diff options
Diffstat (limited to 'libswresample/swresample.c')
| -rw-r--r-- | libswresample/swresample.c | 930 |
1 files changed, 930 insertions, 0 deletions
diff --git a/libswresample/swresample.c b/libswresample/swresample.c new file mode 100644 index 0000000000..8e238998e9 --- /dev/null +++ b/libswresample/swresample.c @@ -0,0 +1,930 @@ +/* + * 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, "tsf", AV_SAMPLE_FMT_NONE, 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); + } + + 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; + + 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_planar_sample_fmt(s-> in_sample_fmt) <= AV_SAMPLE_FMT_S16P + && av_get_planar_sample_fmt(s->out_sample_fmt) <= AV_SAMPLE_FMT_S16P){ + s->int_sample_fmt= AV_SAMPLE_FMT_S16P; + }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) <= AV_SAMPLE_FMT_S16P + && !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->engine != SWR_ENGINE_SOXR){ + s->int_sample_fmt= AV_SAMPLE_FMT_S32P; + }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){ + 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_FLTP + &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ + av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is 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); + 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 s16/s32/flt/dbl\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->in.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(!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); + } + + if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) + goto fail; + + 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->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps); + } + if(!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 + len1, 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; + } +} |