pacho 15/05/30 15:35:59
Added: 0.10.13_p201211-snow-codec.patch
0.10.13_p201211-r_frame_rate.patch
0.10.13_p201211-audioresample.patch
Log:
Support libav-11 too (#509326 by Nikoli, mudler and others).
(Portage version: 2.2.20/cvs/Linux x86_64, signed Manifest commit with key
A188FBD4)
Revision Changes Path
1.1
media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch
file :
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch?rev=1.1&view=markup
plain:
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch?rev=1.1&content-type=text/plain
Index: 0.10.13_p201211-snow-codec.patch
===================================================================
From: Ettore Di Giacinto <[email protected]>
--- ext/ffmpeg/gstffmpegcodecmap.c~ 2015-05-29 10:50:06.207840323 +0200
+++ ext/ffmpeg/gstffmpegcodecmap.c 2015-05-29 10:50:25.638042896 +0200
@@ -1307,7 +1307,6 @@
case AV_CODEC_ID_FLIC:
case AV_CODEC_ID_VMDVIDEO:
case AV_CODEC_ID_VMDAUDIO:
- case AV_CODEC_ID_SNOW:
case AV_CODEC_ID_VIXL:
case AV_CODEC_ID_QPEG:
case AV_CODEC_ID_PGMYUV:
1.1
media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch
file :
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch?rev=1.1&view=markup
plain:
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch?rev=1.1&content-type=text/plain
Index: 0.10.13_p201211-r_frame_rate.patch
===================================================================
From: Ettore Di Giacinto <[email protected]>
--- ext/ffmpeg/gstffmpegdemux.c~ 2015-05-29 00:52:07.601606544 +0200
+++ ext/ffmpeg/gstffmpegdemux.c 2015-05-29 01:18:18.533113323 +0200
@@ -781,8 +781,8 @@
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, GST_FORMAT_DEFAULT,
- gst_util_uint64_scale (timeposition, avstream->r_frame_rate.num,
- GST_SECOND * avstream->r_frame_rate.den));
+ gst_util_uint64_scale (timeposition,
avstream->avg_frame_rate.num,
+ GST_SECOND * avstream->avg_frame_rate.den));
res = TRUE;
break;
case GST_FORMAT_BYTES:
@@ -818,8 +818,8 @@
break;
case GST_FORMAT_DEFAULT:
gst_query_set_duration (query, GST_FORMAT_DEFAULT,
- gst_util_uint64_scale (timeduration, avstream->r_frame_rate.num,
- GST_SECOND * avstream->r_frame_rate.den));
+ gst_util_uint64_scale (timeduration,
avstream->avg_frame_rate.num,
+ GST_SECOND * avstream->avg_frame_rate.den));
res = TRUE;
break;
case GST_FORMAT_BYTES:
1.1
media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch
file :
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch?rev=1.1&view=markup
plain:
http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch?rev=1.1&content-type=text/plain
Index: 0.10.13_p201211-audioresample.patch
===================================================================
--- ext/ffmpeg/gstffmpegaudioresample.c~ 2015-05-29 11:38:03.639001637
+0200
+++ ext/ffmpeg/gstffmpegaudioresample.c 2015-05-29 20:08:24.744107000 +0200
@@ -24,6 +24,7 @@
#include "config.h"
#endif
+#include <libavresample/avresample.h>
#ifdef HAVE_FFMPEG_UNINSTALLED
#include <avcodec.h>
#else
@@ -37,6 +38,60 @@
#include "gstffmpeg.h"
#include "gstffmpegcodecmap.h"
+struct AudioData {
+ const AVClass *class; /**< AVClass for logging */
+ uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
+ uint8_t *buffer; /**< data buffer */
+ unsigned int buffer_size; /**< allocated buffer size */
+ int allocated_samples; /**< number of samples the buffer can
hold */
+ int nb_samples; /**< current number of samples */
+ enum AVSampleFormat sample_fmt; /**< sample format */
+ int channels; /**< channel count */
+ int allocated_channels; /**< allocated channel count */
+ int is_planar; /**< sample format is planar */
+ int planes; /**< number of data planes */
+ int sample_size; /**< bytes per sample */
+ int stride; /**< sample byte offset within a plane
*/
+ int read_only; /**< data is read-only */
+ int allow_realloc; /**< realloc is allowed */
+ int ptr_align; /**< minimum data pointer alignment */
+ int samples_align; /**< allocated samples alignment */
+ const char *name; /**< name for debug logging */
+};
+
+typedef struct AudioData AudioData;
+
+struct ReSampleContext {
+ AVAudioResampleContext *avr;
+ AudioData *buffer;
+ uint8_t *filter_bank;
+ int filter_length;
+ int ideal_dst_incr;
+ int dst_incr;
+ unsigned int index;
+ int frac;
+ int src_incr;
+ int compensation_distance;
+ int phase_shift;
+ int phase_mask;
+ int linear;
+ enum AVResampleFilterType filter_type;
+ int kaiser_beta;
+ void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
+ void (*resample_one)(struct ResampleContext *c, void *dst0,
+ int dst_index, const void *src0,
+ unsigned int index, int frac);
+ void (*resample_nearest)(void *dst0, int dst_index,
+ const void *src0, unsigned int index);
+ int padding_size;
+ int initial_padding_filled;
+ int initial_padding_samples;
+ int final_padding_filled;
+ int final_padding_samples;
+};
+
+typedef struct ReSampleContext ReSampleContext;
+
typedef struct _GstFFMpegAudioResample
{
GstBaseTransform element;
