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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(Buffer buffer);
private uint period_samples;
private uint period_size;
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 <contact@dino.im>");
}
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 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 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 <contact@dino.im>");
}
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 setup_native(void* native);
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 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();
setup_native(native);
return true;
}
public override bool start() {
native = init_native(150);
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 (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(Buffer buffer) {
if (buffer.pts != uint64.MAX) {
last_reverse = buffer.pts;
}
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);
}
}
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;
}
}
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