Unable to do simple audio transformations
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Re: Unable to do simple audio transformations
My 2 cents:
Cent #1: You read from indices "j + ...", but you write the data to "sIndex + j + ..." - effectively overwriting all data with the data from the first 4 frames. That's totally not what you want.
Cent #2: Still not sure about the sample values you posted. Is your input plausibly of an amplitude of within a few percent of the maximum possible 'loudness'? Also notice that e.g. the first sample (-8186112), when multiplied by 1.1, yields -9004723, which is beyond the range of a 24-bit signed value.
Cent #1: You read from indices "j + ...", but you write the data to "sIndex + j + ..." - effectively overwriting all data with the data from the first 4 frames. That's totally not what you want.
Cent #2: Still not sure about the sample values you posted. Is your input plausibly of an amplitude of within a few percent of the maximum possible 'loudness'? Also notice that e.g. the first sample (-8186112), when multiplied by 1.1, yields -9004723, which is beyond the range of a 24-bit signed value.
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Re: Unable to do simple audio transformations
Here's how I would probably draft it:
Code: Select all
static const uint32_t bytes_per_sample = 3;
static const uint32_t samples_per_frame = 2;
// Read a big-endian, signed 24-bit value from src
static int32_t read24(const uint8_t* const src) {
int32_t r = (int8_t)src[0]; // takes care of the sign-extension
r = (r<<8) | src[1];
r = (r<<8) | src[2];
return r;
}
// Write a signed 24-bit value as big-endian to dest
static void write24(const int32_t v, uint8_t* const dest) {
dest[0] = v >> 16;
dest[1] = v >> 8;
dest[2] = v;
}
void setAudioGain(uint8_t *buf, const size_t bLength, const float gain) {
uint32_t frames_left = bLength/(bytes_per_sample * samples_per_frame);
while(frames_left) {
int32_t left = read24(buf);
left = (int32_t)(left * gain);
write24(left, buf);
buf += bytes_per_sample;
int32_t right = read24(buf);
right = (int32_t)(right * gain);
write24(right, buf);
buf += bytes_per_sample;
frames_left -= 1;
}
}
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Re: Unable to do simple audio transformations
So, I've adapted your draft to my code, and there is more strange behavior.MicroController wrote: ↑Thu Oct 12, 2023 9:29 pmHere's how I would probably draft it:Code: Select all
static const uint32_t bytes_per_sample = 3; static const uint32_t samples_per_frame = 2; // Read a big-endian, signed 24-bit value from src static int32_t read24(const uint8_t* const src) { int32_t r = (int8_t)src[0]; // takes care of the sign-extension r = (r<<8) | src[1]; r = (r<<8) | src[2]; return r; } // Write a signed 24-bit value as big-endian to dest static void write24(const int32_t v, uint8_t* const dest) { dest[0] = v >> 16; dest[1] = v >> 8; dest[2] = v; } void setAudioGain(uint8_t *buf, const size_t bLength, const float gain) { uint32_t frames_left = bLength/(bytes_per_sample * samples_per_frame); while(frames_left) { int32_t left = read24(buf); left = (int32_t)(left * gain); write24(left, buf); buf += bytes_per_sample; int32_t right = read24(buf); right = (int32_t)(right * gain); write24(right, buf); buf += bytes_per_sample; frames_left -= 1; } }
When the gain is precisely of 0.5 or 1.5, there is sound coming out, but same as before it comes out saturated, even though I can hear myself, well a glitchy version.
If gain is 1.0, sound is normal, anything else and there is no sound at all (except for 0.5 and 1.5)
Here is the code that I wrote with your draft
- #define CONVERT8bits(n,o) (uint8_t)((n>>8*o)&0xFF)
- static int32_t read24(const uint8_t *buffer) {
- int32_t value = (int8_t)buffer[0];
- value = (value<<8) | buffer[1];
- value = (value<<8) | buffer[2];
- return value;
- }
- void setAudioGain(uint8_t *buffer, size_t bLength, float gain) {
- int32_t *sample = calloc(2,sizeof(int32_t)); // L : R
- for (uint32_t sIndex = 0; sIndex < bLength; sIndex += 6) {
- sample[Left] = read24(buffer+0);
- sample[Right] = read24(buffer+3);
- sample[Left] = (int32_t)(sample[Left] * gain);
- sample[Right] = (int32_t)(sample[Right] * gain);
- *((uint8_t *)(buffer + 0 )) = CONVERT8bits(sample[Left],2); // Left Channel MSB
- *((uint8_t *)(buffer + 1 )) = CONVERT8bits(sample[Left],1); // Left Channel
- *((uint8_t *)(buffer + 2 )) = CONVERT8bits(sample[Left],0); // Left Channel LSB
- *((uint8_t *)(buffer + 3 )) = CONVERT8bits(sample[Right],2); // Right Channel MSB
- *((uint8_t *)(buffer + 4 )) = CONVERT8bits(sample[Right],1); // Right Channel
- *((uint8_t *)(buffer + 5 )) = CONVERT8bits(sample[Right],0); // Right Channel LSB
- buffer += 6;
- }
- free(sample);
- }
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Re: Unable to do simple audio transformations
I don't know how and why, but I finally managed to do it.
Two months, I tried every witchery and randomly when clearing the bloated lines and the commented definitly not working ones, it works !
I thank both of you guys (MicroController and ESP_Sprite)
Here is the working code (for one channel), if you want more pm me, I'll be glad to help
Next step, an audio mixer.
Two months, I tried every witchery and randomly when clearing the bloated lines and the commented definitly not working ones, it works !
I thank both of you guys (MicroController and ESP_Sprite)
Here is the working code (for one channel), if you want more pm me, I'll be glad to help
- void setAudioGain(uint8_t *buffer, size_t bLength, double gain) {
- int32_t sample = 0;
- for (uint32_t sIndex = 0; sIndex < bLength; sIndex += 6) {
- sample = (int32_t)(((int8_t)buffer[sIndex + 2])<<16 | (buffer[sIndex + 1])<<8 | (buffer[sIndex + 0]));
- sample = (int32_t)((double)(sample) * gain);
- buffer[sIndex + 0] = sample >> 0;
- buffer[sIndex + 1] = sample >> 8;
- buffer[sIndex + 2] = sample >> 16;
- }
- }
-
- Posts: 1688
- Joined: Mon Oct 17, 2022 7:38 pm
- Location: Europe, Germany
Re: Unable to do simple audio transformations
Code: Select all
sample = (int32_t)(((int8_t)buffer[sIndex + 2])<<16 | (buffer[sIndex + 1])<<8 | (buffer[sIndex + 0]));
Btw, for performance reasons, I would definitely steer clear of using double, and generally would avoid/reduce floating point calculations, see also here. Fixed-point integer calculations are an alternative worth exploring.
Last edited by MicroController on Fri Oct 13, 2023 9:21 pm, edited 1 time in total.
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- Posts: 9
- Joined: Mon Oct 09, 2023 8:31 pm
Re: Unable to do simple audio transformations
Do you mean instead of doing that:MicroController wrote: ↑Fri Oct 13, 2023 8:27 pmSo the byte order actually was little-endian all alongCode: Select all
sample = (int32_t)(((int8_t)buffer[sIndex + 2])<<16 | (buffer[sIndex + 1])<<8 | (buffer[sIndex + 0]));
Btw, for performance reasons, I would definitely steer clear of using double, and generally would avoid/reduce floating point calculations, see also here. Fixed-point integer calculations are an alternative worth exploring.
Code: Select all
float gain = 2.00;
sample[Left] = (int32_t)((double)(sample[Left]) * gain);
Code: Select all
uint32_t gain = 200; // actual gain is -> gain/100
sample[Left] = (int32_t)((int64_t)(sample[Left]) * gain / 100);
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- Posts: 1688
- Joined: Mon Oct 17, 2022 7:38 pm
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Re: Unable to do simple audio transformations
Yes, exactly.
Usually one chooses a power of two as scale, e.g. 256=2^8, so that scaling can be handled via cheap bit-shifting instead of (integer) multiplication and division.
Like
One must always be aware that multiplications 'add' bitlengths and appropriate data types must be chosen for intermediate values.
However, 64-bit multiplication on a 32-bit machine may or may not be faster than using the FPU for a float multiplication; needs benchmarking. If the scale is <= 8 bits, no int64_t intermediate is needed and things should be faster than float.
Usually one chooses a power of two as scale, e.g. 256=2^8, so that scaling can be handled via cheap bit-shifting instead of (integer) multiplication and division.
Like
Code: Select all
const uint32_t scale_bits = 10; // Taking 10 bits as an example specifically because 24+10 bits won't fit into an int32_t
const uint32_t gain_factor = (uint32_t)(gain * (1<<scale_bits));
for (...) {
...
sample = (int32_t)(((int64_t)sample * gain_factor) >> scale_bits);
...
}
However, 64-bit multiplication on a 32-bit machine may or may not be faster than using the FPU for a float multiplication; needs benchmarking. If the scale is <= 8 bits, no int64_t intermediate is needed and things should be faster than float.
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Re: Unable to do simple audio transformations
For my opinion have 2 problem:
1) when inizialize i2s use a PHILIPS schema with this configuration (input and output):
void init_microphone(void)
{
i2s_chan_config_t chan_cfg = I2S_CHANNEL_DEFAULT_CONFIG(I2S_NUM_AUTO, I2S_ROLE_MASTER);
ESP_ERROR_CHECK(i2s_new_channel(&chan_cfg, NULL, &rx_handle));
i2s_std_config_t std_rx_cfg = {
.clk_cfg = I2S_STD_CLK_DEFAULT_CONFIG(CONFIG_EXAMPLE_MIC_SAMPLE_RATE),
.slot_cfg = I2S_STD_PHILIPS_SLOT_DEFAULT_CONFIG(CONFIG_EXAMPLE_MIC_BIT_SAMPLE, CONFIG_EXAMPLE_MIC_NUM_CHANNELS),
.gpio_cfg = {
.mclk = I2S_GPIO_UNUSED,
.bclk = CONFIG_EXAMPLE_I2S_MIC_BCLK_GPIO,
.ws = CONFIG_EXAMPLE_I2S_MIC_WS_GPIO,
.din = CONFIG_EXAMPLE_I2S_MIC_DATA_GPIO,
.invert_flags = {
.mclk_inv = false,
.bclk_inv = false,
.ws_inv = false,
},
},
};
ESP_ERROR_CHECK(i2s_channel_init_std_mode(rx_handle, &std_rx_cfg));
ESP_ERROR_CHECK(i2s_channel_enable(rx_handle));
}
Philips schema is perfect for edit buffer.
Second point is use 32bit data buffer
this function is for apply gain with test for top gain available.
void apply_volume_control(wave_info_t *wave_info, bool Normalize)
{
// Se Normalize è TRUE, imposta la scala del volume a 1.0 per normalizzare il segnale
float volume_scale = Normalize ? 1.0f : wave_info->volume_level / 9.0f;
// Trova il picco massimo del segnale e applica la trasformazione del volume in un solo ciclo
int16_t max_sample = 0;
for (size_t i = 0; i < wave_info->buffer_size / sizeof(int16_t); i++) {
// Step 1: Scala il volume in base all'impostazione utente o la normalizzazione
int32_t sample = (int32_t)(wave_info->buffer * volume_scale);
// Trova il picco massimo attuale
if (abs(sample) > max_sample) {
max_sample = abs(sample);
}
// Sovrascrive temporaneamente il campione nel buffer
wave_info->buffer = (int16_t)sample;
}
// Se Normalize è TRUE, calcola un fattore di amplificazione per normalizzare il segnale
float amplification_factor = 1.0f;
if (Normalize && max_sample > 0) {
// Normalizza il segnale audio facendo sì che il picco più alto raggiunga INT16_MAX
amplification_factor = (float)INT16_MAX / max_sample;
} else if (!Normalize && max_sample > 0) {
// Se non si sta normalizzando, applica il fattore di amplificazione solo se il segnale è basso
amplification_factor = (float)INT16_MAX / max_sample;
}
// Applica l'amplificazione e il soft clipping
for (size_t i = 0; i < wave_info->buffer_size / sizeof(int16_t); i++) {
int32_t sample = wave_info->buffer;
// Step 2: Applica l'amplificazione
sample = (int32_t)(sample * amplification_factor);
// Step 3: Applica il soft clipping per evitare distorsioni
if (sample > INT16_MAX) {
sample = INT16_MAX - ((sample - INT16_MAX) / 2); // Soft clipping sopra INT16_MAX
} else if (sample < INT16_MIN) {
sample = INT16_MIN + ((sample - INT16_MIN) / 2); // Soft clipping sotto INT16_MIN
}
// Step 4: Aggiorna il buffer con il campione modificato
wave_info->buffer = (int16_t)sample;
}
}
i have same problem
in this moment i try to convert a 48khz to a 44100hz
Best Regards.
1) when inizialize i2s use a PHILIPS schema with this configuration (input and output):
void init_microphone(void)
{
i2s_chan_config_t chan_cfg = I2S_CHANNEL_DEFAULT_CONFIG(I2S_NUM_AUTO, I2S_ROLE_MASTER);
ESP_ERROR_CHECK(i2s_new_channel(&chan_cfg, NULL, &rx_handle));
i2s_std_config_t std_rx_cfg = {
.clk_cfg = I2S_STD_CLK_DEFAULT_CONFIG(CONFIG_EXAMPLE_MIC_SAMPLE_RATE),
.slot_cfg = I2S_STD_PHILIPS_SLOT_DEFAULT_CONFIG(CONFIG_EXAMPLE_MIC_BIT_SAMPLE, CONFIG_EXAMPLE_MIC_NUM_CHANNELS),
.gpio_cfg = {
.mclk = I2S_GPIO_UNUSED,
.bclk = CONFIG_EXAMPLE_I2S_MIC_BCLK_GPIO,
.ws = CONFIG_EXAMPLE_I2S_MIC_WS_GPIO,
.din = CONFIG_EXAMPLE_I2S_MIC_DATA_GPIO,
.invert_flags = {
.mclk_inv = false,
.bclk_inv = false,
.ws_inv = false,
},
},
};
ESP_ERROR_CHECK(i2s_channel_init_std_mode(rx_handle, &std_rx_cfg));
ESP_ERROR_CHECK(i2s_channel_enable(rx_handle));
}
Philips schema is perfect for edit buffer.
Second point is use 32bit data buffer
this function is for apply gain with test for top gain available.
void apply_volume_control(wave_info_t *wave_info, bool Normalize)
{
// Se Normalize è TRUE, imposta la scala del volume a 1.0 per normalizzare il segnale
float volume_scale = Normalize ? 1.0f : wave_info->volume_level / 9.0f;
// Trova il picco massimo del segnale e applica la trasformazione del volume in un solo ciclo
int16_t max_sample = 0;
for (size_t i = 0; i < wave_info->buffer_size / sizeof(int16_t); i++) {
// Step 1: Scala il volume in base all'impostazione utente o la normalizzazione
int32_t sample = (int32_t)(wave_info->buffer * volume_scale);
// Trova il picco massimo attuale
if (abs(sample) > max_sample) {
max_sample = abs(sample);
}
// Sovrascrive temporaneamente il campione nel buffer
wave_info->buffer = (int16_t)sample;
}
// Se Normalize è TRUE, calcola un fattore di amplificazione per normalizzare il segnale
float amplification_factor = 1.0f;
if (Normalize && max_sample > 0) {
// Normalizza il segnale audio facendo sì che il picco più alto raggiunga INT16_MAX
amplification_factor = (float)INT16_MAX / max_sample;
} else if (!Normalize && max_sample > 0) {
// Se non si sta normalizzando, applica il fattore di amplificazione solo se il segnale è basso
amplification_factor = (float)INT16_MAX / max_sample;
}
// Applica l'amplificazione e il soft clipping
for (size_t i = 0; i < wave_info->buffer_size / sizeof(int16_t); i++) {
int32_t sample = wave_info->buffer;
// Step 2: Applica l'amplificazione
sample = (int32_t)(sample * amplification_factor);
// Step 3: Applica il soft clipping per evitare distorsioni
if (sample > INT16_MAX) {
sample = INT16_MAX - ((sample - INT16_MAX) / 2); // Soft clipping sopra INT16_MAX
} else if (sample < INT16_MIN) {
sample = INT16_MIN + ((sample - INT16_MIN) / 2); // Soft clipping sotto INT16_MIN
}
// Step 4: Aggiorna il buffer con il campione modificato
wave_info->buffer = (int16_t)sample;
}
}
i have same problem
in this moment i try to convert a 48khz to a 44100hz
Best Regards.
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