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Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  pcm-mpeg: improve log message wording
  fate: add missing $(TARGET_PATH) to ac3-fixed-encode
  fate: fix md5sum replacement on some systems
  avprobe: correctly set the default formatter
  lavr: add x86-optimized function for mixing 2 to 1 s16p with q8 coeffs
  lavr: add x86-optimized functions for mixing 2 to 1 s16p with float coeffs
  lavr: add C functions for mixing 2 to 1 channels with s16p format
  avprobe: move formatter functions in the context

Conflicts:
	ffprobe.c
	libavcodec/pcm-mpeg.c
	tests/fate/ac3.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/59/head
Michael Niedermayer 13 years ago
parent 39f0a45a1a d041dec3cb
commit a1fc1d2e1b
7 changed files with 195 additions and 7 deletions
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  1. 8
      libavcodec/pcm-mpeg.c
  2. 50
      libavresample/audio_mix.c
  3. 86
      libavresample/x86/audio_mix.asm
  4. 18
      libavresample/x86/audio_mix_init.c
  5. 34
      libavresample/x86/util.asm
  6. 2
      tests/fate/ac3.mak
  7. 4
      tests/md5.sh

8
libavcodec/pcm-mpeg.c
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@ -78,7 +78,7 @@ static int pcm_bluray_parse_header(AVCodecContext *avctx,
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
avctx->bits_per_raw_sample = avctx->bits_per_coded_sample;
/* get the sample rate. Not all values are known or exist. */
/* get the sample rate. Not all values are used. */
switch (header[2] & 0x0f) {
case 1:
avctx->sample_rate = 48000;
@ -91,13 +91,13 @@ static int pcm_bluray_parse_header(AVCodecContext *avctx,
break;
default:
avctx->sample_rate = 0;
av_log(avctx, AV_LOG_ERROR, "unsupported sample rate (%d)\n",
av_log(avctx, AV_LOG_ERROR, "reserved sample rate (%d)\n",
header[2] & 0x0f);
return -1;
}
/*
* get the channel number (and mapping). Not all values are known or exist.
* get the channel number (and mapping). Not all values are used.
* It must be noted that the number of channels in the MPEG stream can
* differ from the actual meaningful number, e.g. mono audio still has two
* channels, one being empty.
@ -105,7 +105,7 @@ static int pcm_bluray_parse_header(AVCodecContext *avctx,
avctx->channel_layout = channel_layouts[channel_layout];
avctx->channels = channels[channel_layout];
if (!avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "unsupported channel configuration (%d)\n",
av_log(avctx, AV_LOG_ERROR, "reserved channel configuration (%d)\n",
channel_layout);
return -1;
}

50
libavresample/audio_mix.c
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@ -115,6 +115,50 @@ static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
}
}
static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
while (len > 4) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len -= 4;
}
while (len > 0) {
*dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1));
len--;
}
}
static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len,
int out_ch, int in_ch)
{
int16_t *src0 = samples[0];
int16_t *src1 = samples[1];
int16_t *dst = src0;
int16_t m0 = matrix[0][0];
int16_t m1 = matrix[0][1];
while (len > 4) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len -= 4;
}
while (len > 0) {
*dst++ = (*src0++ * m0 + *src1++ * m1) >> 8;
len--;
}
}
static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
@ -229,6 +273,12 @@ static int mix_function_init(AudioMix *am)
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);

86
libavresample/x86/audio_mix.asm
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@ -21,6 +21,7 @@
%include "x86inc.asm"
%include "x86util.asm"
%include "util.asm"
SECTION_TEXT
@ -64,3 +65,88 @@ MIX_2_TO_1_FLTP_FLT
INIT_YMM avx
MIX_2_TO_1_FLTP_FLT
%endif
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_s16p_flt(int16_t **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
%macro MIX_2_TO_1_S16P_FLT 0
cglobal mix_2_to_1_s16p_flt, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq]
sub src1q, srcq
mov matrixq, [matrixq ]
VBROADCASTSS m4, [matrixq ]
VBROADCASTSS m5, [matrixq+4]
ALIGN 16
.loop:
mova m0, [srcq ]
mova m2, [srcq+src1q]
S16_TO_S32_SX 0, 1
S16_TO_S32_SX 2, 3
cvtdq2ps m0, m0
cvtdq2ps m1, m1
cvtdq2ps m2, m2
cvtdq2ps m3, m3
mulps m0, m4
mulps m1, m4
mulps m2, m5
mulps m3, m5
addps m0, m2
addps m1, m3
cvtps2dq m0, m0
cvtps2dq m1, m1
packssdw m0, m1
mova [srcq], m0
add srcq, mmsize
sub lend, mmsize/2
jg .loop
REP_RET
%endmacro
INIT_XMM sse2
MIX_2_TO_1_S16P_FLT
INIT_XMM sse4
MIX_2_TO_1_S16P_FLT
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_s16p_q8(int16_t **src, int16_t **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
INIT_XMM sse2
cglobal mix_2_to_1_s16p_q8, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq]
sub src1q, srcq
mov matrixq, [matrixq]
movd m4, [matrixq]
movd m5, [matrixq]
SPLATW m4, m4, 0
SPLATW m5, m5, 1
pxor m0, m0
punpcklwd m4, m0
punpcklwd m5, m0
ALIGN 16
.loop:
mova m0, [srcq ]
mova m2, [srcq+src1q]
punpckhwd m1, m0, m0
punpcklwd m0, m0
punpckhwd m3, m2, m2
punpcklwd m2, m2
pmaddwd m0, m4
pmaddwd m1, m4
pmaddwd m2, m5
pmaddwd m3, m5
paddd m0, m2
paddd m1, m3
psrad m0, 8
psrad m1, 8
packssdw m0, m1
mova [srcq], m0
add srcq, mmsize
sub lend, mmsize/2
jg .loop
REP_RET

18
libavresample/x86/audio_mix_init.c
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@ -27,6 +27,14 @@ extern void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len,
extern void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_2_to_1_s16p_flt_sse2(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_2_to_1_s16p_flt_sse4(int16_t **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_2_to_1_s16p_q8_sse2(int16_t **src, int16_t **matrix,
int len, int out_ch, int in_ch);
av_cold void ff_audio_mix_init_x86(AudioMix *am)
{
#if HAVE_YASM
@ -36,6 +44,16 @@ av_cold void ff_audio_mix_init_x86(AudioMix *am)
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse);
}
if (mm_flags & AV_CPU_FLAG_SSE2 && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_flt_sse2);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8,
2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_q8_sse2);
}
if (mm_flags & AV_CPU_FLAG_SSE4 && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE4", ff_mix_2_to_1_s16p_flt_sse4);
}
if (mm_flags & AV_CPU_FLAG_AVX && HAVE_AVX) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx);

34
libavresample/x86/util.asm
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@ -0,0 +1,34 @@
;******************************************************************************
;* x86 utility macros for libavresample
;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
;*
;* This file is part of Libav.
;*
;* Libav 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.
;*
;* Libav 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 Libav; if not, write to the Free Software
;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
;******************************************************************************
%macro S16_TO_S32_SX 2 ; src/low dst, high dst
%if cpuflag(sse4)
pmovsxwd m%2, m%1
psrldq m%1, 8
pmovsxwd m%1, m%1
SWAP %1, %2
%else
punpckhwd m%2, m%1
punpcklwd m%1, m%1
psrad m%2, 16
psrad m%1, 16
%endif
%endmacro

2
tests/fate/ac3.mak
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@ -48,7 +48,7 @@ fate-eac3-encode: FUZZ = 3
FATE_AC3 += fate-ac3-fixed-encode
fate-ac3-fixed-encode: tests/data/asynth-44100-2.wav
fate-ac3-fixed-encode: SRC = tests/data/asynth-44100-2.wav
fate-ac3-fixed-encode: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-ac3-fixed-encode: CMD = md5 -i $(SRC) -c ac3_fixed -ab 128k -f ac3 -flags +bitexact
fate-ac3-fixed-encode: CMP = oneline
fate-ac3-fixed-encode: REF = a1d1fc116463b771abf5aef7ed37d7b1

4
tests/md5.sh
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@ -2,8 +2,8 @@
if [ X"$(echo | md5sum 2> /dev/null)" != X ]; then
do_md5sum() { md5sum -b $1; }
elif [ X"$(echo | md5 2> /dev/null)" != X ]; then
do_md5sum() { md5 $1 | sed 's#MD5 (\(.*\)) = \(.*\)#\2 *\1#'; }
elif [ X"$(echo | command md5 2> /dev/null)" != X ]; then
do_md5sum() { command md5 $1 | sed 's#MD5 (\(.*\)) = \(.*\)#\2 *\1#'; }
elif [ -x /sbin/md5 ]; then
do_md5sum() { /sbin/md5 -r $1 | sed 's# \**\./# *./#'; }
elif openssl version >/dev/null 2>&1; then

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