@@ -153,7 +153,7 @@ static uint8_t *audio_buf;

 static uint8_t *audio_out;

 static unsigned int allocated_audio_out_size, allocated_audio_buf_size;

-static short *samples;

+static void *samples;

 #define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"

@@ -1586,7 +1586,7 @@ static int output_packet(InputStream *ist, int ist_index,

 {

     AVFormatContext *os;

     OutputStream *ost;

-    int ret, i;

+    int ret = 0, i;

     int got_output;

     void *buffer_to_free = NULL;

     static unsigned int samples_size= 0;

@@ -1641,8 +1641,8 @@ static int output_packet(InputStream *ist, int ist_index,

         if (ist->decoding_needed) {

             switch(ist->st->codec->codec_type) {

             case AVMEDIA_TYPE_AUDIO:{

-                if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) {

-                    samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE);

+                if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {

+                    samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);

                     av_free(samples);

                     samples= av_malloc(samples_size);

                 }

@@ -1731,11 +1731,57 @@ static int output_packet(InputStream *ist, int ist_index,

         // preprocess audio (volume)

         if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {

             if (audio_volume != 256) {

-                short *volp;

-                volp = samples;

-                for(i=0;i<(decoded_data_size / sizeof(short));i++) {

-                    int v = ((*volp) * audio_volume + 128) >> 8;

-                    *volp++ = av_clip_int16(v);

+                switch (ist->st->codec->sample_fmt) {

+                case AV_SAMPLE_FMT_U8:

+                {

+                    uint8_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;

+                        *volp++ = av_clip_uint8(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_S16:

+                {

+                    int16_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int v = ((*volp) * audio_volume + 128) >> 8;

+                        *volp++ = av_clip_int16(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_S32:

+                {

+                    int32_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);

+                        *volp++ = av_clipl_int32(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_FLT:

+                {

+                    float *volp = samples;

+                    float scale = audio_volume / 256.f;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        *volp++ *= scale;

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_DBL:

+                {

+                    double *volp = samples;

+                    double scale = audio_volume / 256.;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        *volp++ *= scale;

+                    }

+                    break;

+                }

+                default:

+                    av_log(NULL, AV_LOG_FATAL,

+                           "Audio volume adjustment on sample format %s is not supported.\n",

+                           av_get_sample_fmt_name(ist->st->codec->sample_fmt));

+                    exit_program(1);

                 }

             }

         }

@@ -1687,6 +1687,9 @@ input sample aspect ratio

 @item dar

 input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar}

+@item sar

+input sample aspect ratio

+

 @item hsub, vsub

 horizontal and vertical chroma subsample values. For example for the

 pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1.

@@ -160,7 +160,7 @@ static uint8_t *audio_buf;

 static uint8_t *audio_out;

 static unsigned int allocated_audio_out_size, allocated_audio_buf_size;

-static short *samples;

+static void *samples;

 static uint8_t *input_tmp= NULL;

 #define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"

@@ -1596,7 +1596,7 @@ static int output_packet(InputStream *ist, int ist_index,

 {

     AVFormatContext *os;

     OutputStream *ost;

-    int ret, i;

+    int ret = 0, i;

     int got_output;

     void *buffer_to_free = NULL;

     static unsigned int samples_size= 0;

@@ -1651,8 +1651,8 @@ static int output_packet(InputStream *ist, int ist_index,

         if (ist->decoding_needed) {

             switch(ist->st->codec->codec_type) {

             case AVMEDIA_TYPE_AUDIO:{

-                if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) {

-                    samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE);

+                if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {

+                    samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);

                     av_free(samples);

                     samples= av_malloc(samples_size);

                 }

@@ -1758,11 +1758,57 @@ static int output_packet(InputStream *ist, int ist_index,

         // preprocess audio (volume)

         if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {

             if (audio_volume != 256) {

-                short *volp;

-                volp = samples;

-                for(i=0;i<(decoded_data_size / sizeof(short));i++) {

-                    int v = ((*volp) * audio_volume + 128) >> 8;

-                    *volp++ = av_clip_int16(v);

+                switch (ist->st->codec->sample_fmt) {

+                case AV_SAMPLE_FMT_U8:

+                {

+                    uint8_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;

+                        *volp++ = av_clip_uint8(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_S16:

+                {

+                    int16_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int v = ((*volp) * audio_volume + 128) >> 8;

+                        *volp++ = av_clip_int16(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_S32:

+                {

+                    int32_t *volp = samples;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);

+                        *volp++ = av_clipl_int32(v);

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_FLT:

+                {

+                    float *volp = samples;

+                    float scale = audio_volume / 256.f;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        *volp++ *= scale;

+                    }

+                    break;

+                }

+                case AV_SAMPLE_FMT_DBL:

+                {

+                    double *volp = samples;

+                    double scale = audio_volume / 256.;

+                    for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

+                        *volp++ *= scale;

+                    }

+                    break;

+                }

+                default:

+                    av_log(NULL, AV_LOG_FATAL,

+                           "Audio volume adjustment on sample format %s is not supported.\n",

+                           av_get_sample_fmt_name(ist->st->codec->sample_fmt));

+                    exit_program(1);

                 }

             }

         }

@@ -143,6 +143,8 @@ void ff_vector_fmul_window_neon(float *dst, const float *src0,

                                 const float *src1, const float *win, int len);

 void ff_vector_fmul_scalar_neon(float *dst, const float *src, float mul,

                                 int len);

+void ff_vector_fmac_scalar_neon(float *dst, const float *src, float mul,

+                                int len);

 void ff_butterflies_float_neon(float *v1, float *v2, int len);

 float ff_scalarproduct_float_neon(const float *v1, const float *v2, int len);

 void ff_vector_fmul_reverse_neon(float *dst, const float *src0,

@@ -305,6 +307,7 @@ void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx)

     c->vector_fmul                = ff_vector_fmul_neon;

     c->vector_fmul_window         = ff_vector_fmul_window_neon;

     c->vector_fmul_scalar         = ff_vector_fmul_scalar_neon;

+    c->vector_fmac_scalar         = ff_vector_fmac_scalar_neon;

     c->butterflies_float          = ff_butterflies_float_neon;

     c->scalarproduct_float        = ff_scalarproduct_float_neon;

     c->vector_fmul_reverse        = ff_vector_fmul_reverse_neon;

@@ -587,6 +587,54 @@ NOVFP   vdup.32         q8,  r2

         .unreq          len

 endfunc

+function ff_vector_fmac_scalar_neon, export=1

+VFP     len .req r2

+VFP     acc .req r3

+NOVFP   len .req r3

+NOVFP   acc .req r2

+VFP     vdup.32         q15, d0[0]

+NOVFP   vdup.32         q15, r2

+        bics            r12, len, #15

+        mov             acc, r0

+        beq             3f

+        vld1.32         {q0},     [r1,:128]!

+        vld1.32         {q8},     [acc,:128]!

+        vld1.32         {q1},     [r1,:128]!

+        vld1.32         {q9},     [acc,:128]!

+1:      vmla.f32        q8,  q0,  q15

+        vld1.32         {q2},     [r1,:128]!

+        vld1.32         {q10},    [acc,:128]!

+        vmla.f32        q9,  q1,  q15

+        vld1.32         {q3},     [r1,:128]!

+        vld1.32         {q11},    [acc,:128]!

+        vmla.f32        q10, q2,  q15

+        vst1.32         {q8},     [r0,:128]!

+        vmla.f32        q11, q3,  q15

+        vst1.32         {q9},     [r0,:128]!

+        subs            r12, r12, #16

+        beq             2f

+        vld1.32         {q0},     [r1,:128]!

+        vld1.32         {q8},     [acc,:128]!

+        vst1.32         {q10},    [r0,:128]!

+        vld1.32         {q1},     [r1,:128]!

+        vld1.32         {q9},     [acc,:128]!

+        vst1.32         {q11},    [r0,:128]!

+        b               1b

+2:      vst1.32         {q10},    [r0,:128]!

+        vst1.32         {q11},    [r0,:128]!

+        ands            len, len, #15

+        it              eq

+        bxeq            lr

+3:      vld1.32         {q0},     [r1,:128]!

+        vld1.32         {q8},     [acc,:128]!

+        vmla.f32        q8,  q0,  q15

+        vst1.32         {q8},     [r0,:128]!

+        subs            len, len, #4

+        bgt             3b

+        bx              lr

+        .unreq          len

+endfunc

+

 function ff_butterflies_float_neon, export=1

 1:      vld1.32         {q0},[r0,:128]

         vld1.32         {q1},[r1,:128]

@@ -1833,11 +1833,8 @@ static int dca_decode_frame(AVCodecContext * avctx,

             float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;

             float* lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;

             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;

-            int j;

-            for(j = 0; j < 256; ++j) {

-                lt_chan[j] -= back_chan[j] * M_SQRT1_2;

-                rt_chan[j] -= back_chan[j] * M_SQRT1_2;

-            }

+            s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);

+            s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);

         }

         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

@@ -2443,6 +2443,14 @@ static void vector_fmul_scalar_c(float *dst, const float *src, float mul,

         dst[i] = src[i] * mul;

 }

+static void vector_fmac_scalar_c(float *dst, const float *src, float mul,

+                                 int len)

+{

+    int i;

+    for (i = 0; i < len; i++)

+        dst[i] += src[i] * mul;

+}

+

 static void butterflies_float_c(float *restrict v1, float *restrict v2,

                                 int len)

 {

@@ -2978,6 +2986,7 @@ av_cold void dsputil_init(DSPContext* c, AVCodecContext *avctx)

     c->scalarproduct_float = scalarproduct_float_c;

     c->butterflies_float = butterflies_float_c;

     c->vector_fmul_scalar = vector_fmul_scalar_c;

+    c->vector_fmac_scalar = vector_fmac_scalar_c;

     c->shrink[0]= av_image_copy_plane;

     c->shrink[1]= ff_shrink22;

@@ -424,6 +424,17 @@ typedef struct DSPContext {

     void (*vector_fmul_scalar)(float *dst, const float *src, float mul,

                                int len);

     /**

+     * Multiply a vector of floats by a scalar float and add to

+     * destination vector.  Source and destination vectors must

+     * overlap exactly or not at all.

+     * @param dst result vector, 16-byte aligned

+     * @param src input vector, 16-byte aligned

+     * @param mul scalar value

+     * @param len length of vector, multiple of 4

+     */

+    void (*vector_fmac_scalar)(float *dst, const float *src, float mul,

+                               int len);

+    /**

      * Calculate the scalar product of two vectors of floats.

      * @param v1  first vector, 16-byte aligned

      * @param v2  second vector, 16-byte aligned

@@ -30,7 +30,7 @@

 #define LIBAVFILTER_VERSION_MAJOR  2

 #define LIBAVFILTER_VERSION_MINOR 43

-#define LIBAVFILTER_VERSION_MICRO  5

+#define LIBAVFILTER_VERSION_MICRO  6

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

                                                LIBAVFILTER_VERSION_MINOR, \

@@ -171,4 +171,5 @@ int avfilter_graph_send_command(AVFilterGraph *graph, const char *target, const

 int avfilter_graph_queue_command(AVFilterGraph *graph, const char *target, const char *cmd, const char *arg, int flags, double ts);

+

 #endif /* AVFILTER_AVFILTERGRAPH_H */

@@ -28,6 +28,7 @@

 #include "rawenc.h"

 typedef struct {

+    AVClass *av_class;

     int off;

     int channel_conf;

     int object_type;

@@ -967,6 +967,7 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,

     uint8_t* data = *buf;

     int isize = *buf_size;

     uint8_t* pkt_data = NULL;

+    uint8_t* newpktdata;

     int pkt_size = isize;

     int result = 0;

     int olen;

@@ -996,7 +997,12 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,

         zstream.avail_in = isize;

         do {

             pkt_size *= 3;

-            pkt_data = av_realloc(pkt_data, pkt_size);

+            newpktdata = av_realloc(pkt_data, pkt_size);

+            if (!newpktdata) {

+                inflateEnd(&zstream);

+                goto failed;

+            }

+            pkt_data = newpktdata;

             zstream.avail_out = pkt_size - zstream.total_out;

             zstream.next_out = pkt_data + zstream.total_out;

             if (pkt_data) {

@@ -1020,7 +1026,12 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,

         bzstream.avail_in = isize;

         do {

             pkt_size *= 3;

-            pkt_data = av_realloc(pkt_data, pkt_size);

+            newpktdata = av_realloc(pkt_data, pkt_size);

+            if (!newpktdata) {

+                BZ2_bzDecompressEnd(&bzstream);

+                goto failed;

+            }

+            pkt_data = newpktdata;

             bzstream.avail_out = pkt_size - bzstream.total_out_lo32;

             bzstream.next_out = pkt_data + bzstream.total_out_lo32;

             if (pkt_data) {

@@ -105,7 +105,7 @@ seektest(){

                  file=$(echo tests/data/$d/$file)

                  ;;

     esac

-    $target_exec $target_path/libavformat/seek-test $target_path/$file

+    run libavformat/seek-test $target_path/$file

 }

 mkdir -p "$outdir"