Add m-esr52 at 52.6.0

1 files changed, 120 insertions, 0 deletions

diff --git a/media/ffvpx/libavcodec/audioconvert.c b/media/ffvpx/libavcodec/audioconvert.c
new file mode 100644
index 000000000..5e46fae2d
--- /dev/null
+++ b/media/ffvpx/libavcodec/audioconvert.c
@@ -0,0 +1,120 @@
+/*
+ * audio conversion
+ * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio conversion
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/common.h"
+#include "libavutil/libm.h"
+#include "libavutil/samplefmt.h"
+#include "avcodec.h"
+#include "audioconvert.h"
+
+#if FF_API_AUDIO_CONVERT
+
+struct AVAudioConvert {
+    int in_channels, out_channels;
+    int fmt_pair;
+};
+
+AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
+                                       enum AVSampleFormat in_fmt, int in_channels,
+                                       const float *matrix, int flags)
+{
+    AVAudioConvert *ctx;
+    if (in_channels!=out_channels)
+        return NULL;  /* FIXME: not supported */
+    ctx = av_malloc(sizeof(AVAudioConvert));
+    if (!ctx)
+        return NULL;
+    ctx->in_channels = in_channels;
+    ctx->out_channels = out_channels;
+    ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
+    return ctx;
+}
+
+void av_audio_convert_free(AVAudioConvert *ctx)
+{
+    av_free(ctx);
+}
+
+int av_audio_convert(AVAudioConvert *ctx,
+                           void * const out[6], const int out_stride[6],
+                     const void * const  in[6], const int  in_stride[6], int len)
+{
+    int ch;
+
+    //FIXME optimize common cases
+
+    for(ch=0; ch<ctx->out_channels; ch++){
+        const int is=  in_stride[ch];
+        const int os= out_stride[ch];
+        const uint8_t *pi=  in[ch];
+        uint8_t *po= out[ch];
+        uint8_t *end= po + os*len;
+        if(!out[ch])
+            continue;
+
+#define CONV(ofmt, otype, ifmt, expr)\
+if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
+    do{\
+        *(otype*)po = expr; pi += is; po += os;\
+    }while(po < end);\
+}
+
+//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
+//FIXME rounding ?
+
+             CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 ,  *(const uint8_t*)pi)
+        else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
+        else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
+        else CONV(AV_SAMPLE_FMT_FLT, float  , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
+        else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
+        else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
+        else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16,  *(const int16_t*)pi)
+        else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16,  *(const int16_t*)pi<<16)
+        else CONV(AV_SAMPLE_FMT_FLT, float  , AV_SAMPLE_FMT_S16,  *(const int16_t*)pi*(1.0 / (1<<15)))
+        else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16,  *(const int16_t*)pi*(1.0 / (1<<15)))
+        else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
+        else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32,  *(const int32_t*)pi>>16)
+        else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32,  *(const int32_t*)pi)
+        else CONV(AV_SAMPLE_FMT_FLT, float  , AV_SAMPLE_FMT_S32,  *(const int32_t*)pi*(1.0 / (1U<<31)))
+        else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32,  *(const int32_t*)pi*(1.0 / (1U<<31)))
+        else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(  lrintf(*(const float*)pi * (1<<7)) + 0x80))
+        else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(  lrintf(*(const float*)pi * (1<<15))))
+        else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
+        else CONV(AV_SAMPLE_FMT_FLT, float  , AV_SAMPLE_FMT_FLT, *(const float*)pi)
+        else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
+        else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(  lrint(*(const double*)pi * (1<<7)) + 0x80))
+        else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(  lrint(*(const double*)pi * (1<<15))))
+        else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
+        else CONV(AV_SAMPLE_FMT_FLT, float  , AV_SAMPLE_FMT_DBL, *(const double*)pi)
+        else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
+        else return -1;
+    }
+    return 0;
+}
+
+#endif /* FF_API_AUDIO_CONVERT */