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https://gitlab.freedesktop.org/pipewire/pipewire.git
synced 2025-11-02 09:01:50 -05:00
module-loopback: improve channel remapping support
Convert capture and playback streams to any global channel position that has been set. This can then be used to implement an upmix sink and add such an example config.
This commit is contained in:
Wim Taymans
parent
088d741cda
commit
1a83ce0908
1 changed files with 109 additions and 28 deletions
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@ -67,6 +67,16 @@
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* prefixed with 'input.' and 'output.' to generate a capture and playback
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* stream node.name respectively.
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*
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* ## Channel handling
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*
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* Channels from the capture stream are copied, in order, to the channels of the
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* output stream. The remaining streams are ignored (when capture has more channels)
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* or filled with silence (when playback has more channels).
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*
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* When a global channel position is set, both capture and playback will be converted
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* to and from this common channel layout. This can be used to implement up or
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* downmixing loopback sinks/sources.
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*
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* ## Example configuration of a virtual sink
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*
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* This Virtual sink routes stereo input to the rear channels of a 7.1 sink.
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@ -104,18 +114,49 @@
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* context.modules = [
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* { name = libpipewire-module-loopback
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* args = {
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* node.description = "Scarlett Focusrite Line 1"
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* capture.props = {
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* audio.position = [ FL ]
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* stream.dont-remix = true
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* node.target = "alsa_input.usb-Focusrite_Scarlett_Solo_USB_Y7ZD17C24495BC-00.analog-stereo"
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* node.passive = true
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* }
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* playback.props = {
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* node.name = "SF_mono_in_1"
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* media.class = "Audio/Source"
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* audio.position = [ MONO ]
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* }
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* node.description = "Scarlett Focusrite Line 1"
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* capture.props = {
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* audio.position = [ FL ]
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* stream.dont-remix = true
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* node.target = "alsa_input.usb-Focusrite_Scarlett_Solo_USB_Y7ZD17C24495BC-00.analog-stereo"
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* node.passive = true
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* }
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* playback.props = {
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* node.name = "SF_mono_in_1"
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* media.class = "Audio/Source"
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* audio.position = [ MONO ]
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* }
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* }
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* }
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* ]
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*\endcode
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*
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* ## Example configuration of an upmix sink
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*
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* This Virtual sink has 2 input channels and 6 output channels. It will perform upmixing
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* using the PSD algorithm on the playback stream.
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*
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*\code{.unparsed}
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* context.modules = [
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* { name = libpipewire-module-loopback
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* args = {
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* node.description = "Upmix Sink"
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* audio.position = [ FL FR ]
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* capture.props = {
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* node.name = "effect_input.upmix"
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* media.class = Audio/Sink
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* }
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* playback.props = {
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* node.name = "effect_output.upmix"
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* audio.position = [ FL FR RL RR FC LFE ]
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* node.passive = true
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* stream.dont-remix = true
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* channelmix.upmix = true
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* channelmix.upmix-method = psd
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* channelmix.lfe-cutoff = 150
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* channelmix.fc-cutoff = 12000
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* channelmix.rear-delay = 12.0
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* }
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* }
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* }
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* ]
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@ -171,6 +212,8 @@ struct impl {
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struct spa_hook core_proxy_listener;
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struct spa_hook core_listener;
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struct spa_audio_info_raw info;
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struct pw_properties *capture_props;
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struct pw_stream *capture;
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struct spa_hook capture_listener;
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@ -185,8 +228,11 @@ struct impl {
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unsigned int recalc_delay:1;
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struct spa_io_position *position;
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struct spa_audio_info_raw info;
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uint32_t target_rate;
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uint32_t rate;
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uint32_t target_channels;
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uint32_t channels;
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float target_delay;
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struct spa_ringbuffer buffer;
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uint8_t *buffer_data;
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@ -345,6 +391,47 @@ static void param_tag_changed(struct impl *impl, const struct spa_pod *param,
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pw_stream_update_params(other, params, 1);
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}
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static void param_format_changed(struct impl *impl, const struct spa_pod *param,
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struct pw_stream *stream, bool capture)
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{
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struct spa_audio_info_raw info;
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spa_zero(info);
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if (param != NULL) {
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if (spa_format_audio_raw_parse(param, &info) < 0 ||
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info.channels == 0 ||
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info.channels > SPA_AUDIO_MAX_CHANNELS)
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return;
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if ((impl->info.format != 0 && impl->info.format != info.format) ||
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(impl->info.rate != 0 && impl->info.rate != info.rate) ||
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(impl->info.channels != 0 &&
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(impl->info.channels != info.channels ||
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memcmp(impl->info.position, info.position,
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info.channels * sizeof(uint32_t)) != 0))) {
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uint8_t buffer[1024];
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struct spa_pod_builder b;
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const struct spa_pod *params[1];
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if (impl->info.format != 0)
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info.format = impl->info.format;
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if (impl->info.rate != 0)
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info.rate = impl->info.rate;
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if (impl->info.channels != 0) {
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info.channels = impl->info.channels;
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memcpy(info.position, impl->info.position, sizeof(info.position));
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}
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spa_pod_builder_init(&b, buffer, sizeof(buffer));
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params[0] = spa_format_audio_raw_build(&b, SPA_PARAM_Format, &info);
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pw_stream_update_params(stream, params, 1);
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}
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}
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if (capture) {
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impl->target_rate = info.rate;
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impl->target_channels = info.channels;
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}
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}
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static void recalculate_buffer(struct impl *impl)
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{
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if (impl->target_delay > 0.0f) {
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@ -352,7 +439,7 @@ static void recalculate_buffer(struct impl *impl)
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void *data;
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impl->buffer_size = (delay + (1u<<15)) * 4;
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data = realloc(impl->buffer_data, impl->buffer_size * impl->info.channels);
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data = realloc(impl->buffer_data, impl->buffer_size * impl->channels);
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if (data == NULL) {
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pw_log_warn("can't allocate delay buffer, delay disabled: %m");
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impl->buffer_size = 0;
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@ -388,12 +475,13 @@ static void stream_state_changed(void *data, enum pw_stream_state old,
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break;
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case PW_STREAM_STATE_STREAMING:
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{
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uint32_t target = impl->info.rate;
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uint32_t target = impl->target_rate;
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if (target == 0)
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target = impl->position ?
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impl->position->clock.target_rate.denom : DEFAULT_RATE;
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if (impl->rate != target) {
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if (impl->rate != target || impl->target_channels != impl->channels) {
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impl->rate = target;
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impl->channels = impl->target_channels;
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recalculate_buffer(impl);
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}
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break;
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@ -409,19 +497,8 @@ static void capture_param_changed(void *data, uint32_t id, const struct spa_pod
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switch (id) {
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case SPA_PARAM_Format:
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{
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struct spa_audio_info_raw info;
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spa_zero(info);
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if (param != NULL) {
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if (spa_format_audio_raw_parse(param, &info) < 0 ||
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info.channels == 0 ||
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info.channels > SPA_AUDIO_MAX_CHANNELS)
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return;
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}
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impl->rate = 0;
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impl->info = info;
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param_format_changed(impl, param, impl->capture, true);
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break;
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}
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case SPA_PARAM_Latency:
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param_latency_changed(impl, param, impl->playback);
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break;
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@ -464,6 +541,9 @@ static void playback_param_changed(void *data, uint32_t id, const struct spa_pod
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struct impl *impl = data;
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switch (id) {
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case SPA_PARAM_Format:
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param_format_changed(impl, param, impl->playback, false);
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break;
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case SPA_PARAM_Latency:
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param_latency_changed(impl, param, impl->capture);
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break;
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@ -747,6 +827,7 @@ int pipewire__module_init(struct pw_impl_module *module, const char *args)
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if (pw_properties_get(impl->playback_props, PW_KEY_NODE_DESCRIPTION) == NULL)
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pw_properties_set(impl->playback_props, PW_KEY_NODE_DESCRIPTION, str);
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parse_audio_info(props, &impl->info);
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parse_audio_info(impl->capture_props, &impl->capture_info);
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parse_audio_info(impl->playback_props, &impl->playback_info);
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