/* * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) * * This file is part of libswresample * * libswresample is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * libswresample is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with libswresample; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/opt.h" #include "swresample_internal.h" #include "audioconvert.h" #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/internal.h" #include <float.h> #define ALIGN 32 #include "libavutil/ffversion.h" const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION; unsigned swresample_version(void) { av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100); return LIBSWRESAMPLE_VERSION_INT; } const char *swresample_configuration(void) { return FFMPEG_CONFIGURATION; } const char *swresample_license(void) { #define LICENSE_PREFIX "libswresample license: " return &LICENSE_PREFIX FFMPEG_LICENSE[sizeof(LICENSE_PREFIX) - 1]; } int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){ if(!s || s->in_convert) // s needs to be allocated but not initialized return AVERROR(EINVAL); s->channel_map = channel_map; return 0; } struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx){ if(!s) s= swr_alloc(); if(!s) return NULL; s->log_level_offset= log_offset; s->log_ctx= log_ctx; if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0) goto fail; if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0) goto fail; if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0) goto fail; if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0) goto fail; if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0) goto fail; if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0) goto fail; if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0) goto fail; if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0) goto fail; av_opt_set_int(s, "uch", 0, 0); return s; fail: av_log(s, AV_LOG_ERROR, "Failed to set option\n"); swr_free(&s); return NULL; } static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){ a->fmt = fmt; a->bps = av_get_bytes_per_sample(fmt); a->planar= av_sample_fmt_is_planar(fmt); if (a->ch_count == 1) a->planar = 1; } static void free_temp(AudioData *a){ av_free(a->data); memset(a, 0, sizeof(*a)); } static void clear_context(SwrContext *s){ s->in_buffer_index= 0; s->in_buffer_count= 0; s->resample_in_constraint= 0; memset(s->in.ch, 0, sizeof(s->in.ch)); memset(s->out.ch, 0, sizeof(s->out.ch)); free_temp(&s->postin); free_temp(&s->midbuf); free_temp(&s->preout); free_temp(&s->in_buffer); free_temp(&s->silence); free_temp(&s->drop_temp); free_temp(&s->dither.noise); free_temp(&s->dither.temp); swri_audio_convert_free(&s-> in_convert); swri_audio_convert_free(&s->out_convert); swri_audio_convert_free(&s->full_convert); swri_rematrix_free(s); s->delayed_samples_fixup = 0; s->flushed = 0; } av_cold void swr_free(SwrContext **ss){ SwrContext *s= *ss; if(s){ clear_context(s); if (s->resampler) s->resampler->free(&s->resample); } av_freep(ss); } av_cold void swr_close(SwrContext *s){ clear_context(s); } av_cold int swr_init(struct SwrContext *s){ int ret; char l1[1024], l2[1024]; clear_context(s); if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){ av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt); return AVERROR(EINVAL); } if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){ av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt); return AVERROR(EINVAL); } if(s-> in_sample_rate <= 0){ av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate); return AVERROR(EINVAL); } if(s->out_sample_rate <= 0){ av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate); return AVERROR(EINVAL); } s->out.ch_count = s-> user_out_ch_count; s-> in.ch_count = s-> user_in_ch_count; s->used_ch_count = s->user_used_ch_count; s-> in_ch_layout = s-> user_in_ch_layout; s->out_ch_layout = s->user_out_ch_layout; s->int_sample_fmt= s->user_int_sample_fmt; s->dither.method = s->user_dither_method; if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) { av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout); s->in_ch_layout = 0; } if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) { av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout); s->out_ch_layout = 0; } switch(s->engine){ #if CONFIG_LIBSOXR case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break; #endif case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; default: av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); return AVERROR(EINVAL); } if(!s->used_ch_count) s->used_ch_count= s->in.ch_count; if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){ av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n"); s-> in_ch_layout= 0; } if(!s-> in_ch_layout) s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count); if(!s->out_ch_layout) s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count); s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 || s->rematrix_custom; if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){ if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2 && av_get_bytes_per_sample(s->out_sample_fmt) <= 2){ s->int_sample_fmt= AV_SAMPLE_FMT_S16P; }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2 && !s->rematrix && s->out_sample_rate==s->in_sample_rate && !(s->flags & SWR_FLAG_RESAMPLE)){ s->int_sample_fmt= AV_SAMPLE_FMT_S16P; }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P && !s->rematrix && s->out_sample_rate == s->in_sample_rate && !(s->flags & SWR_FLAG_RESAMPLE) && s->engine != SWR_ENGINE_SOXR){ s->int_sample_fmt= AV_SAMPLE_FMT_S32P; }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){ s->int_sample_fmt= AV_SAMPLE_FMT_FLTP; }else{ s->int_sample_fmt= AV_SAMPLE_FMT_DBLP; } } av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt)); if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt)); return AVERROR(EINVAL); } set_audiodata_fmt(&s-> in, s-> in_sample_fmt); set_audiodata_fmt(&s->out, s->out_sample_fmt); if (s->firstpts_in_samples != AV_NOPTS_VALUE) { if (!s->async && s->min_compensation >= FLT_MAX/2) s->async = 1; s->firstpts = s->outpts = s->firstpts_in_samples * s->out_sample_rate; } else s->firstpts = AV_NOPTS_VALUE; if (s->async) { if (s->min_compensation >= FLT_MAX/2) s->min_compensation = 0.001; if (s->async > 1.0001) { s->max_soft_compensation = s->async / (double) s->in_sample_rate; } } if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ s->resample = s->resampler->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, s->filter_type, s->kaiser_beta, s->precision, s->cheby, s->exact_rational); if (!s->resample) { av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n"); return AVERROR(ENOMEM); } }else s->resampler->free(&s->resample); if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P && s->int_sample_fmt != AV_SAMPLE_FMT_S32P && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP && s->resample){ av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n"); ret = AVERROR(EINVAL); goto fail; } #define RSC 1 //FIXME finetune if(!s-> in.ch_count) s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout); if(!s->used_ch_count) s->used_ch_count= s->in.ch_count; if(!s->out.ch_count) s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout); if(!s-> in.ch_count){ av_assert0(!s->in_ch_layout); av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n"); ret = AVERROR(EINVAL); goto fail; } av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout); av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout); if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) { av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count); ret = AVERROR(EINVAL); goto fail; } if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) { av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count); ret = AVERROR(EINVAL); goto fail; } if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) { av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s " "but there is not enough information to do it\n", l1, l2); ret = AVERROR(EINVAL); goto fail; } av_assert0(s->used_ch_count); av_assert0(s->out.ch_count); s->resample_first= RSC*s->out.ch_count/s->used_ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0; s->in_buffer= s->in; s->silence = s->in; s->drop_temp= s->out; if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) goto fail; if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){ s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt, s-> in_sample_fmt, s-> in.ch_count, NULL, 0); return 0; } s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt, s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0); s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt, s->int_sample_fmt, s->out.ch_count, NULL, 0); if (!s->in_convert || !s->out_convert) { ret = AVERROR(ENOMEM); goto fail; } s->postin= s->in; s->preout= s->out; s->midbuf= s->in; if(s->channel_map){ s->postin.ch_count= s->midbuf.ch_count= s->used_ch_count; if(s->resample) s->in_buffer.ch_count= s->used_ch_count; } if(!s->resample_first){ s->midbuf.ch_count= s->out.ch_count; if(s->resample) s->in_buffer.ch_count = s->out.ch_count; } set_audiodata_fmt(&s->postin, s->int_sample_fmt); set_audiodata_fmt(&s->midbuf, s->int_sample_fmt); set_audiodata_fmt(&s->preout, s->int_sample_fmt); if(s->resample){ set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt); } av_assert0(!s->preout.count); s->dither.noise = s->preout; s->dither.temp = s->preout; if (s->dither.method > SWR_DITHER_NS) { s->dither.noise.bps = 4; s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP; s->dither.noise_scale = 1; } if(s->rematrix || s->dither.method) { ret = swri_rematrix_init(s); if (ret < 0) goto fail; } return 0; fail: swr_close(s); return ret; } int swri_realloc_audio(AudioData *a, int count){ int i, countb; AudioData old; if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count) return AVERROR(EINVAL); if(a->count >= count) return 0; count*=2; countb= FFALIGN(count*a->bps, ALIGN); old= *a; av_assert0(a->bps); av_assert0(a->ch_count); a->data= av_mallocz_array(countb, a->ch_count); if(!a->data) return AVERROR(ENOMEM); for(i=0; i<a->ch_count; i++){ a->ch[i]= a->data + i*(a->planar ? countb : a->bps); if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps); } if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps); av_freep(&old.data); a->count= count; return 1; } static void copy(AudioData *out, AudioData *in, int count){ av_assert0(out->planar == in->planar); av_assert0(out->bps == in->bps); av_assert0(out->ch_count == in->ch_count); if(out->planar){ int ch; for(ch=0; ch<out->ch_count; ch++) memcpy(out->ch[ch], in->ch[ch], count*out->bps); }else memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps); } static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ int i; if(!in_arg){ memset(out->ch, 0, sizeof(out->ch)); }else if(out->planar){ for(i=0; i<out->ch_count; i++) out->ch[i]= in_arg[i]; }else{ for(i=0; i<out->ch_count; i++) out->ch[i]= in_arg[0] + i*out->bps; } } static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ int i; if(out->planar){ for(i=0; i<out->ch_count; i++) in_arg[i]= out->ch[i]; }else{ in_arg[0]= out->ch[0]; } } /** * * out may be equal in. */ static void buf_set(AudioData *out, AudioData *in, int count){ int ch; if(in->planar){ for(ch=0; ch<out->ch_count; ch++) out->ch[ch]= in->ch[ch] + count*out->bps; }else{ for(ch=out->ch_count-1; ch>=0; ch--) out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps; } } /** * * @return number of samples output per channel */ static int resample(SwrContext *s, AudioData *out_param, int out_count, const AudioData * in_param, int in_count){ AudioData in, out, tmp; int ret_sum=0; int border=0; int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0; av_assert1(s->in_buffer.ch_count == in_param->ch_count); av_assert1(s->in_buffer.planar == in_param->planar); av_assert1(s->in_buffer.fmt == in_param->fmt); tmp=out=*out_param; in = *in_param; border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer, &in, in_count, &s->in_buffer_index, &s->in_buffer_count); if (border == INT_MAX) { return 0; } else if (border < 0) { return border; } else if (border) { buf_set(&in, &in, border); in_count -= border; s->resample_in_constraint = 0; } do{ int ret, size, consumed; if(!s->resample_in_constraint && s->in_buffer_count){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index); ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); out_count -= ret; ret_sum += ret; buf_set(&out, &out, ret); s->in_buffer_count -= consumed; s->in_buffer_index += consumed; if(!in_count) break; if(s->in_buffer_count <= border){ buf_set(&in, &in, -s->in_buffer_count); in_count += s->in_buffer_count; s->in_buffer_count=0; s->in_buffer_index=0; border = 0; } } if((s->flushed || in_count > padless) && !s->in_buffer_count){ s->in_buffer_index=0; ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed); out_count -= ret; ret_sum += ret; buf_set(&out, &out, ret); in_count -= consumed; buf_set(&in, &in, consumed); } //TODO is this check sane considering the advanced copy avoidance below size= s->in_buffer_index + s->in_buffer_count + in_count; if( size > s->in_buffer.count && s->in_buffer_count + in_count <= s->in_buffer_index){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index); copy(&s->in_buffer, &tmp, s->in_buffer_count); s->in_buffer_index=0; }else if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) return ret; if(in_count){ int count= in_count; if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2; buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); copy(&tmp, &in, /*in_*/count); s->in_buffer_count += count; in_count -= count; border += count; buf_set(&in, &in, count); s->resample_in_constraint= 0; if(s->in_buffer_count != count || in_count) continue; if (padless) { padless = 0; continue; } } break; }while(1); s->resample_in_constraint= !!out_count; return ret_sum; } static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, AudioData *in , int in_count){ AudioData *postin, *midbuf, *preout; int ret/*, in_max*/; AudioData preout_tmp, midbuf_tmp; if(s->full_convert){ av_assert0(!s->resample); swri_audio_convert(s->full_convert, out, in, in_count); return out_count; } // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps; // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count); if((ret=swri_realloc_audio(&s->postin, in_count))<0) return ret; if(s->resample_first){ av_assert0(s->midbuf.ch_count == s->used_ch_count); if((ret=swri_realloc_audio(&s->midbuf, out_count))<0) return ret; }else{ av_assert0(s->midbuf.ch_count == s->out.ch_count); if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) return ret; } if((ret=swri_realloc_audio(&s->preout, out_count))<0) return ret; postin= &s->postin; midbuf_tmp= s->midbuf; midbuf= &midbuf_tmp; preout_tmp= s->preout; preout= &preout_tmp; if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map) postin= in; if(s->resample_first ? !s->resample : !s->rematrix) midbuf= postin; if(s->resample_first ? !s->rematrix : !s->resample) preout= midbuf; if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){ if(preout==in){ out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though copy(out, in, out_count); return out_count; } else if(preout==postin) preout= midbuf= postin= out; else if(preout==midbuf) preout= midbuf= out; else preout= out; } if(in != postin){ swri_audio_convert(s->in_convert, postin, in, in_count); } if(s->resample_first){ if(postin != midbuf) out_count= resample(s, midbuf, out_count, postin, in_count); if(midbuf != preout) swri_rematrix(s, preout, midbuf, out_count, preout==out); }else{ if(postin != midbuf) swri_rematrix(s, midbuf, postin, in_count, midbuf==out); if(midbuf != preout) out_count= resample(s, preout, out_count, midbuf, in_count); } if(preout != out && out_count){ AudioData *conv_src = preout; if(s->dither.method){ int ch; int dither_count= FFMAX(out_count, 1<<16); if (preout == in) { conv_src = &s->dither.temp; if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) return ret; } if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) return ret; if(ret) for(ch=0; ch<s->dither.noise.ch_count; ch++) if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0) return ret; av_assert0(s->dither.noise.ch_count == preout->ch_count); if(s->dither.noise_pos + out_count > s->dither.noise.count) s->dither.noise_pos = 0; if (s->dither.method < SWR_DITHER_NS){ if (s->mix_2_1_simd) { int len1= out_count&~15; int off = len1 * preout->bps; if(len1) for(ch=0; ch<preout->ch_count; ch++) s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1); if(out_count != len1) for(ch=0; ch<preout->ch_count; ch++) s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off, s->native_one, 0, 0, out_count - len1); } else { for(ch=0; ch<preout->ch_count; ch++) s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count); } } else { switch(s->int_sample_fmt) { case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; } } s->dither.noise_pos += out_count; } //FIXME packed doesn't need more than 1 chan here! swri_audio_convert(s->out_convert, out, conv_src, out_count); } return out_count; } int swr_is_initialized(struct SwrContext *s) { return !!s->in_buffer.ch_count; } int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg [SWR_CH_MAX], int in_count){ AudioData * in= &s->in; AudioData *out= &s->out; int av_unused max_output; if (!swr_is_initialized(s)) { av_log(s, AV_LOG_ERROR, "Context has not been initialized\n"); return AVERROR(EINVAL); } #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1 max_output = swr_get_out_samples(s, in_count); #endif while(s->drop_output > 0){ int ret; uint8_t *tmp_arg[SWR_CH_MAX]; #define MAX_DROP_STEP 16384 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) return ret; reversefill_audiodata(&s->drop_temp, tmp_arg); s->drop_output *= -1; //FIXME find a less hackish solution ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter s->drop_output *= -1; in_count = 0; if(ret>0) { s->drop_output -= ret; if (!s->drop_output && !out_arg) return 0; continue; } av_assert0(s->drop_output); return 0; } if(!in_arg){ if(s->resample){ if (!s->flushed) s->resampler->flush(s); s->resample_in_constraint = 0; s->flushed = 1; }else if(!s->in_buffer_count){ return 0; } }else fill_audiodata(in , (void*)in_arg); fill_audiodata(out, out_arg); if(s->resample){ int ret = swr_convert_internal(s, out, out_count, in, in_count); if(ret>0 && !s->drop_output) s->outpts += ret * (int64_t)s->in_sample_rate; av_assert2(max_output < 0 || ret < 0 || ret <= max_output); return ret; }else{ AudioData tmp= *in; int ret2=0; int ret, size; size = FFMIN(out_count, s->in_buffer_count); if(size){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index); ret= swr_convert_internal(s, out, size, &tmp, size); if(ret<0) return ret; ret2= ret; s->in_buffer_count -= ret; s->in_buffer_index += ret; buf_set(out, out, ret); out_count -= ret; if(!s->in_buffer_count) s->in_buffer_index = 0; } if(in_count){ size= s->in_buffer_index + s->in_buffer_count + in_count - out_count; if(in_count > out_count) { //FIXME move after swr_convert_internal if( size > s->in_buffer.count && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index); copy(&s->in_buffer, &tmp, s->in_buffer_count); s->in_buffer_index=0; }else if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) return ret; } if(out_count){ size = FFMIN(in_count, out_count); ret= swr_convert_internal(s, out, size, in, size); if(ret<0) return ret; buf_set(in, in, ret); in_count -= ret; ret2 += ret; } if(in_count){ buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); copy(&tmp, in, in_count); s->in_buffer_count += in_count; } } if(ret2>0 && !s->drop_output) s->outpts += ret2 * (int64_t)s->in_sample_rate; av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output); return ret2; } } int swr_drop_output(struct SwrContext *s, int count){ const uint8_t *tmp_arg[SWR_CH_MAX]; s->drop_output += count; if(s->drop_output <= 0) return 0; av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count); return swr_convert(s, NULL, s->drop_output, tmp_arg, 0); } int swr_inject_silence(struct SwrContext *s, int count){ int ret, i; uint8_t *tmp_arg[SWR_CH_MAX]; if(count <= 0) return 0; #define MAX_SILENCE_STEP 16384 while (count > MAX_SILENCE_STEP) { if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) return ret; count -= MAX_SILENCE_STEP; } if((ret=swri_realloc_audio(&s->silence, count))<0) return ret; if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) { memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); } else memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); reversefill_audiodata(&s->silence, tmp_arg); av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count); ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count); return ret; } int64_t swr_get_delay(struct SwrContext *s, int64_t base){ if (s->resampler && s->resample){ return s->resampler->get_delay(s, base); }else{ return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; } } int swr_get_out_samples(struct SwrContext *s, int in_samples) { int64_t out_samples; if (in_samples < 0) return AVERROR(EINVAL); if (s->resampler && s->resample) { if (!s->resampler->get_out_samples) return AVERROR(ENOSYS); out_samples = s->resampler->get_out_samples(s, in_samples); } else { out_samples = s->in_buffer_count + in_samples; av_assert0(s->out_sample_rate == s->in_sample_rate); } if (out_samples > INT_MAX) return AVERROR(EINVAL); return out_samples; } int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ int ret; if (!s || compensation_distance < 0) return AVERROR(EINVAL); if (!compensation_distance && sample_delta) return AVERROR(EINVAL); if (!s->resample) { s->flags |= SWR_FLAG_RESAMPLE; ret = swr_init(s); if (ret < 0) return ret; } if (!s->resampler->set_compensation){ return AVERROR(EINVAL); }else{ return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); } } int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ if(pts == INT64_MIN) return s->outpts; if (s->firstpts == AV_NOPTS_VALUE) s->outpts = s->firstpts = pts; if(s->min_compensation >= FLT_MAX) { return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate)); } else { int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate; double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); if(fabs(fdelta) > s->min_compensation) { if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){ int ret; if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate); else ret = swr_drop_output (s, -delta / s-> in_sample_rate); if(ret<0){ av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta); } } else if(s->soft_compensation_duration && s->max_soft_compensation) { int duration = s->out_sample_rate * s->soft_compensation_duration; double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1); int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ; av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration); swr_set_compensation(s, comp, duration); } } return s->outpts; } }