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From: "Pacho Ramos (pacho)" <pacho@g.o>
To: gentoo-commits@l.g.o
Subject: [gentoo-commits] gentoo-x86 commit in media-plugins/gst-plugins-ffmpeg/files: 0.10.13_p201211-snow-codec.patch 0.10.13_p201211-r_frame_rate.patch 0.10.13_p201211-audioresample.patch
Date: Sat, 30 May 2015 15:36:04
Message-Id: 20150530153559.42B76A10@oystercatcher.gentoo.org
1 pacho 15/05/30 15:35:59
2
3 Added: 0.10.13_p201211-snow-codec.patch
4 0.10.13_p201211-r_frame_rate.patch
5 0.10.13_p201211-audioresample.patch
6 Log:
7 Support libav-11 too (#509326 by Nikoli, mudler and others).
8
9 (Portage version: 2.2.20/cvs/Linux x86_64, signed Manifest commit with key A188FBD4)
10
11 Revision Changes Path
12 1.1 media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch
13
14 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
15 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
16
17 Index: 0.10.13_p201211-snow-codec.patch
18 ===================================================================
19 From: Ettore Di Giacinto <mudler@××××××××××××.org>
20 --- ext/ffmpeg/gstffmpegcodecmap.c~ 2015-05-29 10:50:06.207840323 +0200
21 +++ ext/ffmpeg/gstffmpegcodecmap.c 2015-05-29 10:50:25.638042896 +0200
22 @@ -1307,7 +1307,6 @@
23 case AV_CODEC_ID_FLIC:
24 case AV_CODEC_ID_VMDVIDEO:
25 case AV_CODEC_ID_VMDAUDIO:
26 - case AV_CODEC_ID_SNOW:
27 case AV_CODEC_ID_VIXL:
28 case AV_CODEC_ID_QPEG:
29 case AV_CODEC_ID_PGMYUV:
30
31
32
33 1.1 media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch
34
35 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
36 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
37
38 Index: 0.10.13_p201211-r_frame_rate.patch
39 ===================================================================
40 From: Ettore Di Giacinto <mudler@××××××××××××.org>
41 --- ext/ffmpeg/gstffmpegdemux.c~ 2015-05-29 00:52:07.601606544 +0200
42 +++ ext/ffmpeg/gstffmpegdemux.c 2015-05-29 01:18:18.533113323 +0200
43 @@ -781,8 +781,8 @@
44 break;
45 case GST_FORMAT_DEFAULT:
46 gst_query_set_position (query, GST_FORMAT_DEFAULT,
47 - gst_util_uint64_scale (timeposition, avstream->r_frame_rate.num,
48 - GST_SECOND * avstream->r_frame_rate.den));
49 + gst_util_uint64_scale (timeposition, avstream->avg_frame_rate.num,
50 + GST_SECOND * avstream->avg_frame_rate.den));
51 res = TRUE;
52 break;
53 case GST_FORMAT_BYTES:
54 @@ -818,8 +818,8 @@
55 break;
56 case GST_FORMAT_DEFAULT:
57 gst_query_set_duration (query, GST_FORMAT_DEFAULT,
58 - gst_util_uint64_scale (timeduration, avstream->r_frame_rate.num,
59 - GST_SECOND * avstream->r_frame_rate.den));
60 + gst_util_uint64_scale (timeduration, avstream->avg_frame_rate.num,
61 + GST_SECOND * avstream->avg_frame_rate.den));
62 res = TRUE;
63 break;
64 case GST_FORMAT_BYTES:
65
66
67
68 1.1 media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch
69
70 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
71 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
72
73 Index: 0.10.13_p201211-audioresample.patch
74 ===================================================================
75 --- ext/ffmpeg/gstffmpegaudioresample.c~ 2015-05-29 11:38:03.639001637 +0200
76 +++ ext/ffmpeg/gstffmpegaudioresample.c 2015-05-29 20:08:24.744107000 +0200
77 @@ -24,6 +24,7 @@
78 #include "config.h"
79 #endif
80
81 +#include <libavresample/avresample.h>
82 #ifdef HAVE_FFMPEG_UNINSTALLED
83 #include <avcodec.h>
84 #else
85 @@ -37,6 +38,60 @@
86 #include "gstffmpeg.h"
87 #include "gstffmpegcodecmap.h"
88
89 +struct AudioData {
90 + const AVClass *class; /**< AVClass for logging */
91 + uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
92 + uint8_t *buffer; /**< data buffer */
93 + unsigned int buffer_size; /**< allocated buffer size */
94 + int allocated_samples; /**< number of samples the buffer can hold */
95 + int nb_samples; /**< current number of samples */
96 + enum AVSampleFormat sample_fmt; /**< sample format */
97 + int channels; /**< channel count */
98 + int allocated_channels; /**< allocated channel count */
99 + int is_planar; /**< sample format is planar */
100 + int planes; /**< number of data planes */
101 + int sample_size; /**< bytes per sample */
102 + int stride; /**< sample byte offset within a plane */
103 + int read_only; /**< data is read-only */
104 + int allow_realloc; /**< realloc is allowed */
105 + int ptr_align; /**< minimum data pointer alignment */
106 + int samples_align; /**< allocated samples alignment */
107 + const char *name; /**< name for debug logging */
108 +};
109 +
110 +typedef struct AudioData AudioData;
111 +
112 +struct ReSampleContext {
113 + AVAudioResampleContext *avr;
114 + AudioData *buffer;
115 + uint8_t *filter_bank;
116 + int filter_length;
117 + int ideal_dst_incr;
118 + int dst_incr;
119 + unsigned int index;
120 + int frac;
121 + int src_incr;
122 + int compensation_distance;
123 + int phase_shift;
124 + int phase_mask;
125 + int linear;
126 + enum AVResampleFilterType filter_type;
127 + int kaiser_beta;
128 + void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
129 + void (*resample_one)(struct ResampleContext *c, void *dst0,
130 + int dst_index, const void *src0,
131 + unsigned int index, int frac);
132 + void (*resample_nearest)(void *dst0, int dst_index,
133 + const void *src0, unsigned int index);
134 + int padding_size;
135 + int initial_padding_filled;
136 + int initial_padding_samples;
137 + int final_padding_filled;
138 + int final_padding_samples;
139 +};
140 +
141 +typedef struct ReSampleContext ReSampleContext;
142 +
143 typedef struct _GstFFMpegAudioResample
144 {
145 GstBaseTransform element;
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