@@ -86,6 +86,7 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)

         default:

           return -1;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -221,6 +221,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)

             return AVERROR_NOMEM;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -1364,5 +1364,6 @@ AVCodec ac3_encoder = {

     AC3_encode_frame,

     AC3_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 / AC-3"),

 };

@@ -698,6 +698,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)

     default:

         break;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -1599,6 +1600,7 @@ AVCodec name ## _encoder = {                    \

     adpcm_encode_frame,                         \

     adpcm_encode_close,                         \

     NULL,                                       \

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, \

     .long_name = NULL_IF_CONFIG_SMALL(long_name_), \

 };

 #else

@@ -30,6 +30,12 @@

  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/

  */

+static av_cold void adx_decode_init(AVCodecContext *avctx)

+{

+    avctx->sample_fmt = SAMPLE_FMT_S16;

+    return 0;

+}

+

 /* 18 bytes <-> 32 samples */

 static void adx_decode(short *out,const unsigned char *in,PREV *prev)

@@ -161,7 +167,7 @@ AVCodec adpcm_adx_decoder = {

     CODEC_TYPE_AUDIO,

     CODEC_ID_ADPCM_ADX,

     sizeof(ADXContext),

-    NULL,

+    adx_decode_init,

     NULL,

     NULL,

     adx_decode_frame,

@@ -190,5 +190,6 @@ AVCodec adpcm_adx_encoder = {

     adx_encode_frame,

     adx_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX"),

 };

@@ -594,6 +594,7 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)

     alac->numchannels = alac->avctx->channels;

     alac->bytespersample = (avctx->bits_per_sample / 8) * alac->numchannels;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -198,6 +198,7 @@ static av_cold int ape_decode_init(AVCodecContext * avctx)

     }

     dsputil_init(&s->dsp, avctx);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -1058,6 +1058,7 @@ static int atrac3_decode_init(AVCodecContext *avctx)

         return AVERROR(ENOMEM);

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -1178,6 +1178,8 @@ static int cook_decode_init(AVCodecContext *avctx)

         return -1;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

+

 #ifdef COOKDEBUG

     dump_cook_context(q);

 #endif

@@ -1253,6 +1253,7 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)

         avctx->channels = avctx->request_channels;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -154,6 +154,7 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx)

         break;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -305,6 +305,7 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx)

     cin->avctx = avctx;

     cin->initial_decode_frame = 1;

     cin->delta = 0;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -113,6 +113,7 @@ static av_cold int flac_decode_init(AVCodecContext * avctx)

         }

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -1485,5 +1485,6 @@ AVCodec flac_encoder = {

     flac_encode_close,

     NULL,

     .capabilities = CODEC_CAP_SMALL_LAST_FRAME,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),

 };

@@ -323,6 +323,9 @@ static av_cold int g726_init(AVCodecContext * avctx)

         return AVERROR(ENOMEM);

     avctx->coded_frame->key_frame = 1;

+    if (avctx->codec->decode)

+        avctx->sample_fmt = SAMPLE_FMT_S16;

+

     return 0;

 }

@@ -381,6 +384,7 @@ AVCodec adpcm_g726_encoder = {

     g726_encode_frame,

     g726_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),

 };

 #endif //CONFIG_ENCODERS

@@ -154,6 +154,7 @@ static av_cold int imc_decode_init(AVCodecContext * avctx)

     ff_fft_init(&q->fft, 7, 1);

     dsputil_init(&q->dsp, avctx);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -119,6 +119,7 @@ static av_cold int a52_decode_init(AVCodecContext *avctx)

         avctx->channels = avctx->request_channels;

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -134,6 +134,7 @@ static void amr_decode_fix_avctx(AVCodecContext * avctx)

     }

     avctx->frame_size = 160 * is_amr_wb;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

 }

 #ifdef CONFIG_LIBAMR_NB_FIXED

@@ -516,6 +517,7 @@ AVCodec libamr_nb_encoder =

     amr_nb_encode_frame,

     amr_nb_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("libamr-nb Adaptive Multi-Rate (AMR) Narrow-Band"),

 };

@@ -710,6 +712,7 @@ AVCodec libamr_wb_encoder =

     amr_wb_encode_frame,

     amr_wb_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("libamr-wb Adaptive Multi-Rate (AMR) Wide-Band"),

 };

@@ -151,5 +151,6 @@ AVCodec libfaac_encoder = {

     Faac_encode_init,

     Faac_encode_frame,

     Faac_encode_close,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Codec)"),

 };

@@ -313,6 +313,7 @@ static av_cold int faac_decode_init(AVCodecContext *avctx)

     if(!s->init && avctx->channels > 0)

         channel_setup(avctx);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -48,6 +48,8 @@ static av_cold int libgsm_init(AVCodecContext *avctx) {

         if(!avctx->sample_rate)

             avctx->sample_rate= 8000;

+

+        avctx->sample_fmt = SAMPLE_FMT_S16;

     }else{

         if (avctx->sample_rate != 8000) {

             av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",

@@ -117,6 +119,7 @@ AVCodec libgsm_encoder = {

     libgsm_init,

     libgsm_encode_frame,

     libgsm_close,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),

 };

@@ -128,6 +131,7 @@ AVCodec libgsm_ms_encoder = {

     libgsm_init,

     libgsm_encode_frame,

     libgsm_close,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),

 };

@@ -218,5 +218,6 @@ AVCodec libmp3lame_encoder = {

     MP3lame_encode_frame,

     MP3lame_encode_close,

     .capabilities= CODEC_CAP_DELAY,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),

 };

@@ -217,5 +217,6 @@ AVCodec libvorbis_encoder = {

     oggvorbis_encode_frame,

     oggvorbis_encode_close,

     .capabilities= CODEC_CAP_DELAY,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),

 } ;

@@ -396,6 +396,7 @@ static av_cold int mace_decode_init(AVCodecContext * avctx)

 {

     if (avctx->channels > 2)

         return -1;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -336,6 +336,7 @@ static av_cold int mlp_decode_init(AVCodecContext *avctx)

     m->avctx = avctx;

     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)

         m->substream[substr].lossless_check_data = 0xffffffff;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -108,6 +108,7 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx)

         }

     }

     vlc_initialized = 1;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -177,6 +177,7 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx)

                  &mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_STATIC);

     }

     vlc_initialized = 1;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -796,6 +796,7 @@ AVCodec mp2_encoder = {

     MPA_encode_frame,

     MPA_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

 };

@@ -149,6 +149,7 @@ static av_cold int decode_init(AVCodecContext * avctx) {

     if (!sine_window[0])

         ff_sine_window_init(sine_window, 128);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -553,7 +553,7 @@ static int pcm_decode_frame(AVCodecContext *avctx,

 }

 #ifdef CONFIG_ENCODERS

-#define PCM_ENCODER(id,name,long_name_)         \

+#define PCM_ENCODER(id,sample_fmt_,name,long_name_) \

 AVCodec name ## _encoder = {                    \

     #name,                                      \

     CODEC_TYPE_AUDIO,                           \

@@ -563,10 +563,11 @@ AVCodec name ## _encoder = {                    \

     pcm_encode_frame,                           \

     pcm_encode_close,                           \

     NULL,                                       \

+    .sample_fmts = (enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \

     .long_name = NULL_IF_CONFIG_SMALL(long_name_), \

 };

 #else

-#define PCM_ENCODER(id,name,long_name_)

+#define PCM_ENCODER(id,sample_fmt_,name,long_name_)

 #endif

 #ifdef CONFIG_DECODERS

@@ -586,28 +587,28 @@ AVCodec name ## _decoder = {                    \

 #define PCM_DECODER(id,name,long_name_)

 #endif

-#define PCM_CODEC(id, name, long_name_)         \

-    PCM_ENCODER(id,name,long_name_) PCM_DECODER(id,name,long_name_)

+#define PCM_CODEC(id, sample_fmt_, name, long_name_)         \

+    PCM_ENCODER(id,sample_fmt_,name,long_name_) PCM_DECODER(id,name,long_name_)

 /* Note: Do not forget to add new entries to the Makefile as well. */

-PCM_CODEC  (CODEC_ID_PCM_ALAW, pcm_alaw, "A-law PCM");

-PCM_CODEC  (CODEC_ID_PCM_DVD, pcm_dvd, "signed 16|20|24-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_F32BE, pcm_f32be, "32-bit floating point big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_MULAW, pcm_mulaw, "mu-law PCM");

-PCM_CODEC  (CODEC_ID_PCM_S8, pcm_s8, "signed 8-bit PCM");

-PCM_CODEC  (CODEC_ID_PCM_S16BE, pcm_s16be, "signed 16-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_S16LE, pcm_s16le, "signed 16-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_ALAW,  SAMPLE_FMT_S16, pcm_alaw, "A-law PCM");

+PCM_CODEC  (CODEC_ID_PCM_DVD,   SAMPLE_FMT_S16, pcm_dvd, "signed 16|20|24-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_F32BE, SAMPLE_FMT_FLT, pcm_f32be, "32-bit floating point big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_MULAW, SAMPLE_FMT_S16, pcm_mulaw, "mu-law PCM");

+PCM_CODEC  (CODEC_ID_PCM_S8,    SAMPLE_FMT_S16, pcm_s8, "signed 8-bit PCM");

+PCM_CODEC  (CODEC_ID_PCM_S16BE, SAMPLE_FMT_S16, pcm_s16be, "signed 16-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_S16LE, SAMPLE_FMT_S16, pcm_s16le, "signed 16-bit little-endian PCM");

 PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, pcm_s16le_planar, "16-bit little-endian planar PCM");

-PCM_CODEC  (CODEC_ID_PCM_S24BE, pcm_s24be, "signed 24-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_S24DAUD, pcm_s24daud, "D-Cinema audio signed 24-bit PCM");

-PCM_CODEC  (CODEC_ID_PCM_S24LE, pcm_s24le, "signed 24-bit little-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_S32BE, pcm_s32be, "signed 32-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_S32LE, pcm_s32le, "signed 32-bit little-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U8, pcm_u8, "unsigned 8-bit PCM");

-PCM_CODEC  (CODEC_ID_PCM_U16BE, pcm_u16be, "unsigned 16-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U16LE, pcm_u16le, "unsigned 16-bit little-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U24BE, pcm_u24be, "unsigned 24-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U24LE, pcm_u24le, "unsigned 24-bit little-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U32BE, pcm_u32be, "unsigned 32-bit big-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_U32LE, pcm_u32le, "unsigned 32-bit little-endian PCM");

-PCM_CODEC  (CODEC_ID_PCM_ZORK, pcm_zork, "Zork PCM");

+PCM_CODEC  (CODEC_ID_PCM_S24BE, SAMPLE_FMT_S16, pcm_s24be, "signed 24-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_S24DAUD, SAMPLE_FMT_S16,  pcm_s24daud, "D-Cinema audio signed 24-bit PCM");

+PCM_CODEC  (CODEC_ID_PCM_S24LE, SAMPLE_FMT_S16, pcm_s24le, "signed 24-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_S32BE, SAMPLE_FMT_S16, pcm_s32be, "signed 32-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_S32LE, SAMPLE_FMT_S16, pcm_s32le, "signed 32-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U8,    SAMPLE_FMT_S16, pcm_u8, "unsigned 8-bit PCM");

+PCM_CODEC  (CODEC_ID_PCM_U16BE, SAMPLE_FMT_S16, pcm_u16be, "unsigned 16-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U16LE, SAMPLE_FMT_S16, pcm_u16le, "unsigned 16-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U24BE, SAMPLE_FMT_S16, pcm_u24be, "unsigned 24-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U24LE, SAMPLE_FMT_S16, pcm_u24le, "unsigned 24-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U32BE, SAMPLE_FMT_S16, pcm_u32be, "unsigned 32-bit big-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_U32LE, SAMPLE_FMT_S16, pcm_u32le, "unsigned 32-bit little-endian PCM");

+PCM_CODEC  (CODEC_ID_PCM_ZORK,  SAMPLE_FMT_S16, pcm_zork, "Zork PCM");

@@ -1931,6 +1931,8 @@ static int qdm2_decode_init(AVCodecContext *avctx)

     qdm2_init(s);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

+

 //    dump_context(s);

     return 0;

 }

@@ -58,6 +58,7 @@ static int ra144_decode_init(AVCodecContext * avctx)

     ractx->lpc_coef[0] = ractx->lpc_tables[0];

     ractx->lpc_coef[1] = ractx->lpc_tables[1];

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -42,6 +42,12 @@ typedef struct {

     float gain_block[10];  ///< Gain data of four blocks (spec: GSTATE)

 } RA288Context;

+static av_cold int ra288_decode_init(AVCodecContext *avctx)

+{

+    avctx->sample_fmt = SAMPLE_FMT_S16;

+    return 0;

+}

+

 static inline float scalar_product_float(const float * v1, const float * v2,

                                          int size)

 {

@@ -258,7 +264,7 @@ AVCodec ra_288_decoder =

     CODEC_TYPE_AUDIO,

     CODEC_ID_RA_288,

     sizeof(RA288Context),

-    NULL,

+    ra288_decode_init,

     NULL,

     NULL,

     ra288_decode_frame,

@@ -174,5 +174,6 @@ AVCodec roq_dpcm_encoder = {

     roq_dpcm_encode_frame,

     roq_dpcm_encode_close,

     NULL,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),

 };

@@ -104,6 +104,7 @@ static av_cold int shorten_decode_init(AVCodecContext * avctx)

 {

     ShortenContext *s = avctx->priv_data;

     s->avctx = avctx;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -558,6 +558,7 @@ static av_cold int decode_end(AVCodecContext *avctx)

 static av_cold int smka_decode_init(AVCodecContext *avctx)

 {

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -828,6 +828,7 @@ static av_cold int sonic_decode_init(AVCodecContext *avctx)

     }

     s->int_samples = av_mallocz(4* s->frame_size);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -54,6 +54,7 @@ static av_cold int truespeech_decode_init(AVCodecContext * avctx)

 {

 //    TSContext *c = avctx->priv_data;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -446,6 +446,7 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)

     s->channels = avctx->channels;

     s->bits = avctx->bits_per_sample;

     s->block_align = avctx->block_align;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     av_log(s->avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, block align = %d, sample rate = %d\n",

             s->channels, s->bits, s->block_align, avctx->sample_rate);

@@ -971,6 +971,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext) {

     avccontext->channels = vc->audio_channels;

     avccontext->sample_rate = vc->audio_samplerate;

     avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1])>>2;

+    avccontext->sample_fmt = SAMPLE_FMT_S16;

     return 0 ;

 }

@@ -1084,5 +1084,6 @@ AVCodec vorbis_encoder = {

     vorbis_encode_frame,

     vorbis_encode_close,

     .capabilities= CODEC_CAP_DELAY,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),

 };

@@ -360,6 +360,7 @@ static av_cold int wavpack_decode_init(AVCodecContext *avctx)

     s->avctx = avctx;

     s->stereo = (avctx->channels == 2);

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -126,6 +126,7 @@ static int wma_decode_init(AVCodecContext * avctx)

         wma_lsp_to_curve_init(s, s->frame_len);

     }

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }

@@ -387,6 +387,7 @@ AVCodec wmav1_encoder =

     encode_init,

     encode_superframe,

     ff_wma_end,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),

 };

@@ -399,5 +400,6 @@ AVCodec wmav2_encoder =

     encode_init,

     encode_superframe,

     ff_wma_end,

+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},

     .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),

 };

@@ -40,6 +40,7 @@ static av_cold int ws_snd_decode_init(AVCodecContext * avctx)

 {

 //    WSSNDContext *c = avctx->priv_data;

+    avctx->sample_fmt = SAMPLE_FMT_S16;

     return 0;

 }