/* GStreamer * Copyright (C) 2007 Sebastian Dröge * (C) 2015 Wim Taymans * * gstaudioquantize.c: quantizes audio to the target format and optionally * applies dithering and noise shaping. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* TODO: - Maybe drop 5-pole noise shaping and use coefficients * generated by dmaker * http://shibatch.sf.net */ #include #include #include #include "gstaudiopack.h" #include "audio-quantize.h" typedef void (*QuantizeFunc) (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint count); struct _GstAudioQuantize { GstAudioDitherMethod dither; GstAudioNoiseShapingMethod ns; GstAudioQuantizeFlags flags; GstAudioFormat format; guint quantizer; guint stride; guint blocks; guint shift; guint32 mask, bias; /* last random number generated per channel for hifreq TPDF dither */ gpointer last_random; /* contains the past quantization errors, error[channels][count] */ guint error_size; gpointer error_buf; /* buffer with dither values */ guint dither_size; gpointer dither_buf; /* noise shaping coefficients */ gpointer coeffs; gint n_coeffs; QuantizeFunc quantize; }; #define ADDSS(res,val) \ if (val > 0 && res > 0 && G_MAXINT32 - res <= val){ \ res = G_MAXINT32; \ } else if (val < 0 && res < 0 && G_MININT32 - res >= val){ \ res = G_MININT32; \ } else \ res += val; static void gst_audio_quantize_quantize_memcpy (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint samples) { if (src != dst) memcpy (dst, src, samples * sizeof (gint32) * quant->stride); } /* Quantize functions for gint32 as intermediate format */ static void gst_audio_quantize_quantize_int_none_none (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint samples) { audio_orc_int_bias (dst, src, quant->bias, ~quant->mask, samples * quant->stride); } /* This is the base function, implementing a linear congruential generator * and returning a pseudo random number between 0 and 2^32 - 1. */ static inline guint32 gst_fast_random_uint32 (void) { static guint32 state = 0xdeadbeef; return (state = state * 1103515245 + 12345); } static inline gint32 gst_fast_random_int32 (void) { return (gint32) gst_fast_random_uint32 (); } /* Assuming dither == 2^n, * returns one of 2^(n+1) possible random values: * -dither <= retval < dither */ #define RANDOM_INT_DITHER(dither) \ (- dither + (gst_fast_random_int32 () & ((dither << 1) - 1))) static void setup_dither_buf (GstAudioQuantize * quant, gint samples) { gboolean need_init = FALSE; gint stride = quant->stride; gint i, len = samples * stride; guint shift = quant->shift; guint32 bias; gint32 dither, *d; if (quant->dither_size < len) { quant->dither_size = len; quant->dither_buf = g_realloc (quant->dither_buf, len * sizeof (gint32)); need_init = TRUE; } bias = quant->bias; d = quant->dither_buf; switch (quant->dither) { case GST_AUDIO_DITHER_NONE: if (need_init) { for (i = 0; i < len; i++) d[i] = 0; } break; case GST_AUDIO_DITHER_RPDF: dither = 1 << (shift); for (i = 0; i < len; i++) d[i] = bias + RANDOM_INT_DITHER (dither); break; case GST_AUDIO_DITHER_TPDF: dither = 1 << (shift - 1); for (i = 0; i < len; i++) d[i] = bias + RANDOM_INT_DITHER (dither) + RANDOM_INT_DITHER (dither); break; case GST_AUDIO_DITHER_TPDF_HF: { gint32 tmp, *last_random = quant->last_random; dither = 1 << (shift - 1); for (i = 0; i < len; i++) { tmp = RANDOM_INT_DITHER (dither); d[i] = bias + tmp - last_random[i % stride]; last_random[i % stride] = tmp; } break; } } } static void gst_audio_quantize_quantize_int_dither_none (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint samples) { setup_dither_buf (quant, samples); audio_orc_int_dither (dst, src, quant->dither_buf, ~quant->mask, samples * quant->stride); } static void setup_error_buf (GstAudioQuantize * quant, gint samples) { gint stride = quant->stride; gint len = (samples + quant->n_coeffs) * stride; if (quant->error_size < len) { quant->error_buf = g_realloc (quant->error_buf, len * sizeof (gint32)); if (quant->error_size == 0) memset ((gint32 *) quant->error_buf, 0, stride * quant->n_coeffs * sizeof (gint32)); quant->error_size = len; } } static void gst_audio_quantize_quantize_int_dither_feedback (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint samples) { guint32 mask; gint i, len, stride; const gint32 *s = src; gint32 *dith, *d = dst, v, o, *e, err; setup_dither_buf (quant, samples); setup_error_buf (quant, samples); stride = quant->stride; len = samples * stride; dith = quant->dither_buf; e = quant->error_buf; mask = ~quant->mask; for (i = 0; i < len; i++) { o = v = s[i]; /* add dither */ err = dith[i]; /* remove error */ err -= e[i]; ADDSS (v, err); v &= mask; /* store new error */ e[i + stride] = e[i] + (v - o); /* store result */ d[i] = v; } memmove (e, &e[len], sizeof (gint32) * stride); } #define SHIFT 10 #define REDUCE 8 #define RROUND (1<<(REDUCE-1)) #define SREDUCE 2 #define SROUND (1<<(SREDUCE-1)) static void gst_audio_quantize_quantize_int_dither_noise_shape (GstAudioQuantize * quant, const gpointer src, gpointer dst, gint samples) { guint32 mask; gint i, j, k, len, stride, nc; const gint32 *s = src; gint32 *c, *dith, *d = dst, v, o, *e, err; setup_dither_buf (quant, samples); setup_error_buf (quant, samples); stride = quant->stride; len = samples * stride; dith = quant->dither_buf; e = quant->error_buf; c = quant->coeffs; nc = quant->n_coeffs; mask = ~quant->mask; for (i = 0; i < len; i++) { v = s[i]; /* combine and remove error */ err = 0; for (j = 0, k = i; j < nc; j++, k += stride) err -= e[k] * c[j]; err = (err + SROUND) >> (SREDUCE); ADDSS (v, err); o = v; /* add dither */ err = dith[i]; ADDSS (v, err); /* quantize */ v &= mask; /* store new error with reduced precision */ e[k] = (v - o + RROUND) >> REDUCE; /* store result */ d[i] = v; } memmove (e, &e[len], sizeof (gint32) * stride * nc); } #define MAKE_QUANTIZE_FUNC_NAME(name) \ gst_audio_quantize_quantize_##name static const QuantizeFunc quantize_funcs[] = { (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_none_none), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape), }; /* Same as error feedback but also add 1/2 of the previous error value. * This moves the noise a bit more into the higher frequencies. */ static const gdouble ns_simple_coeffs[] = { -0.5, 1.0 }; /* Noise shaping coefficients from[1], moves most power of the * error noise into inaudible frequency ranges. * * [1] * "Minimally Audible Noise Shaping", Stanley P. Lipshitz, * John Vanderkooy, and Robert A. Wannamaker, * J. Audio Eng. Soc., Vol. 39, No. 11, November 1991. */ static const gdouble ns_medium_coeffs[] = { 0.6149, -1.590, 1.959, -2.165, 2.033 }; /* Noise shaping coefficients by David Schleef, moves most power of the * error noise into inaudible frequency ranges */ static const gdouble ns_high_coeffs[] = { -0.340122, 0.876066, -1.72008, 2.61339, -3.31399, 3.27918, -2.92975, 2.08484, }; static void gst_audio_quantize_setup_noise_shaping (GstAudioQuantize * quant) { gint i, n_coeffs = 0; gint32 *q; const gdouble *coeffs; switch (quant->ns) { case GST_AUDIO_NOISE_SHAPING_HIGH: n_coeffs = 8; coeffs = ns_high_coeffs; break; case GST_AUDIO_NOISE_SHAPING_MEDIUM: n_coeffs = 5; coeffs = ns_medium_coeffs; break; case GST_AUDIO_NOISE_SHAPING_SIMPLE: n_coeffs = 2; coeffs = ns_simple_coeffs; break; case GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK: break; case GST_AUDIO_NOISE_SHAPING_NONE: default: break; } if (n_coeffs) { quant->n_coeffs = n_coeffs; q = quant->coeffs = g_new0 (gint32, n_coeffs); for (i = 0; i < n_coeffs; i++) q[i] = floor (coeffs[i] * (1 << SHIFT) + 0.5); } return; } static void gst_audio_quantize_setup_dither (GstAudioQuantize * quant) { switch (quant->dither) { case GST_AUDIO_DITHER_TPDF_HF: quant->last_random = g_new0 (gint32, quant->stride); break; case GST_AUDIO_DITHER_RPDF: case GST_AUDIO_DITHER_TPDF: quant->last_random = NULL; break; case GST_AUDIO_DITHER_NONE: default: quant->last_random = NULL; break; } return; } static void gst_audio_quantize_setup_quantize_func (GstAudioQuantize * quant) { gint index; if (quant->shift == 0) { quant->quantize = (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (memcpy); return; } index = 5 * quant->dither + quant->ns; quant->quantize = quantize_funcs[index]; } static gint count_power (guint v) { gint res = 0; while (v > 1) { res++; v >>= 1; } return res; } /** * gst_audio_quantize_new: (skip): * @dither: a #GstAudioDitherMethod * @ns: a #GstAudioNoiseShapingMethod * @flags: #GstAudioQuantizeFlags * @format: the #GstAudioFormat of the samples * @channels: the amount of channels in the samples * @quantizer: the quantizer to use * * Create a new quantizer object with the given parameters. * * Output samples will be quantized to a multiple of @quantizer. Better * performance is achieved when @quantizer is a power of 2. * * Dithering and noise-shaping can be performed during quantization with * the @dither and @ns parameters. * * Returns: a new #GstAudioQuantize. Free with gst_audio_quantize_free(). */ GstAudioQuantize * gst_audio_quantize_new (GstAudioDitherMethod dither, GstAudioNoiseShapingMethod ns, GstAudioQuantizeFlags flags, GstAudioFormat format, guint channels, guint quantizer) { GstAudioQuantize *quant; g_return_val_if_fail (format == GST_AUDIO_FORMAT_S32, NULL); g_return_val_if_fail (channels > 0, NULL); quant = g_slice_new0 (GstAudioQuantize); quant->dither = dither; quant->ns = ns; quant->flags = flags; quant->format = format; if (flags & GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED) { quant->stride = 1; quant->blocks = channels; } else { quant->stride = channels; quant->blocks = 1; } quant->quantizer = quantizer; quant->shift = count_power (quantizer); if (quant->shift > 0) quant->bias = (1U << (quant->shift - 1)); else quant->bias = 0; quant->mask = (1U << quant->shift) - 1; gst_audio_quantize_setup_dither (quant); gst_audio_quantize_setup_noise_shaping (quant); gst_audio_quantize_setup_quantize_func (quant); return quant; } /** * gst_audio_quantize_free: * @quant: a #GstAudioQuantize * * Free a #GstAudioQuantize. */ void gst_audio_quantize_free (GstAudioQuantize * quant) { g_return_if_fail (quant != NULL); g_free (quant->error_buf); g_free (quant->coeffs); g_free (quant->last_random); g_free (quant->dither_buf); g_slice_free (GstAudioQuantize, quant); } /** * gst_audio_quantize_samples: * @quant: a #GstAudioQuantize * @in: input samples * @out: output samples * @samples: number of samples * * Perform quantization on @samples in @in and write the result to @out. * * In case the samples are interleaved, @in and @out must point to an * array with a single element pointing to a block of interleaved samples. * * If non-interleaved samples are used, @in and @out must point to an * array with pointers to memory blocks, one for each channel. * * @in and @out may point to the same memory location, in which case samples will be * modified in-place. */ void gst_audio_quantize_samples (GstAudioQuantize * quant, const gpointer in[], gpointer out[], guint samples) { guint i; g_return_if_fail (quant != NULL); g_return_if_fail (out != NULL || samples == 0); g_return_if_fail (in != NULL || samples == 0); for (i = 0; i < quant->blocks; i++) quant->quantize (quant, in[i], out[i], samples); }