From dba577f39c489b9db29960f2bf5cc78ac4f29d7b Mon Sep 17 00:00:00 2001 From: Marvin W Date: Sun, 4 Oct 2026 14:42:16 +0200 Subject: Move OMEMO, OpenPGP and RTP plugins into main/libdino --- main/src/plugins/rtp/voice_processor.vala | 209 ++++++++++++++++++++++++++++++ 1 file changed, 209 insertions(+) create mode 100644 main/src/plugins/rtp/voice_processor.vala (limited to 'main/src/plugins/rtp/voice_processor.vala') diff --git a/main/src/plugins/rtp/voice_processor.vala b/main/src/plugins/rtp/voice_processor.vala new file mode 100644 index 00000000..064684c5 --- /dev/null +++ b/main/src/plugins/rtp/voice_processor.vala @@ -0,0 +1,209 @@ +using Gst; + +namespace Dino.Plugins.Rtp { +public static extern Buffer adjust_to_running_time(Base.Transform transform, Buffer buf); +} + +public class Dino.Plugins.Rtp.EchoProbe : Audio.Filter { + private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + public Audio.Info audio_info { get; private set; } + public signal void on_new_buffer(owned Buffer buffer); + public signal void on_new_delay(int delay); + private uint period_samples; + private uint period_size; + public int delay { get; private set; } + private Base.Adapter adapter = new Base.Adapter(); + + static construct { + add_static_pad_template(sink_template); + add_static_pad_template(src_template); + set_static_metadata("Acoustic Echo Canceller probe", "Generic/Audio", "Gathers playback buffers for echo cancellation", "Dino Team "); + } + + construct { + set_passthrough(true); + } + + public override bool setup(Audio.Info info) { + audio_info = info; + period_samples = info.rate / 100; // 10ms buffers + period_size = period_samples * info.bpf; + return true; + } + + public override bool src_event(owned Event event) { + Query query = new Query.latency(); + if (event.type == EventType.LATENCY && srcpad != null && srcpad.query(query)) { + ClockTime upstream_latency; + query.parse_latency(null, out upstream_latency, null); + int delay = this.delay; + if (upstream_latency != CLOCK_TIME_NONE) { + delay = (int) (upstream_latency / MSECOND); + } else { + delay = 0; + } + if (delay != this.delay) { + debug("Delay adjusted from %ms to %dms", this.delay, delay); + this.delay = delay; + on_new_delay(delay); + } + } + return base.src_event((owned) event); + } + + public override FlowReturn transform_ip(Buffer buf) { + lock (adapter) { + adapter.push(adjust_to_running_time(this, buf)); + while (adapter.available() > period_size) { + on_new_buffer(adapter.take_buffer(period_size)); + } + } + return FlowReturn.OK; + } + + public override bool stop() { + adapter.clear(); + return true; + } +} + +public class Dino.Plugins.Rtp.VoiceProcessor : Audio.Filter { + private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}}; + public Audio.Info audio_info { get; private set; } + private ulong process_outgoing_buffer_handler_id; + private ulong process_stream_delay_handler_id; + private uint adjust_delay_timeout_id; + private uint period_samples; + private uint period_size; + private Base.Adapter adapter = new Base.Adapter(); + private EchoProbe? echo_probe; + private Audio.StreamVolume? stream_volume; + private ClockTime last_reverse; + private void* native; + + static construct { + add_static_pad_template(sink_template); + add_static_pad_template(src_template); + set_static_metadata("Voice Processor (AGC, AEC, filters, etc.)", "Generic/Audio", "Pre-processes voice with WebRTC Audio Processing Library", "Dino Team "); + } + + construct { + set_passthrough(false); + } + + public VoiceProcessor(EchoProbe? echo_probe = null, Audio.StreamVolume? stream_volume = null) { + this.echo_probe = echo_probe; + this.stream_volume = stream_volume; + } + + private static extern void* init_native(int stream_delay); + private static extern void destroy_native(void* native); + private static extern void analyze_reverse_stream(void* native, Audio.Info info, Buffer buffer); + private static extern void process_stream(void* native, Audio.Info info, Buffer buffer); + private static extern void adjust_stream_delay(void* native); + private static extern void set_stream_delay(void* native, int delay); + private static extern void notify_gain_level(void* native, int gain_level); + private static extern int get_suggested_gain_level(void* native); + private static extern bool get_stream_has_voice(void* native); + + public override bool setup(Audio.Info info) { + debug("VoiceProcessor.setup(%s)", info.to_caps().to_string()); + audio_info = info; + period_samples = info.rate / 100; // 10ms buffers + period_size = period_samples * info.bpf; + adapter.clear(); + return true; + } + + public override bool start() { + native = init_native(echo_probe.delay); + if (process_outgoing_buffer_handler_id == 0 && echo_probe != null) { + process_outgoing_buffer_handler_id = echo_probe.on_new_buffer.connect(process_outgoing_buffer); + } + if (process_stream_delay_handler_id == 0 && echo_probe != null) { + process_stream_delay_handler_id = echo_probe.on_new_delay.connect(process_stream_delay); + } + if (stream_volume == null && sinkpad.get_peer() != null && sinkpad.get_peer().get_parent_element() is Audio.StreamVolume) { + stream_volume = sinkpad.get_peer().get_parent_element() as Audio.StreamVolume; + } + return true; + } + + private bool adjust_delay() { + if (native != null) { + adjust_stream_delay(native); + return Source.CONTINUE; + } else { + adjust_delay_timeout_id = 0; + return Source.REMOVE; + } + } + + private void process_outgoing_buffer(owned Buffer buffer) { + if (buffer.pts != uint64.MAX) { + last_reverse = buffer.pts; + } + if (native != null) { + buffer = (Buffer) buffer.make_writable(); + analyze_reverse_stream(native, echo_probe.audio_info, buffer); + } + if (adjust_delay_timeout_id == 0 && echo_probe != null) { + adjust_delay_timeout_id = Timeout.add(1000, adjust_delay); + } + } + + private void process_stream_delay(int stream_delay) { + if (native != null) { + set_stream_delay(native, stream_delay); + } + } + + public override FlowReturn submit_input_buffer(bool is_discont, Buffer input) { + lock (adapter) { + if (is_discont) { + adapter.clear(); + } + adapter.push(adjust_to_running_time(this, input)); + } + return FlowReturn.OK; + } + + public override FlowReturn generate_output(out Buffer output_buffer) { + lock (adapter) { + if (adapter.available() >= period_size) { + output_buffer = (Gst.Buffer) adapter.take_buffer(period_size).make_writable(); + int old_gain_level = 0; + if (stream_volume != null) { + old_gain_level = (int) (stream_volume.get_volume(Audio.StreamVolumeFormat.LINEAR) * 255.0); + notify_gain_level(native, old_gain_level); + } + process_stream(native, audio_info, output_buffer); + if (stream_volume != null) { + int new_gain_level = get_suggested_gain_level(native); + if (old_gain_level != new_gain_level) { + debug("Gain: %i -> %i", old_gain_level, new_gain_level); + stream_volume.set_volume(Audio.StreamVolumeFormat.LINEAR, ((double)new_gain_level) / 255.0); + } + } + } + } + return FlowReturn.OK; + } + + public override bool stop() { + if (process_outgoing_buffer_handler_id != 0) { + echo_probe.disconnect(process_outgoing_buffer_handler_id); + process_outgoing_buffer_handler_id = 0; + } + if (adjust_delay_timeout_id != 0) { + Source.remove(adjust_delay_timeout_id); + adjust_delay_timeout_id = 0; + } + adapter.clear(); + destroy_native(native); + native = null; + return true; + } +} \ No newline at end of file -- cgit v1.3