Merge remote-tracking branch 'qatar/master'

* 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>

libavcodec/pcm-mpeg.c

@@ -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;
     }

libavresample/audio_mix.c

@@ -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);
 

libavresample/x86/audio_mix.asm

@@ -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

libavresample/x86/audio_mix_init.c

@@ -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);

libavresample/x86/util.asm

@@ -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

tests/fate/ac3.mak

@@ -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

tests/md5.sh

@@ -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