mirror of
https://github.com/esphome/esphome.git
synced 2025-03-20 09:38:18 +00:00
68 lines
2.2 KiB
C++
68 lines
2.2 KiB
C++
|
#include "audio.h"
|
||
|
|
|
||
|
|
namespace esphome {
|
||
|
|
namespace audio {
|
||
|
|
|
||
|
|
// Euclidean's algorithm for finding the greatest common divisor
|
||
|
|
static uint32_t gcd(uint32_t a, uint32_t b) {
|
||
|
|
while (b != 0) {
|
||
|
|
uint32_t t = b;
|
||
|
|
b = a % b;
|
||
|
|
a = t;
|
||
|
|
}
|
||
|
|
return a;
|
||
|
|
}
|
||
|
|
|
||
|
|
AudioStreamInfo::AudioStreamInfo(uint8_t bits_per_sample, uint8_t channels, uint32_t sample_rate)
|
||
|
|
: bits_per_sample_(bits_per_sample), channels_(channels), sample_rate_(sample_rate) {
|
||
|
|
this->ms_sample_rate_gcd_ = gcd(1000, this->sample_rate_);
|
||
|
|
this->bytes_per_sample_ = (this->bits_per_sample_ + 7) / 8;
|
||
|
|
}
|
||
|
|
|
||
|
|
uint32_t AudioStreamInfo::frames_to_microseconds(uint32_t frames) const {
|
||
|
|
return (frames * 1000000 + (this->sample_rate_ >> 1)) / this->sample_rate_;
|
||
|
|
}
|
||
|
|
|
||
|
|
uint32_t AudioStreamInfo::frames_to_milliseconds_with_remainder(uint32_t *total_frames) const {
|
||
|
|
uint32_t unprocessable_frames = *total_frames % (this->sample_rate_ / this->ms_sample_rate_gcd_);
|
||
|
|
uint32_t frames_for_ms_calculation = *total_frames - unprocessable_frames;
|
||
|
|
|
||
|
|
uint32_t playback_ms = (frames_for_ms_calculation * 1000) / this->sample_rate_;
|
||
|
|
*total_frames = unprocessable_frames;
|
||
|
|
return playback_ms;
|
||
|
|
}
|
||
|
|
|
||
|
|
bool AudioStreamInfo::operator==(const AudioStreamInfo &rhs) const {
|
||
|
|
return (this->bits_per_sample_ == rhs.get_bits_per_sample()) && (this->channels_ == rhs.get_channels()) &&
|
||
|
|
(this->sample_rate_ == rhs.get_sample_rate());
|
||
|
|
}
|
||
|
|
|
||
|
|
const char *audio_file_type_to_string(AudioFileType file_type) {
|
||
|
|
switch (file_type) {
|
||
|
|
#ifdef USE_AUDIO_FLAC_SUPPORT
|
||
|
|
case AudioFileType::FLAC:
|
||
|
|
return "FLAC";
|
||
|
|
#endif
|
||
|
|
#ifdef USE_AUDIO_MP3_SUPPORT
|
||
|
|
case AudioFileType::MP3:
|
||
|
|
return "MP3";
|
||
|
|
#endif
|
||
|
|
case AudioFileType::WAV:
|
||
|
|
return "WAV";
|
||
|
|
default:
|
||
|
|
return "unknown";
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
void scale_audio_samples(const int16_t *audio_samples, int16_t *output_buffer, int16_t scale_factor,
|
||
|
|
size_t samples_to_scale) {
|
||
|
|
// Note the assembly dsps_mulc function has audio glitches if the input and output buffers are the same.
|
||
|
|
for (int i = 0; i < samples_to_scale; i++) {
|
||
|
|
int32_t acc = (int32_t) audio_samples[i] * (int32_t) scale_factor;
|
||
|
|
output_buffer[i] = (int16_t) (acc >> 15);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
} // namespace audio
|
||
|
|
} // namespace esphome
|