Files
UltraGrid/src/audio/utils.cpp
Martin Pulec 505e648b8a chage_bps2: small fixes
- if in/out bps matches, use memcpy
- static assert implementation-defined behavior we depend on
  (right-shifting negative LHS)
2022-02-08 08:37:32 +01:00

565 lines
18 KiB
C++

/**
* @file audio/utils.cpp
* @author Martin Pulec <pulec@cesnet.cz>
* @author Martin Piatka <piatka@cesnet.cz>
*/
/*
* Copyright (c) 2011-2021 CESNET z.s.p.o.
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, is permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* 3. Neither the name of CESNET nor the names of its contributors may be
* used to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#include "config_unix.h"
#include "config_win32.h"
#endif // HAVE_CONFIG_H
#include <cassert>
#include <climits>
#include <cmath>
#include <cstring>
#include "audio/codec.h"
#include "audio/types.h"
#include "audio/utils.h"
#include "debug.h"
#include "host.h" // ADD_TO_PARAM
#ifdef WORDS_BIGENDIAN
#error "This code will not run with a big-endian machine. Please report a bug to " PACKAGE_BUGREPORT " if you reach here."
#endif // WORDS_BIGENDIAN
using namespace std;
/**
* Loads sample with BPS width and returns it cast to
* int32_t.
*/
template<int BPS> static int32_t load_sample(const char *data);
template<> int32_t load_sample<1>(const char *data) {
return *reinterpret_cast<const int8_t *>(data);
}
template<> int32_t load_sample<2>(const char *data) {
return *reinterpret_cast<const int16_t *>(data);
}
template<> int32_t load_sample<3>(const char *data) {
int32_t in_value = 0;
memcpy(&in_value, data, 3);
if ((in_value & 1U<<23U) != 0U) { // negative
in_value |= 0xFF000000U;
}
return in_value;
}
template<> int32_t load_sample<4>(const char *data) {
return *reinterpret_cast<const int32_t *>(data);
}
template<int BPS> static void store_sample(char *data, int32_t val);
template<> void store_sample<1>(char *data, int32_t val) {
*reinterpret_cast<int8_t *>(data) = clamp(val, INT8_MIN, INT8_MAX);
}
template<> void store_sample<2>(char *data, int32_t val) {
*reinterpret_cast<int16_t *>(data) = clamp(val, INT16_MIN, INT16_MAX);
}
template<> void store_sample<3>(char *data, int32_t val) {
val = clamp<int32_t>(val, -(1L<<24), (1L<<24) - 1);
memcpy(data, &val, 3);
}
template<> void store_sample<4>(char *data, int32_t val) {
*reinterpret_cast<int32_t *>(data) = val;
}
/**
* @brief Calculates mean and peak RMS from audio samples
*
* @param[in] frame audio frame
* @param[in] channel channel index to calculate RMS to
* @param[out] peak peak RMS
* @returns mean RMS
*/
template<int BPS>
static double calculate_rms_helper(const char *channel_data, int sample_count, double *peak)
{
double sum = 0;
*peak = 0;
for (int i = 0; i < sample_count; i += 1) {
double val = load_sample<BPS>(channel_data + i * BPS) / static_cast<double>(1U << (BPS * CHAR_BIT - 1U));
sum += val;
*peak = max(fabs(val), *peak);
}
double average = sum / sample_count;
double sumMeanSquare = 0.0;
for (int i = 0; i < sample_count; i += 1) {
sumMeanSquare += pow(load_sample<BPS>(channel_data + i * BPS) / static_cast<double>(1U << (BPS * CHAR_BIT - 1U))
- average, 2.0);
}
double averageMeanSquare = sumMeanSquare / sample_count;
double rootMeanSquare = sqrt(averageMeanSquare);
return rootMeanSquare;
}
/**
* @brief Calculates mean and peak RMS from audio samples
*
* @param[in] frame audio frame
* @param[in] channel channel index to calculate RMS to
* @param[out] peak peak RMS
* @returns mean RMS
*/
double calculate_rms(audio_frame2 *frame, int channel, double *peak)
{
assert(frame->get_codec() == AC_PCM);
int sample_count = frame->get_data_len(channel) / frame->get_bps();
switch (frame->get_bps()) {
case 1:
return calculate_rms_helper<1>(frame->get_data(channel), sample_count, peak);
case 2:
return calculate_rms_helper<2>(frame->get_data(channel), sample_count, peak);
case 3:
return calculate_rms_helper<3>(frame->get_data(channel), sample_count, peak);
case 4:
return calculate_rms_helper<4>(frame->get_data(channel), sample_count, peak);
default:
LOG(LOG_LEVEL_FATAL) << "Wrong BPS " << frame->get_bps() << "\n";
abort();
}
}
bool audio_desc_eq(struct audio_desc a1, struct audio_desc a2) {
return a1.bps == a2.bps &&
a1.sample_rate == a2.sample_rate &&
a1.ch_count == a2.ch_count &&
a1.codec == a2.codec;
}
struct audio_desc audio_desc_from_audio_frame(struct audio_frame *frame) {
return audio_desc { frame->bps,
frame->sample_rate,
frame->ch_count,
AC_PCM
};
}
struct audio_desc audio_desc_from_audio_channel(audio_channel *channel) {
return audio_desc { channel->bps,
channel->sample_rate,
1,
channel->codec
};
}
/**
* Copies desc from desc to f.
*
* @note
* Doesn't clear/set other members of f, thus caller needs to do that first if needed.
*/
void audio_frame_write_desc(struct audio_frame *f, struct audio_desc desc)
{
f->bps = desc.bps;
f->sample_rate = desc.sample_rate;
f->ch_count = desc.ch_count;
}
int32_t downshift_with_dither(int32_t val, int shift){
static thread_local std::uint_fast32_t last_rand = 1;
const int mask = (1 << shift) - 1;
//Pseudorandom number generation, same parameters as std::minstd_rand
last_rand = (last_rand * 48271) % 2147483647;
int triangle_dither = last_rand & mask;
last_rand = (last_rand * 48271) % 2147483647;
triangle_dither -= last_rand & mask; //triangle probability distribution
/* Prevent over/underflow when val is big.
*
* abs(val) could cause problems if val is INT32_MIN, but integer
* 32-bit pcm is rare and should not contain the value INT32_MIN
* anyway because of symmetry, as specified by AES17 and IEC 61606-3
*/
if(INT32_MAX - abs(val) < mask)
return val >> shift;
return (val + triangle_dither) >> shift;
}
#define NO_DITHER_PARAM "no-dither"
ADD_TO_PARAM(NO_DITHER_PARAM, "* " NO_DITHER_PARAM "\n"
" Disable audio dithering when resampling\n");
void change_bps(char *out, int out_bps, const char *in, int in_bps, int in_len /* bytes */) {
static const bool dither = commandline_params.find(NO_DITHER_PARAM) == commandline_params.end();
change_bps2(out, out_bps, in, in_bps, in_len, dither);
}
void change_bps2(char *out, int out_bps, const char *in, int in_bps, int in_len /* bytes */, bool dither)
{
assert ((unsigned int) out_bps <= sizeof(int32_t));
static_assert(-2>>1 == -1, "Implementation-defined behavior doesn't work as expected by the implementation.");
if (in_bps == out_bps ) {
memcpy(out, in, in_len);
return;
}
if (in_bps < out_bps ) {
for (int i = 0; i < in_len / in_bps; i++) {
int32_t in_value = format_from_in_bps(in, in_bps);
int32_t out_value = in_value << (out_bps * 8 - in_bps * 8);
format_to_out_bps(out, out_bps, out_value);
in += in_bps;
out += out_bps;
}
return;
}
// downsampling
if (dither) {
const int downshift = in_bps * 8 - out_bps * 8;
for (int i = 0; i < in_len / in_bps; i++) {
int32_t in_value = format_from_in_bps(in, in_bps);
int32_t out_value = downshift_with_dither(in_value, downshift);
format_to_out_bps(out, out_bps, out_value);
in += in_bps;
out += out_bps;
}
return;
}
// no dithering
for (int i = 0; i < in_len / in_bps; i++) {
int32_t in_value = format_from_in_bps(in, in_bps);
int32_t out_value = in_value >> (in_bps * 8 - out_bps * 8);
format_to_out_bps(out, out_bps, out_value);
in += in_bps;
out += out_bps;
}
}
void copy_channel(char *out, const char *in, int bps, int in_len /* bytes */, int out_channel_count)
{
int samples = in_len / bps;
int i;
assert(out_channel_count > 0);
assert(bps > 0);
assert(in_len >= 0);
in += in_len;
out += in_len * out_channel_count;
for (i = samples; i > 0 ; --i) {
int j;
in -= bps;
for (j = out_channel_count + 0; j > 0; --j) {
out -= bps;
memmove(out, in, bps);
}
}
}
void audio_frame_multiply_channel(struct audio_frame *frame, int new_channel_count) {
assert((unsigned int) frame->max_size >= (unsigned int) frame->data_len * new_channel_count / frame->ch_count);
copy_channel(frame->data, frame->data, frame->bps, frame->data_len, new_channel_count);
}
void demux_channel(char *out, char *in, int bps, int in_len, int in_stream_channels, int pos_in_stream)
{
int samples = in_len / (in_stream_channels * bps);
int i;
assert (bps <= 4);
in += pos_in_stream * bps;
for (i = 0; i < samples; ++i) {
memcpy(out, in, bps);
out += bps;
in += in_stream_channels * bps;
}
}
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)
{
int samples = in_len / (in_stream_channels * bps);
int i;
assert (bps <= 4);
in += pos_in_stream * bps;
out += pos_out_stream * bps;
for (i = 0; i < samples; ++i) {
memcpy(out, in, bps);
out += bps * out_stream_channels;
in += bps * in_stream_channels;
}
}
void mux_channel(char *out, const char *in, int bps, int in_len, int out_stream_channels, int pos_in_stream, double scale)
{
int samples = in_len / bps;
int i;
assert (bps <= 4);
out += pos_in_stream * bps;
if(scale == 1.0) {
for (i = 0; i < samples; ++i) {
memcpy(out, in, bps);
in += bps;
out += out_stream_channels * bps;
}
} else {
for (i = 0; i < samples; ++i) {
int32_t in_value = format_from_in_bps(in, bps);
in_value *= scale;
format_to_out_bps(out, bps, in_value);
in += bps;
out += out_stream_channels * bps;
}
}
}
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)
{
int i;
assert (bps <= 4);
out += pos_in_stream * bps;
for(i = 0; i < in_len / bps; i++) {
int32_t in_value = format_from_in_bps(in, bps);
int32_t out_value = format_from_in_bps(out, bps);
int32_t new_value = (double)in_value * scale + out_value;
format_to_out_bps(out, bps, new_value);
in += bps;
out += out_stream_channels * bps;
}
}
template<int BPS>
static double get_avg_volume_helper(const char *data, int sample_count, int stream_channels, int pos_in_stream)
{
int64_t vol = 0;
data += pos_in_stream * BPS;
for (int i = 0; i < sample_count; i++) {
int32_t in_value = load_sample<BPS> (data + i * BPS * stream_channels);
vol += labs(in_value);
}
return static_cast<double>(vol) / sample_count / ((1U << (BPS * 8U - 1U)));
}
double get_avg_volume(char *data, int bps, int sample_count, int stream_channels, int pos_in_stream) {
switch (bps) {
case 1:
return get_avg_volume_helper<1>(data, sample_count, stream_channels, pos_in_stream);
case 2:
return get_avg_volume_helper<2>(data, sample_count, stream_channels, pos_in_stream);
case 3:
return get_avg_volume_helper<3>(data, sample_count, stream_channels, pos_in_stream);
case 4:
return get_avg_volume_helper<4>(data, sample_count, stream_channels, pos_in_stream);
default:
LOG(LOG_LEVEL_FATAL) << "Wrong BPS " << bps << "\n";
abort();
}
}
const float INT_MAX_FLT = nexttowardf((float) INT_MAX, INT_MAX); // max int representable as float
/**
* Can be used in situ.
*/
void float2int(char *out, const char *in, int len)
{
const float *inf = (const float *)(const void *) in;
int32_t *outi = (int32_t *)(void *) out;
int items = len / sizeof(int32_t);
while(items-- > 0) {
float sample = *inf++;
if(sample > 1.0) sample = 1.0;
if(sample < -1.0) sample = -1.0;
*outi++ = sample * INT_MAX_FLT;
}
}
void int2float(char *out, const char *in, int len)
{
const int32_t *ini = (const int32_t *)(const void *) in;
float *outf = (float *)(void *) out;
int items = len / sizeof(int32_t);
while(items-- > 0) {
*outf++ = (float) *ini++ / (float) INT_MAX;
}
}
void short_int2float(char *out, const char *in, int in_len)
{
const auto *ini = reinterpret_cast<const int16_t *>(in);
float *outf = (float *)(void *) out;
int items = in_len / sizeof(int16_t);
while(items-- > 0) {
*outf++ = (float) *ini++ / SHRT_MAX;
}
}
/**
* Converts int8_t samples to uint8_t by adding 128 (standard
* shifted zero unsigned samples).
*
* Works also in the opposite direction!
*
* Can be used in situ.
*/
void signed2unsigned(char *out, const char *in, int in_len)
{
const auto *inch = reinterpret_cast<const int8_t *>(in);
uint8_t *outch = (uint8_t *) out;
int items = in_len / sizeof(int8_t);
while(items-- > 0) {
int8_t in_value = *inch++;
uint8_t out_value = (int) 128 + in_value;
*outch++ = out_value;
}
}
void audio_channel_demux(const audio_frame2 *frame, int index, audio_channel *channel)
{
channel->data = frame->get_data(index);
channel->data_len = frame->get_data_len(index);
channel->codec = frame->get_codec();
channel->bps = frame->get_bps();
channel->sample_rate = frame->get_sample_rate();
}
int32_t format_from_in_bps(const char * in, int bps) {
switch (bps) {
case 1: return load_sample<1>(in);
case 2: return load_sample<2>(in);
case 3: return load_sample<3>(in);
case 4: return load_sample<4>(in);
default: abort();
}
}
void format_to_out_bps(char *out, int bps, int32_t out_value) {
switch (bps) {
case 1: store_sample<1>(out, out_value); break;
case 2: store_sample<2>(out, out_value); break;
case 3: store_sample<3>(out, out_value); break;
case 4: store_sample<4>(out, out_value); break;
}
}
void interleaved2noninterleaved(char *out, const char *in, int bps, int in_len, int channel_count)
{
vector<char *> out_ch(channel_count);
for (int i = 0; i < channel_count; ++i) {
out_ch[i] = out + in_len / channel_count * i;
}
int offset = 0;
int index = 0;
while (offset < in_len) {
memcpy(out_ch[index], in, bps);
out_ch[index] += bps;
index = (index + 1) % channel_count;
in += bps;
offset += bps;
}
}
bool append_audio_frame(struct audio_frame *frame, char *data, size_t data_len) {
bool ret = true;
if (frame->data_len + data_len > (size_t) frame->max_size) {
log_msg(LOG_LEVEL_WARNING, "Audio frame overrun, discarding some data.\n");
data_len = frame->max_size - frame->data_len;
ret = false;
}
memcpy(frame->data + frame->data_len, data, data_len);
frame->data_len += data_len;
return ret;
}
struct audio_frame *audio_frame_copy(const struct audio_frame *src, bool keep_size) {
struct audio_frame *ret = (struct audio_frame *) malloc(sizeof(struct audio_frame));
memcpy(ret, src, sizeof *ret);
if (!keep_size) {
ret->max_size = src->data_len;
}
ret->data = (char *) malloc(ret->max_size);
memcpy(ret->data, src->data, src->data_len);
return ret;
}