Audio capture and streaming app for webOS 5/6

Captures the TV's audio and sends it out over several transports. The
primary one is HyperHDR: RTP/L16 to a host-side loopback device, since
HyperHDR has no network audio input of its own. A second route renders
the spectrum on the TV and sends FlatBuffers images to port 19400
instead, for setups where touching the host's sound config is not an
option.

  native/       the service: capture backends (PulseAudio, ALSA, exec,
                test tone, all dlopen-based), DSP, and one file per sink
  frontend/     D-pad driven UI at a fixed 1920x1080
  servicefiles/ native service manifest plus the boot script
  host/         RTP receiver and the loopback installer for the HyperHDR
                machine
  tools/        build/package, asset generation, Homebrew Channel
                manifest, on-TV probe
  test/         host-side suites: FlatBuffers and RTP verified against
                real decoders, the engine end to end, the page in jsdom

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>

test/rtp_send.c

@@ -0,0 +1,77 @@
+// Harness for verify_rtp.py: opens the real HyperHDR RTP sink, points it at a
+// port on the loopback and writes a deterministic ramp through it. The Python
+// side receives the datagrams and checks that what comes out of the wire is
+// exactly what went in.
+//
+//   rtp_send <port> <blocks> [frames-per-block] [sap]
+
+#include "../native/src/common/audio.h"
+#include "../native/src/common/json.h"
+#include "../native/src/common/log.h"
+#include "../native/src/dsp.h"
+#include "../native/src/sinks/sink.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+// Must match sample_at() in verify_rtp.py.
+static int16_t sample_at(long index)
+{
+    return (int16_t)((index * 251) % 65536 - 32768);
+}
+
+int main(int argc, char** argv)
+{
+    if (argc < 3) {
+        fprintf(stderr, "usage: %s <port> <blocks> [frames]\n", argv[0]);
+        return 2;
+    }
+
+    int port = atoi(argv[1]);
+    int blocks = atoi(argv[2]);
+    int frames = argc > 3 ? atoi(argv[3]) : AUDIO_BLOCK_FRAMES;
+    bool sap = argc > 4 && strcmp(argv[4], "sap") == 0;
+
+    log_set_level(LOG_ERROR);
+
+    char cfg_text[256];
+    snprintf(cfg_text, sizeof(cfg_text),
+        "{\"hyperhdr\":{\"host\":\"127.0.0.1\",\"port\":%d,"
+        "\"multicast\":false,\"sapAnnounce\":%s}}",
+        port, sap ? "true" : "false");
+
+    json_value_t* cfg = json_parse(cfg_text);
+    if (!cfg) {
+        fprintf(stderr, "bad config\n");
+        return 1;
+    }
+
+    audio_format_t fmt = { .rate = 48000, .channels = 2 };
+    char err[256] = { 0 };
+    sink_t* sink = sink_open("hyperhdr", cfg, &fmt, err, sizeof(err));
+    if (!sink) {
+        fprintf(stderr, "sink_open failed: %s\n", err);
+        json_free(cfg);
+        return 1;
+    }
+
+    int16_t* pcm = malloc((size_t)frames * fmt.channels * sizeof(int16_t));
+    dsp_levels_t levels;
+    memset(&levels, 0, sizeof(levels));
+
+    long index = 0;
+    for (int b = 0; b < blocks; b++) {
+        for (int i = 0; i < frames * fmt.channels; i++)
+            pcm[i] = sample_at(index++);
+        sink->write(sink, pcm, frames, &levels);
+    }
+
+    printf("%ld\n", index); // samples written, for the receiver to expect
+    fflush(stdout);
+
+    free(pcm);
+    sink_close(sink);
+    json_free(cfg);
+    return 0;
+}