mirror of
https://github.com/outbackdingo/UltraGrid.git
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This avoids pops and buzzing caused by integer overflow when the input sample falls outside of the range [-1.0, 1.0] (I encountered this when using jack capture).
445 lines
14 KiB
C++
445 lines
14 KiB
C++
/**
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* @file audio/utils.cpp
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* @author Martin Pulec <pulec@cesnet.cz>
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*/
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/*
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* Copyright (c) 2011-2014 CESNET z.s.p.o.
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, is permitted provided that the following conditions
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* are met:
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*
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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*
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* 3. Neither the name of CESNET nor the names of its contributors may be
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* used to endorse or promote products derived from this software without
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* specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
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* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
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* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
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* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
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* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
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* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#include "config_unix.h"
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#include "config_win32.h"
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#endif // HAVE_CONFIG_H
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#include "audio/audio.h"
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#include "audio/codec.h"
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#include "audio/utils.h"
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#include "debug.h"
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#include <assert.h>
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#include <limits.h>
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#include <math.h>
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#include <stdio.h>
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#include <string.h>
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#ifdef WORDS_BIGENDIAN
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#error "This code will not run with a big-endian machine. Please report a bug to " PACKAGE_BUGREPORT " if you reach here."
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#endif // WORDS_BIGENDIAN
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using namespace std;
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static double get_normalized(const int8_t *in, int bps) {
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int64_t sample = 0;
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bool negative = false;
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for (int j = 0; j < bps; ++j) {
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sample = (sample | ((((const uint8_t *)in)[j]) << (uint64_t)(8ull * j)));
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}
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if ((int8_t)(in[bps - 1] < 0))
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negative = true;
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if (negative) {
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for (int i = bps; i < 8; ++i) {
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sample = (sample | (255ull << (8ull * i)));
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}
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}
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return (double) sample / ((1 << (bps * 8 - 1)));
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}
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/**
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* @brief Calculates mean and peak RMS from audio samples
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*
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* @param[in] frame audio frame
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* @param[in] channel channel index to calculate RMS to
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* @param[out] peak peak RMS
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* @returns mean RMS
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*/
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double calculate_rms(audio_frame2 *frame, int channel, double *peak)
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{
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assert(frame->get_codec() == AC_PCM);
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double sum = 0;
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*peak = 0;
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int sample_count = frame->get_data_len(channel) / frame->get_bps();
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const char *channel_data = frame->get_data(channel);
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for (size_t i = 0; i < frame->get_data_len(channel); i += frame->get_bps()) {
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double val = get_normalized((const int8_t *) channel_data + i, frame->get_bps());
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sum += val;
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if (fabs(val) > *peak) {
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*peak = fabs(val);
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}
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}
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double average = sum / sample_count;
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double sumMeanSquare = 0.0;
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for (size_t i = 0; i < frame->get_data_len(channel); i += frame->get_bps()) {
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sumMeanSquare += pow(get_normalized((const int8_t *) channel_data + i, frame->get_bps())
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- average, 2.0);
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}
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double averageMeanSquare = sumMeanSquare / sample_count;
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double rootMeanSquare = sqrt(averageMeanSquare);
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return rootMeanSquare;
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}
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bool audio_desc_eq(struct audio_desc a1, struct audio_desc a2) {
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return a1.bps == a2.bps &&
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a1.sample_rate == a2.sample_rate &&
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a1.ch_count == a2.ch_count &&
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a1.codec == a2.codec;
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}
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struct audio_desc audio_desc_from_audio_frame(struct audio_frame *frame) {
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return audio_desc { frame->bps,
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frame->sample_rate,
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frame->ch_count,
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AC_PCM
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};
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}
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struct audio_desc audio_desc_from_audio_channel(audio_channel *channel) {
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return audio_desc { channel->bps,
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channel->sample_rate,
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1,
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channel->codec
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};
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}
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/**
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* Copies desc from desc to f.
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*
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* @note
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* Doesn't clear/set other members of f, thus caller needs to do that first if needed.
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*/
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void audio_frame_write_desc(struct audio_frame *f, struct audio_desc desc)
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{
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f->bps = desc.bps;
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f->sample_rate = desc.sample_rate;
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f->ch_count = desc.ch_count;
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}
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void change_bps(char *out, int out_bps, const char *in, int in_bps, int in_len /* bytes */)
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{
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int i;
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assert ((unsigned int) out_bps <= sizeof(int32_t));
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for(i = 0; i < in_len / in_bps; i++) {
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int32_t in_value = format_from_in_bps(in, in_bps);
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int32_t out_value;
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if(in_bps > out_bps) {
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out_value = in_value >> (in_bps * 8 - out_bps * 8);
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} else {
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out_value = in_value << (out_bps * 8 - in_bps * 8);
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}
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format_to_out_bps(out, out_bps, out_value);
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in += in_bps;
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out += out_bps;
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}
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}
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void copy_channel(char *out, const char *in, int bps, int in_len /* bytes */, int out_channel_count)
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{
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int samples = in_len / bps;
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int i;
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assert(out_channel_count > 0);
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assert(bps > 0);
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assert(in_len >= 0);
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in += in_len;
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out += in_len * out_channel_count;
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for (i = samples; i > 0 ; --i) {
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int j;
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in -= bps;
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for (j = out_channel_count + 0; j > 0; --j) {
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out -= bps;
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memmove(out, in, bps);
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}
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}
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}
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void audio_frame_multiply_channel(struct audio_frame *frame, int new_channel_count) {
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assert((unsigned int) frame->max_size >= (unsigned int) frame->data_len * new_channel_count / frame->ch_count);
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copy_channel(frame->data, frame->data, frame->bps, frame->data_len, new_channel_count);
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}
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void demux_channel(char *out, char *in, int bps, int in_len, int in_stream_channels, int pos_in_stream)
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{
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int samples = in_len / (in_stream_channels * bps);
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int i;
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assert (bps <= 4);
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in += pos_in_stream * bps;
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for (i = 0; i < samples; ++i) {
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memcpy(out, in, bps);
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out += bps;
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in += in_stream_channels * bps;
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}
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}
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void remux_channel(char *out, const char *in, int bps, int in_len, int in_stream_channels, int out_stream_channels, int pos_in_stream, int pos_out_stream)
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{
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int samples = in_len / (in_stream_channels * bps);
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int i;
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assert (bps <= 4);
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in += pos_in_stream * bps;
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out += pos_out_stream * bps;
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for (i = 0; i < samples; ++i) {
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memcpy(out, in, bps);
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out += bps * out_stream_channels;
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in += bps * in_stream_channels;
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}
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}
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void mux_channel(char *out, const char *in, int bps, int in_len, int out_stream_channels, int pos_in_stream, double scale)
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{
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int samples = in_len / bps;
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int i;
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assert (bps <= 4);
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out += pos_in_stream * bps;
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if(scale == 1.0) {
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for (i = 0; i < samples; ++i) {
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memcpy(out, in, bps);
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in += bps;
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out += out_stream_channels * bps;
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}
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} else {
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for (i = 0; i < samples; ++i) {
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int32_t in_value = format_from_in_bps(in, bps);
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in_value *= scale;
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format_to_out_bps(out, bps, in_value);
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in += bps;
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out += out_stream_channels * bps;
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}
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}
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}
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void mux_and_mix_channel(char *out, const char *in, int bps, int in_len, int out_stream_channels, int pos_in_stream, double scale)
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{
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int i;
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assert (bps <= 4);
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out += pos_in_stream * bps;
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for(i = 0; i < in_len / bps; i++) {
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int32_t in_value = format_from_in_bps(in, bps);
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int32_t out_value = format_from_in_bps(out, bps);
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int32_t new_value = (double)in_value * scale + out_value;
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format_to_out_bps(out, bps, new_value);
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in += bps;
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out += out_stream_channels * bps;
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}
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}
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double get_avg_volume(char *data, int bps, int in_len, int stream_channels, int pos_in_stream)
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{
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float average_vol = 0;
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int i;
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assert ((unsigned int) bps <= sizeof(int32_t));
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data += pos_in_stream * bps;
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for(i = 0; i < in_len / bps; i++) {
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int32_t in_value = format_from_in_bps(data, bps);
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//if(pos_in_stream) fprintf(stderr, "%d-%d ", pos_in_stream, data);
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average_vol = average_vol * (i / ((double) i + 1)) +
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fabs(((double) in_value / ((1 << (bps * 8 - 1)) - 1)) / (i + 1));
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data += bps * stream_channels;
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}
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return average_vol;
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}
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void float2int(char *out, const char *in, int len)
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{
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const float *inf = (const float *)(const void *) in;
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int32_t *outi = (int32_t *)(void *) out;
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int items = len / sizeof(int32_t);
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while(items-- > 0) {
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float sample = *inf++;
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if(sample > 1.0) sample = 1.0;
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if(sample < -1.0) sample = -1.0;
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*outi++ = sample * INT_MAX;
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}
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}
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void int2float(char *out, const char *in, int len)
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{
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const int32_t *ini = (const int32_t *)(const void *) in;
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float *outf = (float *)(void *) out;
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int items = len / sizeof(int32_t);
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while(items-- > 0) {
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*outf++ = (float) *ini++ / INT_MAX;
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}
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}
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void short_int2float(char *out, char *in, int in_len)
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{
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int16_t *ini = (int16_t *)(void *) in;
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float *outf = (float *)(void *) out;
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int items = in_len / sizeof(int16_t);
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while(items-- > 0) {
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*outf++ = (float) *ini++ / SHRT_MAX;
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}
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}
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void signed2unsigned(char *out, char *in, int in_len)
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{
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int8_t *inch = (int8_t *) in;
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uint8_t *outch = (uint8_t *) out;
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int items = in_len / sizeof(int8_t);
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while(items-- > 0) {
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int8_t in_value = *inch++;
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uint8_t out_value = (int) 128 + in_value;
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*outch++ = out_value;
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}
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}
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void audio_channel_demux(const audio_frame2 *frame, int index, audio_channel *channel)
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{
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channel->data = frame->get_data(index);
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channel->data_len = frame->get_data_len(index);
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channel->codec = frame->get_codec();
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channel->bps = frame->get_bps();
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channel->sample_rate = frame->get_sample_rate();
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}
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int32_t format_from_in_bps(const char * in, int bps) {
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int32_t in_value = 0;
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memcpy(&in_value, in, bps);
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if(in_value >> (bps * 8 - 1) && bps != 4) { //negative
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in_value |= ((1<<(32 - bps * 8)) - 1) << (bps * 8);
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}
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return in_value;
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}
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void format_to_out_bps(char *out, int bps, int32_t out_value) {
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uint32_t mask = ((1ll << (bps * 8)) - 1);
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// clamp
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if(out_value > (1ll << (bps * 8 - 1)) -1) {
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out_value = (1ll << (bps * 8 - 1)) -1;
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}
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if(out_value < -(1ll << (bps * 8 - 1))) {
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out_value = -(1ll << (bps * 8 - 1));
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}
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uint32_t out_value_formatted = (1 * (0x1 & (out_value >> 31))) << (bps * 8 - 1) | (out_value & mask);
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memcpy(out, &out_value_formatted, bps);
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}
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void interleaved2noninterleaved(char *out, const char *in, int bps, int in_len, int channel_count)
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{
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vector<char *> out_ch(channel_count);
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for (int i = 0; i < channel_count; ++i) {
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out_ch[i] = out + in_len / channel_count * i;
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}
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int offset = 0;
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int index = 0;
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while (offset < in_len) {
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memcpy(out_ch[index], in, bps);
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out_ch[index] += bps;
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index = (index + 1) % channel_count;
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in += bps;
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offset += bps;
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}
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}
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bool append_audio_frame(struct audio_frame *frame, char *data, size_t data_len) {
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bool ret = true;
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if (frame->data_len + data_len > (size_t) frame->max_size) {
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log_msg(LOG_LEVEL_WARNING, "Audio frame overrun, discarding some data.\n");
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data_len = frame->max_size - frame->data_len;
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ret = false;
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}
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memcpy(frame->data + frame->data_len, data, data_len);
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frame->data_len += data_len;
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return ret;
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}
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struct audio_frame *audio_frame_copy(const struct audio_frame *src, bool keep_size) {
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struct audio_frame *ret = (struct audio_frame *) malloc(sizeof(struct audio_frame));
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memcpy(ret, src, sizeof *ret);
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if (!keep_size) {
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ret->max_size = src->data_len;
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}
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ret->data = (char *) malloc(ret->max_size);
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memcpy(ret->data, src->data, src->data_len);
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return ret;
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}
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