mirror of https://git.ffmpeg.org/ffmpeg.git
swr: change sample format representation so as to maintain the planer/packed distinction.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
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9d4a1aca88
commit
edbde52226
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@ -122,7 +122,7 @@ AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt,
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int flags)
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{
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AudioConvert *ctx;
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conv_func_type *f = fmt_pair_to_conv_functions[out_fmt + AV_SAMPLE_FMT_NB*in_fmt];
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conv_func_type *f = fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)];
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if (!f)
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return NULL;
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@ -27,6 +27,9 @@ void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSa
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double *tmp = av_malloc((len + TMP_EXTRA) * sizeof(double));
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int i;
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out_fmt = av_get_packed_sample_fmt(out_fmt);
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in_fmt = av_get_packed_sample_fmt( in_fmt);
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if(in_fmt == AV_SAMPLE_FMT_FLT || in_fmt == AV_SAMPLE_FMT_DBL){
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if(out_fmt == AV_SAMPLE_FMT_S32) scale = 1.0/(1L<<31);
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if(out_fmt == AV_SAMPLE_FMT_S16) scale = 1.0/(1L<<15);
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@ -227,8 +227,8 @@ static int auto_matrix(SwrContext *s)
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if(s->rematrix_volume < 0)
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maxcoef = -s->rematrix_volume;
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if(( s->out_sample_fmt < AV_SAMPLE_FMT_FLT
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|| s->int_sample_fmt < AV_SAMPLE_FMT_FLT) && maxcoef > 1.0){
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if(( av_get_packed_sample_fmt(s->out_sample_fmt) < AV_SAMPLE_FMT_FLT
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|| av_get_packed_sample_fmt(s->int_sample_fmt) < AV_SAMPLE_FMT_FLT) && maxcoef > 1.0){
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for(i=0; i<SWR_CH_MAX; i++)
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for(j=0; j<SWR_CH_MAX; j++){
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s->matrix[i][j] /= maxcoef;
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@ -273,10 +273,10 @@ int swri_rematrix_init(SwrContext *s){
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}
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void swri_sum2(enum AVSampleFormat format, void *dst, const void *src0, const void *src1, float coef0, float coef1, int len){
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if(format == AV_SAMPLE_FMT_FLT){
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if(format == AV_SAMPLE_FMT_FLTP){
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sum2_float((float *)dst, (const float *)src0, (const float *)src1, coef0, coef1, len);
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}else{
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av_assert1(format == AV_SAMPLE_FMT_S16);
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av_assert1(format == AV_SAMPLE_FMT_S16P);
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sum2_s16 ((int16_t*)dst, (const int16_t*)src0, (const int16_t*)src1, lrintf(coef0 * 32768), lrintf(coef1 * 32768), len);
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}
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}
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@ -295,7 +295,7 @@ int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mus
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case 1:
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in_i= s->matrix_ch[out_i][1];
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if(mustcopy || s->matrix[out_i][in_i]!=1.0){
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if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){
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if(s->int_sample_fmt == AV_SAMPLE_FMT_FLTP){
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copy_float((float *)out->ch[out_i], (const float *)in->ch[in_i], s->matrix [out_i][in_i], len);
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}else
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copy_s16 ((int16_t*)out->ch[out_i], (const int16_t*)in->ch[in_i], s->matrix32[out_i][in_i], len);
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@ -308,7 +308,7 @@ int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mus
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s->matrix[out_i][ s->matrix_ch[out_i][1] ], s->matrix[out_i][ s->matrix_ch[out_i][2] ], len);
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break;
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default:
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if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){
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if(s->int_sample_fmt == AV_SAMPLE_FMT_FLTP){
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for(i=0; i<len; i++){
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float v=0;
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for(j=0; j<s->matrix_ch[out_i][0]; j++){
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@ -131,19 +131,19 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
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/* normalize so that an uniform color remains the same */
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switch(c->format){
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case AV_SAMPLE_FMT_S16:
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case AV_SAMPLE_FMT_S16P:
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for(i=0;i<tap_count;i++)
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((int16_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
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break;
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case AV_SAMPLE_FMT_S32:
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case AV_SAMPLE_FMT_S32P:
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for(i=0;i<tap_count;i++)
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((int32_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
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break;
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case AV_SAMPLE_FMT_FLT:
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case AV_SAMPLE_FMT_FLTP:
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for(i=0;i<tap_count;i++)
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((float*)filter)[ph * tap_count + i] = tab[i] * scale / norm;
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break;
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case AV_SAMPLE_FMT_DBL:
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case AV_SAMPLE_FMT_DBLP:
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for(i=0;i<tap_count;i++)
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((double*)filter)[ph * tap_count + i] = tab[i] * scale / norm;
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break;
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@ -205,14 +205,14 @@ ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rat
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c->felem_size= av_get_bytes_per_sample(c->format);
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switch(c->format){
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case AV_SAMPLE_FMT_S16:
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case AV_SAMPLE_FMT_S16P:
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c->filter_shift = 15;
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break;
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case AV_SAMPLE_FMT_S32:
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case AV_SAMPLE_FMT_S32P:
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c->filter_shift = 30;
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break;
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case AV_SAMPLE_FMT_FLT:
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case AV_SAMPLE_FMT_DBL:
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case AV_SAMPLE_FMT_FLTP:
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case AV_SAMPLE_FMT_DBLP:
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c->filter_shift = 0;
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break;
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default:
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@ -359,10 +359,10 @@ int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, Aud
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int i, ret= -1;
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for(i=0; i<dst->ch_count; i++){
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if(c->format == AV_SAMPLE_FMT_S16) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_S32) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_FLT) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_DBL) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_S16P) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_S32P) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_FLTP) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_DBLP) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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}
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return ret;
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@ -52,8 +52,8 @@ static const AVOption options[]={
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{"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLT, PARAM},
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{"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLT, PARAM},
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{"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
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{"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
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{"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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@ -193,8 +193,6 @@ int swr_init(struct SwrContext *s){
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s-> in.planar= av_sample_fmt_is_planar(s-> in_sample_fmt);
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s->out.planar= av_sample_fmt_is_planar(s->out_sample_fmt);
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s-> in_sample_fmt= av_get_alt_sample_fmt(s-> in_sample_fmt, 0);
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s->out_sample_fmt= av_get_alt_sample_fmt(s->out_sample_fmt, 0);
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if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
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av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
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@ -206,14 +204,14 @@ int swr_init(struct SwrContext *s){
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}
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//FIXME should we allow/support using FLT on material that doesnt need it ?
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if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
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s->int_sample_fmt= AV_SAMPLE_FMT_S16;
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if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P || s->int_sample_fmt==AV_SAMPLE_FMT_S16P){
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s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
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}else
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s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
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s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
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&&s->int_sample_fmt != AV_SAMPLE_FMT_S32
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&&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
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&&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
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&&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP){
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av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
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return AVERROR(EINVAL);
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}
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@ -222,9 +220,9 @@ int swr_init(struct SwrContext *s){
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s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt);
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}else
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swri_resample_free(&s->resample);
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
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&& s->int_sample_fmt != AV_SAMPLE_FMT_S32
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&& s->int_sample_fmt != AV_SAMPLE_FMT_FLT
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_S32P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
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&& s->resample){
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av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt\n");
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return -1;
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@ -37,7 +37,7 @@ struct SwrContext {
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int log_level_offset; ///< logging level offset
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void *log_ctx; ///< parent logging context
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enum AVSampleFormat in_sample_fmt; ///< input sample format
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enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLT or AV_SAMPLE_FMT_S16)
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enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
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enum AVSampleFormat out_sample_fmt; ///< output sample format
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int64_t in_ch_layout; ///< input channel layout
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int64_t out_ch_layout; ///< output channel layout
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