Replace deprecated symbols SAMPLE_FMT_* with AV_SAMPLE_FMT_*, and enum

SampleFormat with AVSampleFormat.

Originally committed as revision 25730 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 11 insertions, 11 deletions

diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index 89e2d71e53..272831520d 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -47,7 +47,7 @@ struct ReSampleContext {
     /* channel convert */
     int input_channels, output_channels, filter_channels;
     AVAudioConvert *convert_ctx[2];
-    enum SampleFormat sample_fmt[2]; ///< input and output sample format
+    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
@@ -144,8 +144,8 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 
 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
                                         int output_rate, int input_rate,
-                                        enum SampleFormat sample_fmt_out,
-                                        enum SampleFormat sample_fmt_in,
+                                        enum AVSampleFormat sample_fmt_out,
+                                        enum AVSampleFormat sample_fmt_in,
                                         int filter_length, int log2_phase_count,
                                         int linear, double cutoff)
 {
@@ -178,8 +178,8 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
     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] != SAMPLE_FMT_S16) {
-        if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
+    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
+        if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
                                                          s->sample_fmt[0], 1, NULL, 0))) {
             av_log(s, AV_LOG_ERROR,
                    "Cannot convert %s sample format to s16 sample format\n",
@@ -189,9 +189,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
         }
     }
 
-    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
-                                                         SAMPLE_FMT_S16, 1, NULL, 0))) {
+                                                         AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
             av_log(s, AV_LOG_ERROR,
                    "Cannot convert s16 sample format to %s sample format\n",
                    av_get_sample_fmt_name(s->sample_fmt[1]));
@@ -224,7 +224,7 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 {
     return av_audio_resample_init(output_channels, input_channels,
                                   output_rate, input_rate,
-                                  SAMPLE_FMT_S16, SAMPLE_FMT_S16,
+                                  AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
                                   TAPS, 10, 0, 0.8);
 }
 #endif
@@ -246,7 +246,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         return nb_samples;
     }
 
-    if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
+    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
         int istride[1] = { s->sample_size[0] };
         int ostride[1] = { 2 };
         const void *ibuf[1] = { input };
@@ -276,7 +276,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
 
     lenout= 4*nb_samples * s->ratio + 16;
 
-    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         output_bak = output;
 
         if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
@@ -341,7 +341,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
     }
 
-    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
         int istride[1] = { 2 };
         int ostride[1] = { s->sample_size[1] };
         const void *ibuf[1] = { output };