diff --git a/backend/cpp/audio-cpp/CMakeLists.txt b/backend/cpp/audio-cpp/CMakeLists.txt
index 53eb4eea6..729dc70e2 100644
--- a/backend/cpp/audio-cpp/CMakeLists.txt
+++ b/backend/cpp/audio-cpp/CMakeLists.txt
@@ -88,6 +88,7 @@ add_executable(${TARGET}
     grpc-server.cpp
     model_options.cpp
     capability_routing.cpp
+    audio_units.cpp
 )
 
 target_include_directories(${TARGET} PRIVATE
diff --git a/backend/cpp/audio-cpp/audio_units.cpp b/backend/cpp/audio-cpp/audio_units.cpp
new file mode 100644
index 000000000..6e15cce33
--- /dev/null
+++ b/backend/cpp/audio-cpp/audio_units.cpp
@@ -0,0 +1,66 @@
+#include "audio_units.h"
+
+#include <algorithm>
+#include <cmath>
+
+namespace audiocpp_backend {
+
+std::int64_t samples_to_nanoseconds(std::int64_t samples, int sample_rate) {
+    if (sample_rate <= 0) {
+        return 0;
+    }
+    // Split into whole seconds plus a remainder so the intermediate product
+    // cannot overflow on long recordings, and so rates like 44100 stay exact.
+    const std::int64_t rate = static_cast<std::int64_t>(sample_rate);
+    const std::int64_t whole_seconds = samples / rate;
+    const std::int64_t remainder = samples % rate;
+    return whole_seconds * 1000000000LL + (remainder * 1000000000LL) / rate;
+}
+
+float samples_to_seconds(std::int64_t samples, int sample_rate) {
+    if (sample_rate <= 0) {
+        return 0.0f;
+    }
+    return static_cast<float>(static_cast<double>(samples) /
+                              static_cast<double>(sample_rate));
+}
+
+std::int64_t seconds_to_samples(double seconds, int sample_rate) {
+    if (sample_rate <= 0 || seconds <= 0.0) {
+        return 0;
+    }
+    return static_cast<std::int64_t>(seconds * static_cast<double>(sample_rate));
+}
+
+std::vector<float> s16le_to_f32(const std::string &bytes) {
+    std::vector<float> samples;
+    const size_t count = bytes.size() / 2;
+    samples.reserve(count);
+    for (size_t i = 0; i < count; ++i) {
+        const auto low = static_cast<unsigned char>(bytes[i * 2]);
+        const auto high = static_cast<unsigned char>(bytes[i * 2 + 1]);
+        const auto raw = static_cast<std::int16_t>(
+            static_cast<std::uint16_t>(low) |
+            (static_cast<std::uint16_t>(high) << 8));
+        samples.push_back(static_cast<float>(raw) / 32768.0f);
+    }
+    return samples;
+}
+
+std::string f32_to_s16le(const std::vector<float> &samples) {
+    std::string bytes;
+    bytes.reserve(samples.size() * 2);
+    for (const float sample : samples) {
+        const float clamped = std::max(-1.0f, std::min(1.0f, sample));
+        // 32767 rather than 32768 so +1.0 saturates at INT16_MAX instead of
+        // overflowing to INT16_MIN.
+        const auto value =
+            static_cast<std::int16_t>(std::lround(clamped * 32767.0f));
+        const auto raw = static_cast<std::uint16_t>(value);
+        bytes.push_back(static_cast<char>(raw & 0xFF));
+        bytes.push_back(static_cast<char>((raw >> 8) & 0xFF));
+    }
+    return bytes;
+}
+
+} // namespace audiocpp_backend
diff --git a/backend/cpp/audio-cpp/audio_units.h b/backend/cpp/audio-cpp/audio_units.h
new file mode 100644
index 000000000..23f716829
--- /dev/null
+++ b/backend/cpp/audio-cpp/audio_units.h
@@ -0,0 +1,33 @@
+#pragma once
+
+// Time and sample-format conversion between audio.cpp's runtime types (sample
+// indices, float PCM) and LocalAI's proto types. Standard library only.
+//
+// LocalAI uses three different time units:
+//   TranscriptSegment / TranscriptWord start,end : int64 nanoseconds
+//   VADSegment start,end                         : float seconds
+//   DiarizeSegment start,end                     : float seconds
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+namespace audiocpp_backend {
+
+// Returns 0 when sample_rate is not positive rather than dividing by zero.
+// Uses integer arithmetic so 44.1 kHz does not lose precision.
+std::int64_t samples_to_nanoseconds(std::int64_t samples, int sample_rate);
+
+float samples_to_seconds(std::int64_t samples, int sample_rate);
+
+// Negative seconds clamp to 0.
+std::int64_t seconds_to_samples(double seconds, int sample_rate);
+
+// Decodes little-endian signed 16-bit PCM. A trailing odd byte is dropped.
+std::vector<float> s16le_to_f32(const std::string &bytes);
+
+// Encodes to little-endian signed 16-bit PCM, clamping to [-1, 1] first so an
+// overshooting sample saturates instead of wrapping to the opposite sign.
+std::string f32_to_s16le(const std::vector<float> &samples);
+
+} // namespace audiocpp_backend
diff --git a/backend/cpp/audio-cpp/audio_units_test.cpp b/backend/cpp/audio-cpp/audio_units_test.cpp
new file mode 100644
index 000000000..3b56911a0
--- /dev/null
+++ b/backend/cpp/audio-cpp/audio_units_test.cpp
@@ -0,0 +1,118 @@
+// Unit tests for audio_units. Standard library only. The harness compiles this
+// as a single translation unit, so the implementation is included directly.
+
+#include "audio_units.cpp"
+
+#include <cmath>
+#include <cstdio>
+#include <string>
+#include <vector>
+
+static int failures = 0;
+
+static void check(bool ok, const std::string &name) {
+    if (!ok) {
+        failures++;
+        fprintf(stderr, "FAIL: %s\n", name.c_str());
+    } else {
+        fprintf(stderr, "ok:   %s\n", name.c_str());
+    }
+}
+
+static bool close_to(float a, float b, float tol) { return std::fabs(a - b) <= tol; }
+
+using namespace audiocpp_backend;
+
+static void test_nanoseconds() {
+    // LocalAI TranscriptSegment/TranscriptWord times are nanoseconds
+    // (Go reads them as time.Duration).
+    check(samples_to_nanoseconds(16000, 16000) == 1000000000LL, "1s at 16k is 1e9 ns");
+    check(samples_to_nanoseconds(8000, 16000) == 500000000LL, "0.5s at 16k");
+    check(samples_to_nanoseconds(0, 16000) == 0, "zero samples is zero ns");
+    check(samples_to_nanoseconds(1000, 0) == 0, "zero sample rate yields zero, not UB");
+    // 44.1 kHz must not lose precision to float arithmetic.
+    check(samples_to_nanoseconds(44100, 44100) == 1000000000LL, "1s at 44.1k");
+    check(samples_to_nanoseconds(22050, 44100) == 500000000LL, "0.5s at 44.1k");
+    // The cases above all land on values a float happens to hold exactly, so
+    // they do not actually rule float arithmetic out. These do:
+    // a fraction that does not divide evenly, and a duration whose magnitude
+    // exceeds a float's 24-bit mantissa at nanosecond resolution.
+    check(samples_to_nanoseconds(44099, 44100) == 999977324LL,
+          "44.1k fraction is exact, not rounded through a float");
+    check(samples_to_nanoseconds(44100LL * 3600, 44100) == 3600000000000LL,
+          "one hour at 44.1k is exact to the nanosecond");
+    // A naive samples * 1e9 would overflow int64 here; the split into whole
+    // seconds plus a remainder is what keeps this correct.
+    check(samples_to_nanoseconds(44100LL * 360000, 44100) == 360000000000000LL,
+          "100 hours at 44.1k does not overflow");
+    // Double arithmetic is close enough to pass everything above, but still
+    // truncates this one a nanosecond short. Integer division does not.
+    check(samples_to_nanoseconds(4004, 8000) == 500500000LL,
+          "0.5005s at 8k is exact to the nanosecond");
+}
+
+static void test_seconds() {
+    check(close_to(samples_to_seconds(24000, 24000), 1.0f, 1e-6f), "1s at 24k");
+    check(close_to(samples_to_seconds(12000, 24000), 0.5f, 1e-6f), "0.5s at 24k");
+    check(close_to(samples_to_seconds(100, 0), 0.0f, 1e-6f), "zero sample rate is 0s");
+    check(seconds_to_samples(1.0, 16000) == 16000, "1s to samples at 16k");
+    check(seconds_to_samples(0.5, 16000) == 8000, "0.5s to samples at 16k");
+    check(seconds_to_samples(1.0, 0) == 0, "zero sample rate yields zero samples");
+    check(seconds_to_samples(-1.0, 16000) == 0, "negative seconds clamps to zero");
+}
+
+static void test_s16le_round_trip() {
+    const std::vector<float> original = {0.0f, 0.5f, -0.5f, 1.0f, -1.0f};
+    const std::string encoded = f32_to_s16le(original);
+    check(encoded.size() == original.size() * 2, "two bytes per sample");
+
+    const std::vector<float> decoded = s16le_to_f32(encoded);
+    check(decoded.size() == original.size(), "round trip keeps the sample count");
+    for (size_t i = 0; i < original.size(); ++i) {
+        // 16-bit quantisation: one LSB is ~3.05e-5. Guard the index so a short
+        // result reports a named failure instead of aborting the whole suite.
+        check(i < decoded.size() && close_to(decoded[i], original[i], 1e-4f),
+              "round trip preserves sample " + std::to_string(i));
+    }
+}
+
+static void test_s16le_endianness() {
+    // 0.5 encodes to 16384 = 0x4000, little endian is 0x00 0x40.
+    const std::string encoded = f32_to_s16le({0.5f});
+    check(encoded.size() == 2, "one sample is two bytes");
+    check(static_cast<unsigned char>(encoded[0]) == 0x00, "low byte first");
+    check(static_cast<unsigned char>(encoded[1]) == 0x40, "high byte second");
+}
+
+static void test_s16le_clamping() {
+    // Values outside [-1, 1] must clamp, not wrap around to the opposite sign.
+    const std::string encoded = f32_to_s16le({2.0f, -2.0f});
+    const std::vector<float> decoded = s16le_to_f32(encoded);
+    check(decoded.size() == 2, "two samples survive clamping");
+    check(decoded.size() > 0 && decoded[0] > 0.99f,
+          "positive overshoot clamps to full scale");
+    check(decoded.size() > 1 && decoded[1] < -0.99f,
+          "negative overshoot clamps to full scale");
+}
+
+static void test_s16le_odd_length() {
+    // A truncated frame must drop the dangling byte rather than read past it.
+    const std::string odd(5, '\0');
+    check(s16le_to_f32(odd).size() == 2, "odd byte count drops the trailing byte");
+    check(s16le_to_f32(std::string()).empty(), "empty input yields no samples");
+}
+
+int main() {
+    test_nanoseconds();
+    test_seconds();
+    test_s16le_round_trip();
+    test_s16le_endianness();
+    test_s16le_clamping();
+    test_s16le_odd_length();
+    if (failures) {
+        fprintf(stderr, "%d check(s) failed\n", failures);
+        return 1;
+    }
+    fprintf(stderr, "all audio_units checks passed\n");
+    return 0;
+}