Files
shairport-sync/loudness.c
T
Mike Brady b405c0b896 Version 5.0 Major Release.
New Features:
Multi-Channel and High-Resolution Audio Support
48,000 frames per second ("48k") operation.
48k lossless stereo support.
5.1 and 7.1 surround sound support.
Multi-channel and multi-rate operation on ALSA, PipeWire, PulseAudio, FreeBSD, stdout and Unix pipe output backends.
Automatic Audio Format Selection
Flexible and controllable output format selection.
Automatic rate, sample format, and channel count selection.
Full FFmpeg Integration
Support for transcoding.
Advanced resampling capabilities.
New audio format support.
Enhanced Resampling
New vernier resampling and interpolation method optimized for low-power CPUs.
Better performance on resource-constrained devices.

Convolution and Loudness Enhancements:
Convolution system is now multithreaded and works on stereo and multichannel audio at 48k and 44.1k.
Multiple impulse response (IR) files can now be provided via convolution_ir_files setting.
New convolution_thread_pool_size setting for multithreaded processing (defaults to 1).
Loudness processing now works with stereo and multichannel audio at 48k and 44.1k.
Updated to the most recent HiFi-LoFi FFT convolver.

MQTT Enhancements:
Added new publish_retain boolean option. When enabled, published MQTT messages have the retain flag set, so the MQTT broker stores the last message per topic and new subscribers receive the most recent value immediately. Thanks to lululombard for PR #2142.

D-Bus Enhancements:
Added new dbus_default_message_bus command-line argument (can be system or session) to set the default message bus for both D-Bus native service and MPRIS service.

Performance Improvements:
Enhanced compatibility with AirPlay 2 AutoMix and Smart Tracklists resulting in less unexplained track skipping.
Better operation on low-power devices down to Raspberry Pi B.
Improved efficiency on embedded systems.
Enhanced timestamp handling for better synchronization.
Improved sync error calculation.
Rebuilt buffered audio processor for cleaner handling of immediate and deferred flush requests.

Docker Enhancements:
Reduced Docker image sizes with slimmed-down FFmpeg library.
Removed dhclient from Docker images for smaller footprint.

Bug Fixes:
Fixed MQTT warning on service startup: "Could not establish a mqtt connection". The startup script now correctly states that the mosquitto service is required. Thanks to Hugo Villeneuve for PR #2137.
Fixed compatibility with mbedtls library version 3.4+ (present on recent Linux versions). Thanks to Christian Beier for finding and fixing the bug.
Fixed PulseAudio backend so that PA_ERR_NODATA returns "No latency data yet". Thanks to Vladimir Shakov for the report and fix.
Ensured old flush requests are deleted when a new play session starts. Thanks to saujanyashah for the report.
Fixed format warnings on 64-bit and 32-bit systems
Removed compilation warnings on 32-bit builds
Improved argument checking for debug(), inform(), warn() and die() functions
Fixed "daemon" typos throughout codebase. Thanks to Chris Boot for PR #1981.
Added warning if a convolution impulse response file cannot be read due to bad path or permissions

Build System Improvements:
Unified service file with variable substitution for Avahi support, making it easier to add future service dependencies. Thanks to Hugo Villeneuve.
Network interface selection now only considers interfaces that are up, running and not loopback interfaces. Thanks to Carl Johnson for the suggestion.
Configuration File Changes and Deprecations

New settings: convolution_ir_files (replaces convolution_ir_file), convolution_enabled (replaces convolution), convolution_max_length_in_seconds (replaces convolution_max_length), loudness_enabled (replaces loudness).
New convolution_thread_pool_size setting (defaults to 1).
Deprecated settings: convolution_ir_file, convolution, convolution_max_length, loudness.
Corresponding D-Bus methods and properties have been updated.

Deprecation Notice:
The Jack Audio and soundio backends are deprecated and will be removed in a future release. Consider using the updated PipeWire backend instead.

Documentation Updates
Updated BUILD.md with latest build instructions.
Updated AIRPLAY2.md with feature information.
Enhanced convolution and loudness documentation.

Maintenance:
Fixed FFmpeg deprecation warnings.
Bumped actions/checkout from 6.0.1 to 6.0.2.
Bumped docker/login-action from 3.6.0 to 3.7.0.
Bumped docker/build-push-action from 6.13.0 to 6.15.0.
Bumped docker/setup-qemu-action from 3.4.0 to 3.6.0.
Bumped docker/setup-buildx-action from 3.9.0 to 3.10.0.
2026-02-13 15:17:40 +00:00

100 lines
3.3 KiB
C

#include "loudness.h"
#include "common.h"
#include <math.h>
#define MAXCHANNELS 8
// loudness_processor_dynamic loudness_r;
// loudness_processor_dynamic loudness_l;
loudness_processor_dynamic loudness_filters[MAXCHANNELS];
// if the rate or the loudness_reference_volume_db change, recalculate the
// loudness volume parameters
static int loudness_fix_volume_parameter = 0;
static unsigned int loudness_rate_parameter = 0;
static float loudness_volume_reference_parameter = 0.0;
static loudness_processor_static lps = {0.0, 0.0, 0.0, 0.0, 0.0};
void _loudness_set_volume(loudness_processor_static *p, float volume, unsigned int sample_rate) {
float gain = -(volume - config.loudness_reference_volume_db) * 0.5;
if (gain < 0) {
gain = 0;
}
debug(2, "Volume: %.1f dB - Loudness gain @10Hz: %.1f dB", volume, gain);
float Fc = 10.0;
float Q = 0.5;
// Formula from http://www.earlevel.com/main/2011/01/02/biquad-formulas/
float Fs = sample_rate * 1.0;
float K = tan(M_PI * Fc / Fs);
float V = pow(10.0, gain / 20.0);
float norm = 1 / (1 + 1 / Q * K + K * K);
p->a0 = (1 + V / Q * K + K * K) * norm;
p->a1 = 2 * (K * K - 1) * norm;
p->a2 = (1 - V / Q * K + K * K) * norm;
p->b1 = p->a1;
p->b2 = (1 - 1 / Q * K + K * K) * norm;
}
float loudness_process(loudness_processor_dynamic *p, float i0) {
float o0 = lps.a0 * i0 + lps.a1 * p->i1 + lps.a2 * p->i2 - lps.b1 * p->o1 - lps.b2 * p->o2;
p->o2 = p->o1;
p->o1 = o0;
p->i2 = p->i1;
p->i1 = i0;
return o0;
}
void loudness_update(rtsp_conn_info *conn) {
// first, see if loudness can be enabled
int do_loudness = config.loudness_enabled;
if ((config.output->parameters != NULL) && (config.output->parameters()->volume_range != NULL)) {
do_loudness = 0; // if we are using external (usually hardware) volume controls.
}
if (do_loudness) {
// check the volume parameters
if ((conn->fix_volume != loudness_fix_volume_parameter) ||
(conn->input_rate != loudness_rate_parameter) ||
(config.loudness_reference_volume_db != loudness_volume_reference_parameter)) {
debug(1, "update loudness parameters");
float new_volume = 20 * log10((double)conn->fix_volume / 65536);
_loudness_set_volume(&lps, new_volume, conn->input_rate);
// _loudness_set_volume(&loudness_r, new_volume, conn->input_rate);
loudness_fix_volume_parameter = conn->fix_volume;
loudness_rate_parameter = conn->input_rate;
loudness_volume_reference_parameter = config.loudness_reference_volume_db;
}
}
conn->do_loudness = do_loudness;
}
void loudness_process_blocks(float *fbufs, unsigned int channel_length,
unsigned int number_of_channels, float gain) {
unsigned int channel_number, sample_index;
float *sample_pointer = fbufs;
for (channel_number = 0; channel_number < number_of_channels; channel_number++) {
for (sample_index = 0; sample_index < channel_length; sample_index++) {
*sample_pointer = loudness_process(&loudness_filters[channel_number], *sample_pointer * gain);
sample_pointer++;
}
}
}
void loudness_reset() {
unsigned int i;
for (i = 0; i < MAXCHANNELS; i++) {
loudness_filters[i].i1 = 0.0;
loudness_filters[i].i2 = 0.0;
loudness_filters[i].o1 = 0.0;
loudness_filters[i].o2 = 0.0;
}
}