmpeg.c does is a gross hack that avconv
doesn't have or need. Ideally, ffmpeg_qsv.c would look like avconv's
equivalent file:
https://git.libav.org/?p=libav.git;a=blob;f=avconv_qsv.c;hb=HEAD
(It's quite possible that this still lacks avconv changes needed for
this, and which were either skipped during merge, or not merged yet.)
> Signed-off-by: ChaoX A Liu
> Signed-off-by: Huang, Zhengxu
> Signed-off-by: Andrew, Zhang
> ---
> ffmpeg_qsv.c | 95
> +++-
> 1 file changed, 43 insertions(+), 52 deletions(-)
>
> diff --git a/ffmpeg_qsv.c b/ffmpeg_qsv.c
> index 86824b6..8cedb7e 100644
> --- a/ffmpeg_qsv.c
> +++ b/ffmpeg_qsv.c
> @@ -81,25 +81,26 @@ int qsv_init(AVCodecContext *s)
> return ret;
> }
>
> -av_buffer_unref(&ist->hw_frames_ctx);
> -ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
> -if (!ist->hw_frames_ctx)
> -return AVERROR(ENOMEM);
> -
> -frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
> -frames_hwctx = frames_ctx->hwctx;
> -
> -frames_ctx->width = FFALIGN(s->coded_width, 32);
> -frames_ctx->height= FFALIGN(s->coded_height, 32);
> -frames_ctx->format= AV_PIX_FMT_QSV;
> -frames_ctx->sw_format = s->sw_pix_fmt;
> -frames_ctx->initial_pool_size = 64;
> -frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
> -
> -ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
> -if (ret < 0) {
> -av_log(NULL, AV_LOG_ERROR, "Error initializing a QSV frame pool\n");
> -return ret;
> +if(!ist->hw_frames_ctx) {
Why this check? It's pointless - hw_frames_ctx is unset when get_format
is called.
It also changes indentation, which makes reviewing harder.
> +ist->hw_frames_ctx = av_hwframe_ctx_alloc(hw_device_ctx);
> +if (!ist->hw_frames_ctx)
> +return AVERROR(ENOMEM);
> +
> +frames_ctx = (AVHWFramesContext*)ist->hw_frames_ctx->data;
> +frames_hwctx = frames_ctx->hwctx;
> +
> +frames_ctx->width = FFALIGN(s->coded_width, 32);
> +frames_ctx->height= FFALIGN(s->coded_height, 32);
> +frames_ctx->format= AV_PIX_FMT_QSV;
> +frames_ctx->sw_format = s->sw_pix_fmt;
> +frames_ctx->initial_pool_size = 64;
> +frames_hwctx->frame_type =
> MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
> +
> +ret = av_hwframe_ctx_init(ist->hw_frames_ctx);
> +if (ret < 0) {
> +av_log(NULL, AV_LOG_ERROR, "Error initializing a QSV frame
> pool\n");
> +return ret;
> +}
> }
>
> ist->hwaccel_get_buffer = qsv_get_buffer;
> @@ -114,9 +115,8 @@ int qsv_transcode_init(OutputStream *ost)
> const enum AVPixelFormat *pix_fmt;
>
> int err, i;
> -AVBufferRef *encode_frames_ref = NULL;
> -AVHWFramesContext *encode_frames;
> -AVQSVFramesContext *qsv_frames;
> +AVHWFramesContext *frames_ctx;
> +AVQSVFramesContext *frames_hwctx;
>
> /* check if the encoder supports QSV */
> if (!ost->enc->pix_fmts)
> @@ -150,42 +150,33 @@ int qsv_transcode_init(OutputStream *ost)
> if (!hw_device_ctx) {
> err = qsv_device_init(ist);
> if (err < 0)
> -goto fail;
> -}
> -
> -// This creates a dummy hw_frames_ctx for the encoder to be
> -// suitably initialised. It only contains one real frame, so
> -// hopefully doesn't waste too much memory.
> -
> -encode_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx);
> -if (!encode_frames_ref) {
> -err = AVERROR(ENOMEM);
> -goto fail;
> +return err;
> }
> -encode_frames = (AVHWFramesContext*)encode_frames_ref->data;
> -qsv_frames = encode_frames->hwctx;
>
> -encode_frames->width = FFALIGN(ist->resample_width, 32);
> -encode_frames->height= FFALIGN(ist->resample_height, 32);
> -encode_frames->format= AV_PIX_FMT_QSV;
> -encode_frames->sw_format = AV_PIX_FMT_NV12;
> -encode_frames->initial_pool_size = 1;
> +ist->dec_ctx->pix_fmt = AV_PIX_FMT_QSV;
> +ist->resample_pix_fmt = AV_PIX_FMT_QSV;
These lines should be removed. AFAIK they're hacks to make ffmpeg.c
"behave". Please fix the root of the issue instead.
> +ist->hw_frames_ctx= av_hwframe_c
Re: [FFmpeg-devel] dcaenc: use Huffman codes for Bit Allocation Index
On 20 January 2017 at 08:57, Даниил Чередник wrote: > Hi. > > My previous patch introduced Huffman for quantized data, this patch allows > to use Huffman for bit allocation indexes (abits). This is a less > significant improvement but still noticeable in case of limited bitrate. In > case of DIN_45403 input sample I got: > before: > Best PSNR is 44.47 for shift 0 > after: > Best PSNR is 46.64 for shift 0 > > Thank you. > -- > Daniil Cherednik > > ___ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > Hi, I've reviewed and tested the patch, saw nothing wrong. There was indeed a small improvement from signalling bit allocation using Huffman tables. Performance didn't even decrease on my machine. Thanks, pushed. ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] avcodec/atrac3: allow 6 channels (non-joint stereo)
On 1/19/17, bananaman...@gmail.com wrote: > From: bnnm > > Raises max channels to 6 (for non joint-stereo only), > there is no difference decoding 1 or N discrete channels. > Fixes trac issue #5840 > > Signed-off-by: bnnm > --- > libavcodec/atrac3.c | 17 - > 1 file changed, 8 insertions(+), 9 deletions(-) > applied ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH 2/3] avformat/avienc: add reserve_index_space option
On 19.01.2017 18:32, Michael Niedermayer wrote:
On Wed, Jan 18, 2017 at 10:27:02AM +0100, Tobias Rapp wrote:
Allows the user to reserve space for the ODML master index. A sufficient
sized master index in the AVI header avoids storing follow-up master
indexes within the 'movi' data later.
If the option is omitted or zero the index size is estimated from output
duration and bitrate. A worst-case bitrate for video streams is assumed
in case it is not available.
Note: fate reference files changed because the video stream had zero
bitrate before and is guessed now.
Signed-off-by: Tobias Rapp
---
libavformat/avi.h | 1 -
libavformat/avienc.c| 77 ++---
libavformat/version.h | 2 +-
tests/ref/fate/mpeg4-bsf-unpack-bframes | 2 +-
tests/ref/lavf-fate/avi_cram| 2 +-
5 files changed, 74 insertions(+), 10 deletions(-)
this breaks segment:
./ffmpeg -i lena.pnm -f segment test%d.avi
possibly related to avi_init()
Yes, I can reproduce the problem when going back to Git master and just
adding a dummy init (see attached diff). Not sure how to fix this, other
muxers also have an init but seem to work fine (mkv) ...
Of course I could simply work-around it by moving the code from avi_init
into avi_write_header.
Regards,
Tobias
diff --git a/libavformat/avienc.c b/libavformat/avienc.c
index fd16fff..fb78300 100644
--- a/libavformat/avienc.c
+++ b/libavformat/avienc.c
@@ -134,6 +134,12 @@ static int avi_add_ientry(AVFormatContext *s, int
stream_index, char *tag,
return 0;
}
+
+static int avi_init(struct AVFormatContext *s)
+{
+return 0;
+}
+
static int64_t avi_start_new_riff(AVFormatContext *s, AVIOContext *pb,
const char *riff_tag, const char *list_tag)
{
@@ -927,6 +933,7 @@ AVOutputFormat ff_avi_muxer = {
.priv_data_size = sizeof(AVIContext),
.audio_codec= CONFIG_LIBMP3LAME ? AV_CODEC_ID_MP3 : AV_CODEC_ID_AC3,
.video_codec= AV_CODEC_ID_MPEG4,
+.init = avi_init,
.write_header = avi_write_header,
.write_packet = avi_write_packet,
.write_trailer = avi_write_trailer,
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Re: [FFmpeg-devel] [PATCH 2/3] avformat/avienc: add reserve_index_space option
On 20.01.2017 15:56, Tobias Rapp wrote:
On 19.01.2017 18:32, Michael Niedermayer wrote:
On Wed, Jan 18, 2017 at 10:27:02AM +0100, Tobias Rapp wrote:
Allows the user to reserve space for the ODML master index. A sufficient
sized master index in the AVI header avoids storing follow-up master
indexes within the 'movi' data later.
If the option is omitted or zero the index size is estimated from output
duration and bitrate. A worst-case bitrate for video streams is assumed
in case it is not available.
Note: fate reference files changed because the video stream had zero
bitrate before and is guessed now.
Signed-off-by: Tobias Rapp
---
libavformat/avi.h | 1 -
libavformat/avienc.c| 77
++---
libavformat/version.h | 2 +-
tests/ref/fate/mpeg4-bsf-unpack-bframes | 2 +-
tests/ref/lavf-fate/avi_cram| 2 +-
5 files changed, 74 insertions(+), 10 deletions(-)
this breaks segment:
./ffmpeg -i lena.pnm -f segment test%d.avi
possibly related to avi_init()
Yes, I can reproduce the problem when going back to Git master and just
adding a dummy init (see attached diff). Not sure how to fix this, other
muxers also have an init but seem to work fine (mkv) ...
Apparently the codec_tag is cleared in seg_write_header() around line 811:
[...]
if (!oc->oformat->codec_tag ||
av_codec_get_id (oc->oformat->codec_tag, ipar->codec_tag) ==
opar->codec_id ||
av_codec_get_tag(oc->oformat->codec_tag, ipar->codec_id) <= 0) {
opar->codec_tag = ipar->codec_tag;
} else {
opar->codec_tag = 0;
}
[...]
See
http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/segment.c;hb=HEAD#l811
If .init is unset, the header is written before the codec_tag is
cleared. If .init is set, it is written after these lines.
Regards,
Tobias
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Re: [FFmpeg-devel] [PATCH 2/3] avformat/avienc: add reserve_index_space option
On Fri, Jan 20, 2017 at 05:06:59PM +0100, Tobias Rapp wrote:
> On 20.01.2017 15:56, Tobias Rapp wrote:
> >On 19.01.2017 18:32, Michael Niedermayer wrote:
> >>On Wed, Jan 18, 2017 at 10:27:02AM +0100, Tobias Rapp wrote:
> >>>Allows the user to reserve space for the ODML master index. A sufficient
> >>>sized master index in the AVI header avoids storing follow-up master
> >>>indexes within the 'movi' data later.
> >>>
> >>>If the option is omitted or zero the index size is estimated from output
> >>>duration and bitrate. A worst-case bitrate for video streams is assumed
> >>>in case it is not available.
> >>>
> >>>Note: fate reference files changed because the video stream had zero
> >>>bitrate before and is guessed now.
> >>>
> >>>Signed-off-by: Tobias Rapp
> >>>---
> >>> libavformat/avi.h | 1 -
> >>> libavformat/avienc.c| 77
> >>>++---
> >>> libavformat/version.h | 2 +-
> >>> tests/ref/fate/mpeg4-bsf-unpack-bframes | 2 +-
> >>> tests/ref/lavf-fate/avi_cram| 2 +-
> >>> 5 files changed, 74 insertions(+), 10 deletions(-)
> >>
> >>this breaks segment:
> >>./ffmpeg -i lena.pnm -f segment test%d.avi
> >>
> >>possibly related to avi_init()
> >
> >Yes, I can reproduce the problem when going back to Git master and just
> >adding a dummy init (see attached diff). Not sure how to fix this, other
> >muxers also have an init but seem to work fine (mkv) ...
>
> Apparently the codec_tag is cleared in seg_write_header() around line 811:
>
> [...]
> if (!oc->oformat->codec_tag ||
> av_codec_get_id (oc->oformat->codec_tag, ipar->codec_tag) ==
> opar->codec_id ||
> av_codec_get_tag(oc->oformat->codec_tag, ipar->codec_id) <= 0) {
> opar->codec_tag = ipar->codec_tag;
> } else {
> opar->codec_tag = 0;
> }
> [...]
>
> See
> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/segment.c;hb=HEAD#l811
>
> If .init is unset, the header is written before the codec_tag is
> cleared. If .init is set, it is written after these lines.
the return code of your init function is wrong i think
also this patch causes a 1 byte difference in the output of:
./ffmpeg -i ~/tickets/1116/AVI-RLE_MP3.avi -vcodec copy -an out2.avi
is that intended ?
[...]
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[FFmpeg-devel] [PATCH] avformat: add MIDI Sample Dump Standard demuxer
Signed-off-by: Paul B Mahol
---
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/sdsdec.c | 165 +++
3 files changed, 167 insertions(+)
create mode 100644 libavformat/sdsdec.c
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 872b30e..fbb28e3 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -435,6 +435,7 @@ OBJS-$(CONFIG_SAP_DEMUXER) += sapdec.o
OBJS-$(CONFIG_SAP_MUXER) += sapenc.o
OBJS-$(CONFIG_SBG_DEMUXER) += sbgdec.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
+OBJS-$(CONFIG_SDS_DEMUXER) += sdsdec.o
OBJS-$(CONFIG_SDR2_DEMUXER) += sdr2.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
OBJS-$(CONFIG_SEGMENT_MUXER) += segment.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 6a79b75..8787c79 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -274,6 +274,7 @@ void av_register_all(void)
REGISTER_MUXDEMUX(SAP, sap);
REGISTER_DEMUXER (SBG, sbg);
REGISTER_DEMUXER (SDP, sdp);
+REGISTER_DEMUXER (SDS, sds);
REGISTER_DEMUXER (SDR2, sdr2);
#if CONFIG_RTPDEC
ff_register_rtp_dynamic_payload_handlers();
diff --git a/libavformat/sdsdec.c b/libavformat/sdsdec.c
new file mode 100644
index 000..727586a
--- /dev/null
+++ b/libavformat/sdsdec.c
@@ -0,0 +1,165 @@
+/*
+ * MIDI Sample Dump Standard format demuxer
+ * Copyright (c) 2017 Paul B Mahol
+ *
+ * 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/intreadwrite.h"
+#include "avformat.h"
+#include "internal.h"
+
+typedef struct SDSContext {
+uint8_t data[120];
+int bit_depth;
+int size;
+void (*read_block)(const uint8_t *src, int32_t *dst);
+} SDSContext;
+
+static int sds_probe(AVProbeData *p)
+{
+if (AV_RB32(p->buf) == 0xF07E0001 && p->buf[20] == 0xF7 &&
+p->buf[6] >= 8 && p->buf[6] <= 28)
+return AVPROBE_SCORE_EXTENSION;
+return 0;
+}
+
+static void byte2_read(const uint8_t *src, int32_t *dst)
+{
+int i;
+
+for (i = 0; i < 120; i += 2) {
+unsigned sample = (src[i + 0] << 25) + (src[i + 1] << 18);
+
+dst[i / 2] = sample;
+}
+}
+
+static void byte3_read(const uint8_t *src, int32_t *dst)
+{
+int i;
+
+for (i = 0; i < 120; i += 3) {
+unsigned sample;
+
+ sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11);
+dst[i / 3] = sample;
+}
+}
+
+static void byte4_read(const uint8_t *src, int32_t *dst)
+{
+int i;
+
+for (i = 0; i < 120; i += 4) {
+unsigned sample;
+
+ sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11) |
(src[i + 3] << 4);
+dst[i / 4] = sample;
+}
+}
+
+#define SDS_3BYTE_TO_INT_DECODE(x) (((x) & 0x7F) | (((x) & 0x7F00) >> 1) |
(((x) & 0x7F) >> 2))
+
+static int sds_read_header(AVFormatContext *ctx)
+{
+SDSContext *s = ctx->priv_data;
+unsigned sample_period;
+AVIOContext *pb = ctx->pb;
+AVStream *st;
+
+st = avformat_new_stream(ctx, NULL);
+if (!st)
+return AVERROR(ENOMEM);
+
+avio_skip(pb, 4);
+avio_skip(pb, 2);
+
+s->bit_depth = avio_r8(pb);
+if (s->bit_depth < 8 || s->bit_depth > 28)
+return AVERROR_INVALIDDATA;
+
+if (s->bit_depth < 14) {
+s->read_block = byte2_read;
+s->size = 60 * 4;
+} else if (s->bit_depth < 21) {
+s->read_block = byte3_read;
+s->size = 40 * 4;
+} else {
+s->read_block = byte4_read;
+s->size = 30 * 4;
+}
+st->codecpar->codec_id = AV_CODEC_ID_PCM_U32LE;
+
+sample_period = avio_rl24(pb);
+sample_period = SDS_3BYTE_TO_INT_DECODE(sample_period);
+avio_skip(pb, 11);
+
+st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+st->codecpar->channels = 1;
+st->codecpar->sample_rate = sample_period ? 10 / sample_period :
16000;
+st->duration = (avio_size(pb) - 21) / (127) * s->size / 4;
+
+avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
+
+return 0;
+}
+
+static int sds_read_packet(AVFormatContext *ctx, AVPacket *pkt)
+{
+SD
Re: [FFmpeg-devel] [PATCH] ffmpeg.c: Add output file index and stream index to vstats file.
On Thu, Jan 19, 2017 at 02:34:14PM -0800, Sasi Inguva wrote:
> Signed-off-by: Sasi Inguva
> ---
> ffmpeg.c | 2 +-
> 1 file changed, 1 insertion(+), 1 deletion(-)
>
> diff --git a/ffmpeg.c b/ffmpeg.c
> index 977708c069..a566c3fe2a 100644
> --- a/ffmpeg.c
> +++ b/ffmpeg.c
> @@ -1347,7 +1347,7 @@ static void do_video_stats(OutputStream *ost, int
> frame_size)
> enc = ost->enc_ctx;
> if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
> frame_number = ost->st->nb_frames;
> -fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number,
> +fprintf(vstats_file, "out= %2d st= %2d frame= %5d q= %2.1f ",
> ost->file_index, ost->index, frame_number,
> ost->quality / (float)FF_QP2LAMBDA);
Should this be under a check for vstats version ?
So as not to break any code using the vstats file or is there some
declaration in our docs that the format may change at any time ?
[...]
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Re: [FFmpeg-devel] [PATCH] avcodec: Add FF_CODEC_CAP_SKIP_FRAME_FILL_PARAM to most h263 based codecs
On Thu, Dec 22, 2016 at 05:50:13PM +0100, Michael Niedermayer wrote: > Signed-off-by: Michael Niedermayer > --- > libavcodec/flvdec.c| 1 + > libavcodec/h263dec.c | 2 ++ > libavcodec/intelh263dec.c | 1 + > libavcodec/mpeg4videodec.c | 1 + > libavcodec/msmpeg4dec.c| 4 > 5 files changed, 9 insertions(+) applied [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB In fact, the RIAA has been known to suggest that students drop out of college or go to community college in order to be able to afford settlements. -- The RIAA signature.asc Description: Digital signature ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
[FFmpeg-devel] [PATCH] lavf/segment: fix crash when failing to open segment list
This happens because segment_end() returns an error, so seg_write_packet never proceeds to segment_start(), and seg->avf->pb is never re-set, so we crash with a null pb when av_write_trailer flushes the packet queue. This doesn't seem to be clearly recoverable, so I'm just failing more gracefully. Repro: ffmpeg -i input.ts -f segment -c copy -segment_list /noaxx.m3u8 test-%05d.ts (assuming you don't have write access to /) --- libavformat/segment.c | 5 - 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/libavformat/segment.c b/libavformat/segment.c index 9b3dc178eb..9d471483b3 100644 --- a/libavformat/segment.c +++ b/libavformat/segment.c @@ -354,6 +354,9 @@ static int segment_end(AVFormatContext *s, int write_trailer, int is_last) int i; int err; +if (!oc || !oc->pb) +return AVERROR(EINVAL); + av_write_frame(oc, NULL); /* Flush any buffered data (fragmented mp4) */ if (write_trailer) ret = av_write_trailer(oc); @@ -850,7 +853,7 @@ static int seg_write_packet(AVFormatContext *s, AVPacket *pkt) int64_t usecs; int64_t wrapped_val; -if (!seg->avf) +if (!seg->avf || !seg->avf->pb) return AVERROR(EINVAL); calc_times: -- 2.11.0 ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
Re: [FFmpeg-devel] [PATCH] ffmpeg.c: Add output file index and stream index to vstats file.
Couldn't find any version that relates to vstats. There is nothing that
says format may change any time , but there is no defined format either.
Let me know the version enum I have to update if I need to.
On Fri, Jan 20, 2017 at 4:18 PM, Michael Niedermayer
wrote:
> On Thu, Jan 19, 2017 at 02:34:14PM -0800, Sasi Inguva wrote:
> > Signed-off-by: Sasi Inguva
> > ---
> > ffmpeg.c | 2 +-
> > 1 file changed, 1 insertion(+), 1 deletion(-)
> >
> > diff --git a/ffmpeg.c b/ffmpeg.c
> > index 977708c069..a566c3fe2a 100644
> > --- a/ffmpeg.c
> > +++ b/ffmpeg.c
> > @@ -1347,7 +1347,7 @@ static void do_video_stats(OutputStream *ost, int
> frame_size)
> > enc = ost->enc_ctx;
> > if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
> > frame_number = ost->st->nb_frames;
> > -fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number,
> > +fprintf(vstats_file, "out= %2d st= %2d frame= %5d q= %2.1f ",
> ost->file_index, ost->index, frame_number,
> > ost->quality / (float)FF_QP2LAMBDA);
>
> Should this be under a check for vstats version ?
> So as not to break any code using the vstats file or is there some
> declaration in our docs that the format may change at any time ?
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> In a rich man's house there is no place to spit but his face.
> -- Diogenes of Sinope
>
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[FFmpeg-devel] [PATCH] avcodec/alsdec.c: testing MPEG-4 ALS decoder with floating point audio data
If als_07_2ch192k32bF.mp4 isn't already located in fate-suite/lossless-audio/, you can download at: http://standards.iso.org/ittf/PubliclyAvailableStandards/ISO_IEC_14496-26_2010_Bitstreams/DVD1/mpeg4audio-conformance/compressedMp4/als_07_2ch192k32bF.mp4 Reference file can be found at: https://dl.dropboxusercontent.com/u/1519724/als_07_2ch192k32bF.f32 Signed-off-by: Thomas Turner --- tests/fate/als.mak | 4 1 file changed, 4 insertions(+) diff --git a/tests/fate/als.mak b/tests/fate/als.mak index ff2badf..1f4c88f 100644 --- a/tests/fate/als.mak +++ b/tests/fate/als.mak @@ -7,5 +7,9 @@ endef $(foreach N,$(ALS_SUITE),$(eval $(call FATE_ALS_SUITE,$(N +FATE_ALS += fate-mpeg4-als-conformance-07 +fate-mpeg4-als-conformance-07: CMD = ffmpeg -i $(TARGET_SAMPLES)/lossless-audio/als_07_2ch192k32bF.mp4 -af atrim=end_sample=958446 -f f32le - +fate-mpeg4-als-conformance-07: REF = $(TARGET_SAMPLES)/audio-reference/als_07_2ch192k32bF.f32 + FATE_SAMPLES_AVCONV-$(call DEMDEC, MOV, ALS) += $(FATE_ALS) fate-als: $(FATE_ALS) -- 1.9.1 ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
