From b2fe6756e34d1316d0fa799e8a5ace993059c407 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 31 Oct 2012 15:40:12 -0400 Subject: lavr: add option for dithering during sample format conversion to s16 --- libavresample/dither.c | 423 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 423 insertions(+) create mode 100644 libavresample/dither.c (limited to 'libavresample/dither.c') diff --git a/libavresample/dither.c b/libavresample/dither.c new file mode 100644 index 0000000000..9c1e1c1101 --- /dev/null +++ b/libavresample/dither.c @@ -0,0 +1,423 @@ +/* + * Copyright (c) 2012 Justin Ruggles + * + * Triangular with Noise Shaping is based on opusfile. + * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * Dithered Audio Sample Quantization + * + * Converts from dbl, flt, or s32 to s16 using dithering. + */ + +#include +#include + +#include "libavutil/common.h" +#include "libavutil/lfg.h" +#include "libavutil/mem.h" +#include "libavutil/samplefmt.h" +#include "audio_convert.h" +#include "dither.h" +#include "internal.h" + +typedef struct DitherState { + int mute; + unsigned int seed; + AVLFG lfg; + float *noise_buf; + int noise_buf_size; + int noise_buf_ptr; + float dither_a[4]; + float dither_b[4]; +} DitherState; + +struct DitherContext { + DitherDSPContext ddsp; + enum AVResampleDitherMethod method; + + int mute_dither_threshold; // threshold for disabling dither + int mute_reset_threshold; // threshold for resetting noise shaping + const float *ns_coef_b; // noise shaping coeffs + const float *ns_coef_a; // noise shaping coeffs + + int channels; + DitherState *state; // dither states for each channel + + AudioData *flt_data; // input data in fltp + AudioData *s16_data; // dithered output in s16p + AudioConvert *ac_in; // converter for input to fltp + AudioConvert *ac_out; // converter for s16p to s16 (if needed) + + void (*quantize)(int16_t *dst, const float *src, float *dither, int len); + int samples_align; +}; + +/* mute threshold, in seconds */ +#define MUTE_THRESHOLD_SEC 0.000333 + +/* scale factor for 16-bit output. + The signal is attenuated slightly to avoid clipping */ +#define S16_SCALE 32753.0f + +/* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */ +#define LFG_SCALE (1.0f / (2.0f * INT32_MAX)) + +/* noise shaping coefficients */ + +static const float ns_48_coef_b[4] = { + 2.2374f, -0.7339f, -0.1251f, -0.6033f +}; + +static const float ns_48_coef_a[4] = { + 0.9030f, 0.0116f, -0.5853f, -0.2571f +}; + +static const float ns_44_coef_b[4] = { + 2.2061f, -0.4707f, -0.2534f, -0.6213f +}; + +static const float ns_44_coef_a[4] = { + 1.0587f, 0.0676f, -0.6054f, -0.2738f +}; + +static void dither_int_to_float_rectangular_c(float *dst, int *src, int len) +{ + int i; + for (i = 0; i < len; i++) + dst[i] = src[i] * LFG_SCALE; +} + +static void dither_int_to_float_triangular_c(float *dst, int *src0, int len) +{ + int i; + int *src1 = src0 + len; + + for (i = 0; i < len; i++) { + float r = src0[i] * LFG_SCALE; + r += src1[i] * LFG_SCALE; + dst[i] = r; + } +} + +static void quantize_c(int16_t *dst, const float *src, float *dither, int len) +{ + int i; + for (i = 0; i < len; i++) + dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i])); +} + +#define SQRT_1_6 0.40824829046386301723f + +static void dither_highpass_filter(float *src, int len) +{ + int i; + + /* filter is from libswresample in FFmpeg */ + for (i = 0; i < len - 2; i++) + src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6; +} + +static int generate_dither_noise(DitherContext *c, DitherState *state, + int min_samples) +{ + int i; + int nb_samples = FFALIGN(min_samples, 16) + 16; + int buf_samples = nb_samples * + (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2); + unsigned int *noise_buf_ui; + + av_freep(&state->noise_buf); + state->noise_buf_size = state->noise_buf_ptr = 0; + + state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf)); + if (!state->noise_buf) + return AVERROR(ENOMEM); + state->noise_buf_size = FFALIGN(min_samples, 16); + noise_buf_ui = (unsigned int *)state->noise_buf; + + av_lfg_init(&state->lfg, state->seed); + for (i = 0; i < buf_samples; i++) + noise_buf_ui[i] = av_lfg_get(&state->lfg); + + c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples); + + if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP) + dither_highpass_filter(state->noise_buf, nb_samples); + + return 0; +} + +static void quantize_triangular_ns(DitherContext *c, DitherState *state, + int16_t *dst, const float *src, + int nb_samples) +{ + int i, j; + float *dither = &state->noise_buf[state->noise_buf_ptr]; + + if (state->mute > c->mute_reset_threshold) + memset(state->dither_a, 0, sizeof(state->dither_a)); + + for (i = 0; i < nb_samples; i++) { + float err = 0; + float sample = src[i] * S16_SCALE; + + for (j = 0; j < 4; j++) { + err += c->ns_coef_b[j] * state->dither_b[j] - + c->ns_coef_a[j] * state->dither_a[j]; + } + for (j = 3; j > 0; j--) { + state->dither_a[j] = state->dither_a[j - 1]; + state->dither_b[j] = state->dither_b[j - 1]; + } + state->dither_a[0] = err; + sample -= err; + + if (state->mute > c->mute_dither_threshold) { + dst[i] = av_clip_int16(lrintf(sample)); + state->dither_b[0] = 0; + } else { + dst[i] = av_clip_int16(lrintf(sample + dither[i])); + state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f); + } + + state->mute++; + if (src[i]) + state->mute = 0; + } +} + +static int convert_samples(DitherContext *c, int16_t **dst, float * const *src, + int channels, int nb_samples) +{ + int ch, ret; + int aligned_samples = FFALIGN(nb_samples, 16); + + for (ch = 0; ch < channels; ch++) { + DitherState *state = &c->state[ch]; + + if (state->noise_buf_size < aligned_samples) { + ret = generate_dither_noise(c, state, nb_samples); + if (ret < 0) + return ret; + } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) { + state->noise_buf_ptr = 0; + } + + if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { + quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples); + } else { + c->quantize(dst[ch], src[ch], + &state->noise_buf[state->noise_buf_ptr], + FFALIGN(nb_samples, c->samples_align)); + } + + state->noise_buf_ptr += aligned_samples; + } + + return 0; +} + +int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src) +{ + int ret; + AudioData *flt_data; + + /* output directly to dst if it is planar */ + if (dst->sample_fmt == AV_SAMPLE_FMT_S16P) + c->s16_data = dst; + else { + /* make sure s16_data is large enough for the output */ + ret = ff_audio_data_realloc(c->s16_data, src->nb_samples); + if (ret < 0) + return ret; + } + + if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) { + /* make sure flt_data is large enough for the input */ + ret = ff_audio_data_realloc(c->flt_data, src->nb_samples); + if (ret < 0) + return ret; + flt_data = c->flt_data; + + /* convert input samples to fltp and scale to s16 range */ + ret = ff_audio_convert(c->ac_in, flt_data, src); + if (ret < 0) + return ret; + } else { + flt_data = src; + } + + /* check alignment and padding constraints */ + if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) { + int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align); + int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align); + int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align); + + if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) { + c->quantize = c->ddsp.quantize; + c->samples_align = c->ddsp.samples_align; + } else { + c->quantize = quantize_c; + c->samples_align = 1; + } + } + + ret = convert_samples(c, (int16_t **)c->s16_data->data, + (float * const *)flt_data->data, src->channels, + src->nb_samples); + if (ret < 0) + return ret; + + c->s16_data->nb_samples = src->nb_samples; + + /* interleave output to dst if needed */ + if (dst->sample_fmt == AV_SAMPLE_FMT_S16) { + ret = ff_audio_convert(c->ac_out, dst, c->s16_data); + if (ret < 0) + return ret; + } else + c->s16_data = NULL; + + return 0; +} + +void ff_dither_free(DitherContext **cp) +{ + DitherContext *c = *cp; + int ch; + + if (!c) + return; + ff_audio_data_free(&c->flt_data); + ff_audio_data_free(&c->s16_data); + ff_audio_convert_free(&c->ac_in); + ff_audio_convert_free(&c->ac_out); + for (ch = 0; ch < c->channels; ch++) + av_free(c->state[ch].noise_buf); + av_free(c->state); + av_freep(cp); +} + +static void dither_init(DitherDSPContext *ddsp, + enum AVResampleDitherMethod method) +{ + ddsp->quantize = quantize_c; + ddsp->ptr_align = 1; + ddsp->samples_align = 1; + + if (method == AV_RESAMPLE_DITHER_RECTANGULAR) + ddsp->dither_int_to_float = dither_int_to_float_rectangular_c; + else + ddsp->dither_int_to_float = dither_int_to_float_triangular_c; +} + +DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, + enum AVSampleFormat out_fmt, + enum AVSampleFormat in_fmt, + int channels, int sample_rate) +{ + AVLFG seed_gen; + DitherContext *c; + int ch; + + if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 || + av_get_bytes_per_sample(in_fmt) <= 2) { + av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n", + av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt)); + return NULL; + } + + c = av_mallocz(sizeof(*c)); + if (!c) + return NULL; + + if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS && + sample_rate != 48000 && sample_rate != 44100) { + av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz " + "for triangular_ns dither. using triangular_hp instead.\n"); + avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP; + } + c->method = avr->dither_method; + dither_init(&c->ddsp, c->method); + + if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { + if (sample_rate == 48000) { + c->ns_coef_b = ns_48_coef_b; + c->ns_coef_a = ns_48_coef_a; + } else { + c->ns_coef_b = ns_44_coef_b; + c->ns_coef_a = ns_44_coef_a; + } + } + + /* Either s16 or s16p output format is allowed, but s16p is used + internally, so we need to use a temp buffer and interleave if the output + format is s16 */ + if (out_fmt != AV_SAMPLE_FMT_S16P) { + c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P, + "dither s16 buffer"); + if (!c->s16_data) + goto fail; + + c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P, + channels, sample_rate); + if (!c->ac_out) + goto fail; + } + + if (in_fmt != AV_SAMPLE_FMT_FLTP) { + c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP, + "dither flt buffer"); + if (!c->flt_data) + goto fail; + + c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt, + channels, sample_rate); + if (!c->ac_in) + goto fail; + } + + c->state = av_mallocz(channels * sizeof(*c->state)); + if (!c->state) + goto fail; + c->channels = channels; + + /* calculate thresholds for turning off dithering during periods of + silence to avoid replacing digital silence with quiet dither noise */ + c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC); + c->mute_reset_threshold = c->mute_dither_threshold * 4; + + /* initialize dither states */ + av_lfg_init(&seed_gen, 0xC0FFEE); + for (ch = 0; ch < channels; ch++) { + DitherState *state = &c->state[ch]; + state->mute = c->mute_reset_threshold + 1; + state->seed = av_lfg_get(&seed_gen); + generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2)); + } + + return c; + +fail: + ff_dither_free(&c); + return NULL; +} -- cgit v1.2.3