X-Git-Url: https://code.octet-stream.net/m17rt/blobdiff_plain/e67ea96c8a3d7c23ba29c6ed91ddb451927176a1..61c96b473ce4d5dec8c46235595c519e7f868cd5:/m17core/src/modem.rs?ds=sidebyside diff --git a/m17core/src/modem.rs b/m17core/src/modem.rs index af36ab4..35a23a7 100644 --- a/m17core/src/modem.rs +++ b/m17core/src/modem.rs @@ -1,5 +1,10 @@ -use crate::decode::{parse_lsf, parse_stream, sync_burst_correlation, SyncBurst, SYNC_THRESHOLD}; -use crate::protocol::Frame; +use crate::decode::{ + parse_lsf, parse_packet, parse_stream, sync_burst_correlation, SyncBurst, SYNC_THRESHOLD, +}; +use crate::encode::{ + encode_lsf, encode_packet, encode_stream, generate_end_of_transmission, generate_preamble, +}; +use crate::protocol::{Frame, LsfFrame, PacketFrame, StreamFrame}; use crate::shaping::RRC_48K; use log::debug; @@ -131,7 +136,11 @@ impl Demodulator for SoftDemodulator { } } SyncBurst::Packet => { - debug!("Found PACKET at sample {} diff {}", start_sample, c.diff) + debug!("Found PACKET at sample {} diff {}", start_sample, c.diff); + if let Some(frame) = parse_packet(&pkt_samples) { + self.suppress = 191 * 10; + return Some(Frame::Packet(frame)); + } } } } @@ -152,6 +161,308 @@ impl Default for SoftDemodulator { } } +pub trait Modulator { + /// Inform the modulator how many samples remain pending for output and latency updates. + /// + /// For the buffer between `Modulator` and the process which is supplying samples to the + /// output sound card, `samples_to_play` is the number of bytes which the modulator has + /// provided that have not yet been picked up, and `capacity` is the maximum size we can + /// fill this particular buffer, i.e., maximum number of samples. + /// + /// Furthermore we attempt to track and account for the latency between the output + /// soundcard callback, and when those samples will actually be on the wire. CPAL helpfully + /// gives us an estimate. The latest estimate of latency is converted to a duration in terms + /// of number of samples and provided as `output_latency`. Added to this is the current + /// number of samples we expect remain to be processed from the last read. + /// + /// Call this whenever bytes have been read out of the buffer. + fn update_output_buffer( + &mut self, + samples_to_play: usize, + capacity: usize, + output_latency: usize, + ); + + /// Supply the next frame available from the TNC, if it was requested. + fn provide_next_frame(&mut self, frame: Option); + + /// Calculate and write out output samples for the soundcard. + /// + /// Returns the number of bytes valid in `out`. Should generally be called in a loop until + /// 0 is returned. + fn read_output_samples(&mut self, out: &mut [i16]) -> usize; + + /// Run the modulator and receive actions to process. + /// + /// Should be called in a loop until it returns `None`. + fn run(&mut self) -> Option; +} + +pub enum ModulatorAction { + /// If true, once all samples have been exhausted output should revert to equilibrium. + /// + /// If false, failure to pick up enough samples for output sound card is an underrun error. + SetIdle(bool), + + /// Check with the TNC if there is a frame available for transmission. + /// + /// Call `next_frame()` with either the next frame, or `None` if TNC has nothing more to offer. + GetNextFrame, + + /// Modulator wishes to send samples to the output buffer - call `read_output_samples`. + ReadOutput, + + /// Advise the TNC that we will complete sending End Of Transmission after the given number of + /// samples has elapsed, and therefore PTT should be deasserted at this time. + TransmissionWillEnd(usize), +} + +/// Frames for transmission, emitted by the TNC and received by the Modulator. +/// +/// The TNC is responsible for all timing decisions, making sure these frames are emitted in the +/// correct order, breaks between transmissions, PTT and CSMA. If the modulator is given a +/// `ModulatorFrame` value, its job is to transmit it immediately by modulating it into the output +/// buffer, or otherwise directly after any previously-supplied frames. +/// +/// The modulator controls the rate at which frames are drawn out of the TNC. Therefore if the send +/// rate is too high (or there is too much channel activity) then the effect of this backpressure is +/// that the TNC's internal queues will overflow and it will either discard earlier frames in the +/// current stream, or some packets awaiting transmission. +pub enum ModulatorFrame { + Preamble { + /// TNC's configured TxDelay setting, increments of 10ms. + /// + /// TNC fires PTT and it's up to modulator to apply the setting, taking advantage of whatever + /// buffering already exists in the sound card to reduce the artificial delay. + tx_delay: u8, + }, + Lsf(LsfFrame), + Stream(StreamFrame), + Packet(PacketFrame), + // TODO: BertFrame + EndOfTransmission, +} + +pub struct SoftModulator { + // TODO: 2000 was overflowing around EOT, track down why + /// Next modulated frame to output - 1920 samples for 40ms frame plus 80 for ramp-down + next_transmission: [i16; 4000], + /// How much of next_transmission should in fact be transmitted + next_len: usize, + /// How much of next_transmission has been read out + next_read: usize, + /// How many pending zero samples to emit to align start of preamble with PTT taking effect + tx_delay_padding: usize, + + /// Do we need to update idle state? + update_idle: bool, + /// What is that idle status? + idle: bool, + + /// Do we need to calculate a transmission end time? + /// + /// (True after we encoded an EOT.) We will wait until we get a precise timing update. + calculate_tx_end: bool, + /// Do we need to report a transmission end time? + /// + /// This is a duration expressed in number of samples. + report_tx_end: Option, + + /// Circular buffer of most recently output samples for calculating the RRC filtered value. + /// + /// This should naturally degrade to an oldest value plus 80 zeroes after an EOT. + filter_win: [f32; 81], + /// Current position in filter_win + filter_cursor: usize, + + /// Should we ask the TNC for another frame. True after each call to update_output_buffer. + try_get_frame: bool, + + /// Expected delay beyond the buffer to reach the DAC + output_latency: usize, + /// Number of samples we have placed in the buffer for the output soundcard not yet picked up. + samples_in_buf: usize, + /// Total size to which the output buffer is allowed to expand. + buf_capacity: usize, +} + +impl SoftModulator { + pub fn new() -> Self { + Self { + next_transmission: [0i16; 4000], + next_len: 0, + next_read: 0, + tx_delay_padding: 0, + update_idle: true, + idle: true, + calculate_tx_end: false, + report_tx_end: None, + filter_win: [0f32; 81], + filter_cursor: 0, + try_get_frame: false, + output_latency: 0, + samples_in_buf: 0, + buf_capacity: 0, + } + } + + fn push_sample(&mut self, dibit: f32) { + // TODO: 48 kHz assumption again + for i in 0..10 { + // Right now we are encoding everything as 1.0-scaled dibit floats + // This is a bit silly but it will do for a minute + // Max possible gain from the RRC filter with upsampling is about 0.462 + // Let's bump everything to a baseline of 16383 / 0.462 = 35461 + // For normal signals this yields roughly 0.5 magnitude which is plenty + if i == 0 { + self.filter_win[self.filter_cursor] = dibit * 35461.0; + } else { + self.filter_win[self.filter_cursor] = 0.0; + } + self.filter_cursor = (self.filter_cursor + 1) % 81; + let mut out: f32 = 0.0; + for i in 0..81 { + let filter_idx = (self.filter_cursor + i) % 81; + out += RRC_48K[i] * self.filter_win[filter_idx]; + } + self.next_transmission[self.next_len] = out as i16; + self.next_len += 1; + } + } + + fn request_frame_if_space(&mut self) { + if self.buf_capacity - self.samples_in_buf >= 2000 { + self.try_get_frame = true; + } + } +} + +impl Modulator for SoftModulator { + fn update_output_buffer( + &mut self, + samples_to_play: usize, + capacity: usize, + output_latency: usize, + ) { + //log::debug!("modulator update_output_buffer {samples_to_play} {capacity} {output_latency}"); + self.output_latency = output_latency; + self.buf_capacity = capacity; + self.samples_in_buf = samples_to_play; + + if self.calculate_tx_end { + self.calculate_tx_end = false; + // next_transmission should already have been read out to the buffer by now + // so we don't have to consider it + self.report_tx_end = Some(self.samples_in_buf + self.output_latency); + } + + self.request_frame_if_space(); + } + + fn provide_next_frame(&mut self, frame: Option) { + let Some(frame) = frame else { + self.try_get_frame = false; + return; + }; + + self.next_len = 0; + self.next_read = 0; + + match frame { + ModulatorFrame::Preamble { tx_delay } => { + // TODO: Stop assuming 48 kHz everywhere. 24 kHz should be fine too. + let tx_delay_samples = tx_delay as usize * 480; + // TxDelay and output latency have the same effect - account for whichever is bigger. + // We want our sound card DAC hitting preamble right when PTT fully engages. + // The modulator calls the shots here - TNC hands over Preamble and asserts PTT, then + // waits to be told when transmission will be complete. This estimate will not be + // made and delivered until we generate the EOT frame. + self.tx_delay_padding = tx_delay_samples.max(self.output_latency); + + // We should be starting from a filter_win of zeroes + // Transmission is effectively smeared by 80 taps and we'll capture that in EOT + for dibit in generate_preamble() { + self.push_sample(dibit); + } + } + ModulatorFrame::Lsf(lsf_frame) => { + for dibit in encode_lsf(&lsf_frame) { + self.push_sample(dibit); + } + } + ModulatorFrame::Stream(stream_frame) => { + for dibit in encode_stream(&stream_frame) { + self.push_sample(dibit); + } + } + ModulatorFrame::Packet(packet_frame) => { + for dibit in encode_packet(&packet_frame) { + self.push_sample(dibit); + } + } + ModulatorFrame::EndOfTransmission => { + for dibit in generate_end_of_transmission() { + self.push_sample(dibit); + } + for _ in 0..80 { + // This is not a real symbol value + // However we want to flush the filter + self.push_sample(0f32); + } + self.calculate_tx_end = true; + } + } + } + + fn read_output_samples(&mut self, out: &mut [i16]) -> usize { + let mut written = 0; + + // if we have pre-TX padding to accommodate TxDelay then expend that first + if self.tx_delay_padding > 0 { + let len = out.len().max(self.tx_delay_padding); + self.tx_delay_padding -= len; + for x in 0..len { + out[x] = 0; + } + written += len; + } + + // then follow it with whatever might be left in next_transmission + let next_remaining = self.next_len - self.next_read; + if next_remaining > 0 { + let len = (out.len() - written).min(next_remaining); + out[written..(written + len)] + .copy_from_slice(&self.next_transmission[self.next_read..(self.next_read + len)]); + self.next_read += len; + written += len; + } + + written + } + + fn run(&mut self) -> Option { + // Time-sensitive for accuracy, so handle it first + if let Some(end) = self.report_tx_end.take() { + return Some(ModulatorAction::TransmissionWillEnd(end)); + } + + if self.next_read < self.next_len { + return Some(ModulatorAction::ReadOutput); + } + + if self.update_idle { + self.update_idle = false; + return Some(ModulatorAction::SetIdle(self.idle)); + } + + if self.try_get_frame { + return Some(ModulatorAction::GetNextFrame); + } + + None + } +} + #[derive(Debug)] pub(crate) struct DecodeCandidate { burst: SyncBurst,