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