[SCM] ffmpeg/master: Drop patches, included upstream.
aca-guest at users.alioth.debian.org
aca-guest at users.alioth.debian.org
Sun Oct 30 11:31:28 UTC 2016
The following commit has been merged in the master branch:
commit 0a61dd00be42bdf06e0ec6555be842aa204ba7db
Author: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
Date: Sat Oct 29 18:59:10 2016 +0200
Drop patches, included upstream.
diff --git a/debian/patches/doc-fix-spelling-errors.patch b/debian/patches/doc-fix-spelling-errors.patch
deleted file mode 100644
index b4654f6..0000000
--- a/debian/patches/doc-fix-spelling-errors.patch
+++ /dev/null
@@ -1,240 +0,0 @@
-From: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
-Date: Sat, 22 Oct 2016 20:43:40 +0200
-Subject: doc: fix spelling errors
-
-Thanks to Mathieu Malaterre <malat at debian.org> for reporting the
-Que/Queue typo. (https://bugs.debian.org/839542)
-
-Reviewed-by: Lou Logan <lou at lrcd.com>
-Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
----
- doc/encoders.texi | 2 +-
- doc/ffprobe.texi | 2 +-
- doc/filters.texi | 4 ++--
- ffmpeg.c | 2 +-
- ffmpeg_cuvid.c | 4 ++--
- libavcodec/aaccoder_twoloop.h | 2 +-
- libavcodec/cabac.c | 2 +-
- libavcodec/ffjni.c | 2 +-
- libavcodec/mediacodec_wrapper.h | 2 +-
- libavcodec/psymodel.h | 2 +-
- libavcodec/x86/vp9lpf_16bpp.asm | 2 +-
- libavfilter/f_ebur128.c | 2 +-
- libavformat/internal.h | 2 +-
- libavutil/frame.h | 2 +-
- libavutil/tree.h | 2 +-
- 15 files changed, 17 insertions(+), 17 deletions(-)
-
-diff --git a/doc/encoders.texi b/doc/encoders.texi
-index f38cad3..f2a9950 100644
---- a/doc/encoders.texi
-+++ b/doc/encoders.texi
-@@ -1866,7 +1866,7 @@ Enable CAVLC and disable CABAC. It generates the same effect as
- @end table
-
- @item cmp
--Set full pixel motion estimation comparation algorithm. Possible values:
-+Set full pixel motion estimation comparison algorithm. Possible values:
-
- @table @samp
- @item chroma
-diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi
-index 2024eed..1069ae3 100644
---- a/doc/ffprobe.texi
-+++ b/doc/ffprobe.texi
-@@ -245,7 +245,7 @@ continue reading from that.
- Each interval is specified by two optional parts, separated by "%".
-
- The first part specifies the interval start position. It is
--interpreted as an abolute position, or as a relative offset from the
-+interpreted as an absolute position, or as a relative offset from the
- current position if it is preceded by the "+" character. If this first
- part is not specified, no seeking will be performed when reading this
- interval.
-diff --git a/doc/filters.texi b/doc/filters.texi
-index b482236..c602c72 100644
---- a/doc/filters.texi
-+++ b/doc/filters.texi
-@@ -1120,7 +1120,7 @@ Set video stream size. Only useful if curves option is activated.
-
- @item mgain
- Set max gain that will be displayed. Only useful if curves option is activated.
--Setting this to reasonable value allows to display gain which is derived from
-+Setting this to a reasonable value makes it possible to display gain which is derived from
- neighbour bands which are too close to each other and thus produce higher gain
- when both are activated.
-
-@@ -16753,7 +16753,7 @@ magnitude across time and second represents phase across time.
- The filter will transform from frequency domain as displayed in videos back
- to time domain as presented in audio output.
-
--This filter is primarly created for reversing processed @ref{showspectrum}
-+This filter is primarily created for reversing processed @ref{showspectrum}
- filter outputs, but can synthesize sound from other spectrograms too.
- But in such case results are going to be poor if the phase data is not
- available, because in such cases phase data need to be recreated, usually
-diff --git a/ffmpeg.c b/ffmpeg.c
-index cdded86..80c76dd 100644
---- a/ffmpeg.c
-+++ b/ffmpeg.c
-@@ -3546,7 +3546,7 @@ static int check_keyboard_interaction(int64_t cur_time)
- "+ increase verbosity\n"
- "- decrease verbosity\n"
- "c Send command to first matching filter supporting it\n"
-- "C Send/Que command to all matching filters\n"
-+ "C Send/Queue command to all matching filters\n"
- "D cycle through available debug modes\n"
- "h dump packets/hex press to cycle through the 3 states\n"
- "q quit\n"
-diff --git a/ffmpeg_cuvid.c b/ffmpeg_cuvid.c
-index 7fb47a2..fe9ce84 100644
---- a/ffmpeg_cuvid.c
-+++ b/ffmpeg_cuvid.c
-@@ -187,7 +187,7 @@ int cuvid_transcode_init(OutputStream *ost)
- }
-
- /* This is a bit hacky, av_hwframe_ctx_init is called by the cuvid decoder
-- * once it has probed the neccesary format information. But as filters/nvenc
-+ * once it has probed the necessary format information. But as filters/nvenc
- * need to know the format/sw_format, set them here so they are happy.
- * This is fine as long as CUVID doesn't add another supported pix_fmt.
- */
-@@ -229,7 +229,7 @@ error:
-
- cancel:
- if (ist->hwaccel_id == HWACCEL_CUVID) {
-- av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achive.\n");
-+ av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achieve.\n");
- return AVERROR(EINVAL);
- }
-
-diff --git a/libavcodec/aaccoder_twoloop.h b/libavcodec/aaccoder_twoloop.h
-index 42aea52..e44f69a 100644
---- a/libavcodec/aaccoder_twoloop.h
-+++ b/libavcodec/aaccoder_twoloop.h
-@@ -87,7 +87,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
- * will keep iterating until it fails to lower it or it reaches
- * ulimit * rdlambda. Keeping it low increases quality on difficult
- * signals, but lower it too much, and bits will be taken from weak
-- * signals, creating "holes". A balance is necesary.
-+ * signals, creating "holes". A balance is necessary.
- * rdmax and rdmin specify the relative deviation from rdlambda
- * allowed for tonality compensation
- */
-diff --git a/libavcodec/cabac.c b/libavcodec/cabac.c
-index c0abe83..dd2b057 100644
---- a/libavcodec/cabac.c
-+++ b/libavcodec/cabac.c
-@@ -182,7 +182,7 @@ int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
- #if CABAC_BITS == 16
- c->low = (*c->bytestream++)<<18;
- c->low+= (*c->bytestream++)<<10;
-- // Keep our fetches on a 2-byte boundry as this should avoid ever having to
-+ // Keep our fetches on a 2-byte boundary as this should avoid ever having to
- // do unaligned loads if the compiler (or asm) optimises the double byte
- // load into a single instruction
- if(((uintptr_t)c->bytestream & 1) == 0) {
-diff --git a/libavcodec/ffjni.c b/libavcodec/ffjni.c
-index 82ee5d3..cb1884f 100644
---- a/libavcodec/ffjni.c
-+++ b/libavcodec/ffjni.c
-@@ -215,7 +215,7 @@ int ff_jni_exception_get_summary(JNIEnv *env, jthrowable exception, char **error
- } else if (!name && message) {
- av_bprintf(&bp, "Exception: %s", message);
- } else {
-- av_log(log_ctx, AV_LOG_WARNING, "Could not retreive exception name and message\n");
-+ av_log(log_ctx, AV_LOG_WARNING, "Could not retrieve exception name and message\n");
- av_bprintf(&bp, "Exception occurred");
- }
-
-diff --git a/libavcodec/mediacodec_wrapper.h b/libavcodec/mediacodec_wrapper.h
-index cddd420..1b4f3a9 100644
---- a/libavcodec/mediacodec_wrapper.h
-+++ b/libavcodec/mediacodec_wrapper.h
-@@ -44,7 +44,7 @@
- * implementation.
- *
- * The API around MediaCodecList is not part of the NDK (and is lacking as
-- * we still need to retreive the codec name to work around faulty decoders
-+ * we still need to retrieve the codec name to work around faulty decoders
- * and encoders).
- *
- * For documentation, please refers to NdkMediaCodec.h NdkMediaFormat.h and
-diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
-index 35d184c..582f040 100644
---- a/libavcodec/psymodel.h
-+++ b/libavcodec/psymodel.h
-@@ -29,7 +29,7 @@
- /** maximum number of channels */
- #define PSY_MAX_CHANS 20
-
--/* cutoff for VBR is purposedly increased, since LP filtering actually
-+/* cutoff for VBR is purposely increased, since LP filtering actually
- * hinders VBR performance rather than the opposite
- */
- #define AAC_CUTOFF_FROM_BITRATE(bit_rate,channels,sample_rate) (bit_rate ? FFMIN3(FFMIN3( \
-diff --git a/libavcodec/x86/vp9lpf_16bpp.asm b/libavcodec/x86/vp9lpf_16bpp.asm
-index c15437b..c088817 100644
---- a/libavcodec/x86/vp9lpf_16bpp.asm
-+++ b/libavcodec/x86/vp9lpf_16bpp.asm
-@@ -78,7 +78,7 @@ SECTION .text
- %endif
- %endmacro
-
--; calulate p or q portion of flat8out
-+; calculate p or q portion of flat8out
- %macro FLAT8OUT_HALF 0
- psubw m4, m0 ; q4-q0
- psubw m5, m0 ; q5-q0
-diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c
-index 59eaedd..983c889 100644
---- a/libavfilter/f_ebur128.c
-+++ b/libavfilter/f_ebur128.c
-@@ -141,7 +141,7 @@ typedef struct {
- int loglevel; ///< log level for frame logging
- int metadata; ///< whether or not to inject loudness results in frames
- int dual_mono; ///< whether or not to treat single channel input files as dual-mono
-- double pan_law; ///< pan law value used to calulate dual-mono measurements
-+ double pan_law; ///< pan law value used to calculate dual-mono measurements
- } EBUR128Context;
-
- enum {
-diff --git a/libavformat/internal.h b/libavformat/internal.h
-index 647ad65..31214f9 100644
---- a/libavformat/internal.h
-+++ b/libavformat/internal.h
-@@ -626,7 +626,7 @@ int ff_bprint_to_codecpar_extradata(AVCodecParameters *par, struct AVBPrint *buf
- * The packet is not removed from the interleaving queue, but only
- * a pointer to it is returned.
- *
-- * @param ts_offset the ts difference between packet in the que and the muxer.
-+ * @param ts_offset the ts difference between packet in the queue and the muxer.
- *
- * @return a pointer to the next packet, or NULL if no packet is queued
- * for this stream.
-diff --git a/libavutil/frame.h b/libavutil/frame.h
-index 2b5c332..ef87806 100644
---- a/libavutil/frame.h
-+++ b/libavutil/frame.h
-@@ -178,7 +178,7 @@ typedef struct AVFrameSideData {
- * without breaking compatibility with each other.
- *
- * Fields can be accessed through AVOptions, the name string used, matches the
-- * C structure field name for fields accessable through AVOptions. The AVClass
-+ * C structure field name for fields accessible through AVOptions. The AVClass
- * for AVFrame can be obtained from avcodec_get_frame_class()
- */
- typedef struct AVFrame {
-diff --git a/libavutil/tree.h b/libavutil/tree.h
-index 9a9e11b..d5e0aeb 100644
---- a/libavutil/tree.h
-+++ b/libavutil/tree.h
-@@ -58,7 +58,7 @@ struct AVTreeNode *av_tree_node_alloc(void);
- * then the corresponding entry in next is unchanged.
- * @param cmp compare function used to compare elements in the tree,
- * API identical to that of Standard C's qsort
-- * It is guranteed that the first and only the first argument to cmp()
-+ * It is guaranteed that the first and only the first argument to cmp()
- * will be the key parameter to av_tree_find(), thus it could if the
- * user wants, be a different type (like an opaque context).
- * @return An element with cmp(key, elem) == 0 or NULL if no such element
diff --git a/debian/patches/faq-use-relative-links-to-own-documentation.patch b/debian/patches/faq-use-relative-links-to-own-documentation.patch
deleted file mode 100644
index 29781d7..0000000
--- a/debian/patches/faq-use-relative-links-to-own-documentation.patch
+++ /dev/null
@@ -1,71 +0,0 @@
-From: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
-Date: Sat, 22 Oct 2016 20:12:30 +0200
-Subject: faq: use relative links to own documentation
-
-This way locally installed documentation refers to itself instead of the
-website.
-
-Bud-Id: https://bugs.debian.org/841501
-Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
----
- doc/faq.texi | 14 +++++++-------
- 1 file changed, 7 insertions(+), 7 deletions(-)
-
-diff --git a/doc/faq.texi b/doc/faq.texi
-index ef111c7..ff35c89 100644
---- a/doc/faq.texi
-+++ b/doc/faq.texi
-@@ -311,18 +311,18 @@ invoking ffmpeg with several @option{-i} options.
- For audio, to put all channels together in a single stream (example: two
- mono streams into one stereo stream): this is sometimes called to
- @emph{merge} them, and can be done using the
-- at url{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
-+ at url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
-
- @item
- For audio, to play one on top of the other: this is called to @emph{mix}
- them, and can be done by first merging them into a single stream and then
--using the @url{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
-+using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
- the channels at will.
-
- @item
- For video, to display both together, side by side or one on top of a part of
- the other; it can be done using the
-- at url{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
-+ at url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
-
- @end itemize
-
-@@ -333,19 +333,19 @@ There are several solutions, depending on the exact circumstances.
-
- @subsection Concatenating using the concat @emph{filter}
-
--FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat,
-+FFmpeg has a @url{ffmpeg-filters.html#concat,
- @code{concat}} filter designed specifically for that, with examples in the
- documentation. This operation is recommended if you need to re-encode.
-
- @subsection Concatenating using the concat @emph{demuxer}
-
--FFmpeg has a @url{https://www.ffmpeg.org/ffmpeg-formats.html#concat,
-+FFmpeg has a @url{ffmpeg-formats.html#concat,
- @code{concat}} demuxer which you can use when you want to avoid a re-encode and
- your format doesn't support file level concatenation.
-
- @subsection Concatenating using the concat @emph{protocol} (file level)
-
--FFmpeg has a @url{https://ffmpeg.org/ffmpeg-protocols.html#concat,
-+FFmpeg has a @url{ffmpeg-protocols.html#concat,
- @code{concat}} protocol designed specifically for that, with examples in the
- documentation.
-
-@@ -485,7 +485,7 @@ scaling adjusts the SAR to keep the DAR constant.
-
- If you want to stretch, or “unstretch”, the image, you need to override the
- information with the
-- at url{https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
-+ at url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
-
- Do not forget to examine carefully the original video to check whether the
- stretching comes from the image or from the aspect ratio information.
diff --git a/debian/patches/ffmpeg_opt-Suggest-to-use-file-.-if-a-protocol-was-not-fo.patch b/debian/patches/ffmpeg_opt-Suggest-to-use-file-.-if-a-protocol-was-not-fo.patch
deleted file mode 100644
index 2c41f0c..0000000
--- a/debian/patches/ffmpeg_opt-Suggest-to-use-file-.-if-a-protocol-was-not-fo.patch
+++ /dev/null
@@ -1,22 +0,0 @@
-From: Carl Eugen Hoyos <cehoyos at ag.or.at>
-Date: Tue, 6 Sep 2016 12:47:34 +0200
-Subject: ffmpeg_opt: Suggest to use "file:..." if a protocol was not found.
-
-Fixes Debian bug 785690.
----
- ffmpeg_opt.c | 2 ++
- 1 file changed, 2 insertions(+)
-
-diff --git a/ffmpeg_opt.c b/ffmpeg_opt.c
-index 7785a30..f96c655 100644
---- a/ffmpeg_opt.c
-+++ b/ffmpeg_opt.c
-@@ -982,6 +982,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
- err = avformat_open_input(&ic, filename, file_iformat, &o->g->format_opts);
- if (err < 0) {
- print_error(filename, err);
-+ if (err == AVERROR_PROTOCOL_NOT_FOUND)
-+ av_log(NULL, AV_LOG_ERROR, "Did you mean file:%s?\n", filename);
- exit_program(1);
- }
- if (scan_all_pmts_set)
diff --git a/debian/patches/lavf-mp3enc-write-encoder-delay-padding-upon-closing.patch b/debian/patches/lavf-mp3enc-write-encoder-delay-padding-upon-closing.patch
deleted file mode 100644
index 44cc05b..0000000
--- a/debian/patches/lavf-mp3enc-write-encoder-delay-padding-upon-closing.patch
+++ /dev/null
@@ -1,79 +0,0 @@
-From: Jon Toohill <jtoohill at google.com>
-Date: Mon, 17 Oct 2016 14:55:04 -0700
-Subject: lavf/mp3enc: write encoder delay/padding upon closing
-
-Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
----
- libavformat/mp3enc.c | 34 ++++++++++++++++++++++++++++------
- 1 file changed, 28 insertions(+), 6 deletions(-)
-
-diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
-index de63401..4c97fa1 100644
---- a/libavformat/mp3enc.c
-+++ b/libavformat/mp3enc.c
-@@ -111,6 +111,8 @@ typedef struct MP3Context {
- uint64_t bag[XING_NUM_BAGS];
- int initial_bitrate;
- int has_variable_bitrate;
-+ int delay;
-+ int padding;
-
- /* index of the audio stream */
- int audio_stream_idx;
-@@ -247,12 +249,7 @@ static int mp3_write_xing(AVFormatContext *s)
- ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
- avio_w8(dyn_ctx, 0); // unknown encoding flags
- avio_w8(dyn_ctx, 0); // unknown abr/minimal bitrate
--
-- // encoder delay
-- if (par->initial_padding - 528 - 1 >= 1 << 12) {
-- av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-- }
-- avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
-+ avio_wb24(dyn_ctx, 0); // empty encoder delay/padding
-
- avio_w8(dyn_ctx, 0); // misc
- avio_w8(dyn_ctx, 0); // mp3gain
-@@ -345,10 +342,24 @@ static int mp3_write_audio_packet(AVFormatContext *s, AVPacket *pkt)
- #endif
-
- if (mp3->xing_offset) {
-+ uint8_t *side_data = NULL;
-+ int side_data_size = 0;
-+
- mp3_xing_add_frame(mp3, pkt);
- mp3->audio_size += pkt->size;
- mp3->audio_crc = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE),
- mp3->audio_crc, pkt->data, pkt->size);
-+
-+ side_data = av_packet_get_side_data(pkt,
-+ AV_PKT_DATA_SKIP_SAMPLES,
-+ &side_data_size);
-+ if (side_data && side_data_size >= 10) {
-+ mp3->padding = FFMAX(AV_RL32(side_data + 4) + 528 + 1, 0);
-+ if (!mp3->delay)
-+ mp3->delay = FFMAX(AV_RL32(side_data) - 528 - 1, 0);
-+ } else {
-+ mp3->padding = 0;
-+ }
- }
- }
-
-@@ -422,6 +433,17 @@ static void mp3_update_xing(AVFormatContext *s)
- }
- }
-
-+ /* write encoder delay/padding */
-+ if (mp3->delay >= 1 << 12) {
-+ mp3->delay = (1 << 12) - 1;
-+ av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-+ }
-+ if (mp3->padding >= 1 << 12) {
-+ mp3->padding = (1 << 12) - 1;
-+ av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
-+ }
-+ AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (mp3->delay << 12) + mp3->padding);
-+
- AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, mp3->audio_size);
- AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, mp3->audio_crc);
-
diff --git a/debian/patches/series b/debian/patches/series
deleted file mode 100644
index 138dd61..0000000
--- a/debian/patches/series
+++ /dev/null
@@ -1,5 +0,0 @@
-ffmpeg_opt-Suggest-to-use-file-.-if-a-protocol-was-not-fo.patch
-lavf-mp3enc-write-encoder-delay-padding-upon-closing.patch
-doc-fix-spelling-errors.patch
-faq-use-relative-links-to-own-documentation.patch
-tests-checkasm-pixblockdsp-Test-8-byte-aligned-positions.patch
diff --git a/debian/patches/tests-checkasm-pixblockdsp-Test-8-byte-aligned-positions.patch b/debian/patches/tests-checkasm-pixblockdsp-Test-8-byte-aligned-positions.patch
deleted file mode 100644
index b17e135..0000000
--- a/debian/patches/tests-checkasm-pixblockdsp-Test-8-byte-aligned-positions.patch
+++ /dev/null
@@ -1,42 +0,0 @@
-From: Michael Niedermayer <michael at niedermayer.cc>
-Date: Mon, 27 Jun 2016 22:03:14 +0200
-Subject: tests/checkasm/pixblockdsp: Test 8 byte aligned positions
-
-The code is documented as to require 8byte alignment
-
-Signed-off-by: Michael Niedermayer <michael at niedermayer.cc>
----
- tests/checkasm/pixblockdsp.c | 6 +++---
- 1 file changed, 3 insertions(+), 3 deletions(-)
-
-diff --git a/tests/checkasm/pixblockdsp.c b/tests/checkasm/pixblockdsp.c
-index 66bfdb7..2b88e7d 100644
---- a/tests/checkasm/pixblockdsp.c
-+++ b/tests/checkasm/pixblockdsp.c
-@@ -26,7 +26,7 @@
- #include "libavutil/intreadwrite.h"
-
- #define BUF_UNITS 8
--#define BUF_SIZE (BUF_UNITS * 128 + BUF_UNITS)
-+#define BUF_SIZE (BUF_UNITS * 128 + 8 * BUF_UNITS)
-
- #define randomize_buffers() \
- do { \
-@@ -50,7 +50,7 @@
- declare_func_emms(AV_CPU_FLAG_MMX, void, int16_t *block, const uint8_t *pixels, ptrdiff_t line_size); \
- \
- for (i = 0; i < BUF_UNITS; i++) { \
-- int src_offset = i * 64 * sizeof(type) + i; /* Test various alignments */ \
-+ int src_offset = i * 64 * sizeof(type) + 8 * i; /* Test various alignments */ \
- int dst_offset = i * 64; /* dst must be aligned */ \
- randomize_buffers(); \
- call_ref(dst0 + dst_offset, src10 + src_offset, 8); \
-@@ -67,7 +67,7 @@
- declare_func_emms(AV_CPU_FLAG_MMX, void, int16_t *av_restrict block, const uint8_t *s1, const uint8_t *s2, int stride); \
- \
- for (i = 0; i < BUF_UNITS; i++) { \
-- int src_offset = i * 64 * sizeof(type) + i; /* Test various alignments */ \
-+ int src_offset = i * 64 * sizeof(type) + 8 * i; /* Test various alignments */ \
- int dst_offset = i * 64; /* dst must be aligned */ \
- randomize_buffers(); \
- call_ref(dst0 + dst_offset, src10 + src_offset, src20 + src_offset, 8); \
--
ffmpeg packaging
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