cosmetics: Fix crazy formatting in resample.

libavcodec/resample.c

@@ -39,7 +39,9 @@ static const char *context_to_name(void *ptr)
 }
 
 static const AVOption options[] = {{NULL}};
-static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };
+static const AVClass audioresample_context_class = {
+    "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
+};
 
 struct ReSampleContext {
     struct AVResampleContext *resample_context;
@@ -50,9 +52,9 @@ struct ReSampleContext {
     int input_channels, output_channels, filter_channels;
     AVAudioConvert *convert_ctx[2];
     enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
-    unsigned sample_size[2];         ///< size of one sample in sample_fmt
-    short *buffer[2];                ///< buffers used for conversion to S16
-    unsigned buffer_size[2];         ///< sizes of allocated buffers
+    unsigned sample_size[2];           ///< size of one sample in sample_fmt
+    short *buffer[2];                  ///< buffers used for conversion to S16
+    unsigned buffer_size[2];           ///< sizes of allocated buffers
 };
 
 /* n1: number of samples */
@@ -131,17 +133,17 @@ static void interleave(short *output, short **input, int channels, int samples)
 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 {
     int i;
-    short l,r;
-
-    for(i=0;i<n;i++) {
-      l=*input1++;
-      r=*input2++;
-      *output++ = l;           /* left */
-      *output++ = (l/2)+(r/2); /* center */
-      *output++ = r;           /* right */
-      *output++ = 0;           /* left surround */
-      *output++ = 0;           /* right surroud */
-      *output++ = 0;           /* low freq */
+    short l, r;
+
+    for (i = 0; i < n; i++) {
+        l = *input1++;
+        r = *input2++;
+        *output++ = l;                  /* left */
+        *output++ = (l / 2) + (r / 2);  /* center */
+        *output++ = r;                  /* right */
+        *output++ = 0;                  /* left surround */
+        *output++ = 0;                  /* right surroud */
+        *output++ = 0;                  /* low freq */
     }
 }
 
@@ -154,27 +156,25 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
 {
     ReSampleContext *s;
 
-    if (input_channels > MAX_CHANNELS)
-      {
+    if (input_channels > MAX_CHANNELS) {
         av_log(NULL, AV_LOG_ERROR,
                "Resampling with input channels greater than %d is unsupported.\n",
                MAX_CHANNELS);
         return NULL;
-      }
-    if (  output_channels > 2 &&
+    }
+    if (output_channels > 2 &&
         !(output_channels == 6 && input_channels == 2) &&
-          output_channels != input_channels) {
+        output_channels != input_channels) {
         av_log(NULL, AV_LOG_ERROR,
                "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
         return NULL;
     }
 
     s = av_mallocz(sizeof(ReSampleContext));
-    if (!s)
-      {
+    if (!s) {
         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
         return NULL;
-      }
+    }
 
     s->ratio = (float)output_rate / (float)input_rate;
 
@@ -185,10 +185,10 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;
 
-    s->sample_fmt [0] = sample_fmt_in;
-    s->sample_fmt [1] = sample_fmt_out;
-    s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3;
-    s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3;
+    s->sample_fmt[0]  = sample_fmt_in;
+    s->sample_fmt[1]  = sample_fmt_out;
+    s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0]) >> 3;
+    s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;
 
     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
@@ -214,8 +214,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
     }
 
 #define TAPS 16
-    s->resample_context= av_resample_init(output_rate, input_rate,
-                         filter_length, log2_phase_count, linear, cutoff);
+    s->resample_context = av_resample_init(output_rate, input_rate,
+                                           filter_length, log2_phase_count,
+                                           linear, cutoff);
 
     *(const AVClass**)s->resample_context = &audioresample_context_class;
 
@@ -244,7 +245,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         int ostride[1] = { 2 };
         const void *ibuf[1] = { input };
         void       *obuf[1];
-        unsigned input_size = nb_samples*s->input_channels*2;
+        unsigned input_size = nb_samples * s->input_channels * 2;
 
         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
             av_free(s->buffer[0]);
@@ -259,15 +260,16 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         obuf[0] = s->buffer[0];
 
         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
-                             ibuf, istride, nb_samples*s->input_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
+                             ibuf, istride, nb_samples * s->input_channels) < 0) {
+            av_log(s->resample_context, AV_LOG_ERROR,
+                   "Audio sample format conversion failed\n");
             return 0;
         }
 
-        input  = s->buffer[0];
+        input = s->buffer[0];
     }
 
-    lenout= 4*nb_samples * s->ratio + 16;
+    lenout = 4 * nb_samples * s->ratio + 16;
 
     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         output_bak = output;
@@ -286,20 +288,19 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
     }
 
     /* XXX: move those malloc to resample init code */
-    for(i=0; i<s->filter_channels; i++){
-        bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+    for (i = 0; i < s->filter_channels; i++) {
+        bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
         buftmp2[i] = bufin[i] + s->temp_len;
         bufout[i] = av_malloc(lenout * sizeof(short));
     }
 
-    if (s->input_channels == 2 &&
-        s->output_channels == 1) {
+    if (s->input_channels == 2 && s->output_channels == 1) {
         buftmp3[0] = output;
         stereo_to_mono(buftmp2[0], input, nb_samples);
     } else if (s->output_channels >= 2 && s->input_channels == 1) {
         buftmp3[0] = bufout[0];
-        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+        memcpy(buftmp2[0], input, nb_samples * sizeof(short));
     } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
         for (i = 0; i < s->input_channels; i++) {
             buftmp3[i] = bufout[i];
@@ -307,21 +308,22 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         deinterleave(buftmp2, input, s->input_channels, nb_samples);
     } else {
         buftmp3[0] = output;
-        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+        memcpy(buftmp2[0], input, nb_samples * sizeof(short));
     }
 
     nb_samples += s->temp_len;
 
     /* resample each channel */
     nb_samples1 = 0; /* avoid warning */
-    for(i=0;i<s->filter_channels;i++) {
+    for (i = 0; i < s->filter_channels; i++) {
         int consumed;
-        int is_last= i+1 == s->filter_channels;
+        int is_last = i + 1 == s->filter_channels;
 
-        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
-        s->temp_len= nb_samples - consumed;
-        s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
-        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
+        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
+                                  &consumed, nb_samples, lenout, is_last);
+        s->temp_len = nb_samples - consumed;
+        s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
+        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
     }
 
     if (s->output_channels == 2 && s->input_channels == 1) {
@@ -339,8 +341,9 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         void       *obuf[1] = { output_bak };
 
         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
-                             ibuf, istride, nb_samples1*s->output_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
+                             ibuf, istride, nb_samples1 * s->output_channels) < 0) {
+            av_log(s->resample_context, AV_LOG_ERROR,
+                   "Audio sample format convertion failed\n");
             return 0;
         }
     }