From e0837a17b5a497476d67237c768836e51f8a4ce7 Mon Sep 17 00:00:00 2001
From: "Steinar H. Gunderson" <sgunderson@bigfoot.com>
Date: Fri, 29 Jul 2016 12:32:16 +0200
Subject: [PATCH] Make FakeCapture capable of outputting audio (a simple sine
 tone).

---
 bmusb/fake_capture.h | 11 ++++++++--
 fake_capture.cpp     | 48 +++++++++++++++++++++++++++++++++++++++-----
 2 files changed, 52 insertions(+), 7 deletions(-)

diff --git a/bmusb/fake_capture.h b/bmusb/fake_capture.h
index 48dca06..17982ea 100644
--- a/bmusb/fake_capture.h
+++ b/bmusb/fake_capture.h
@@ -12,7 +12,7 @@ namespace bmusb {
 class FakeCapture : public CaptureInterface
 {
 public:
-	FakeCapture(unsigned width, unsigned height, unsigned fps, unsigned audio_frequency, int card_index);
+	FakeCapture(unsigned width, unsigned height, unsigned fps, unsigned audio_sample_frequency, int card_index, bool has_audio = false);
 	~FakeCapture();
 
 	// CaptureInterface.
@@ -79,10 +79,17 @@ public:
 
 private:
 	void producer_thread_func();
+	void make_tone(int32_t *out, unsigned num_stereo_samples);
 
-	unsigned width, height, fps, audio_frequency;
+	unsigned width, height, fps, audio_sample_frequency;
 	uint8_t y, cb, cr;
 
+	// sin(2 * pi * f / F) and similar for cos. Used for fast sine generation.
+	// Zero when no audio.
+	float audio_sin = 0.0f, audio_cos = 0.0f;
+	float audio_real = 0.0f, audio_imag = 0.0f;  // Current state of the audio phaser.
+	float audio_ref_level;
+
 	bool has_dequeue_callbacks = false;
 	std::function<void()> dequeue_init_callback = nullptr;
 	std::function<void()> dequeue_cleanup_callback = nullptr;
diff --git a/fake_capture.cpp b/fake_capture.cpp
index 432cf42..961700e 100644
--- a/fake_capture.cpp
+++ b/fake_capture.cpp
@@ -86,8 +86,8 @@ void memset4(uint8_t *s, const uint8_t c[4], size_t n)
 
 }  // namespace
 
-FakeCapture::FakeCapture(unsigned width, unsigned height, unsigned fps, unsigned audio_frequency, int card_index)
-	: width(width), height(height), fps(fps), audio_frequency(audio_frequency)
+FakeCapture::FakeCapture(unsigned width, unsigned height, unsigned fps, unsigned audio_sample_frequency, int card_index, bool has_audio)
+	: width(width), height(height), fps(fps), audio_sample_frequency(audio_sample_frequency)
 {
 	char buf[256];
 	snprintf(buf, sizeof(buf), "Fake card %d", card_index + 1);
@@ -96,6 +96,15 @@ FakeCapture::FakeCapture(unsigned width, unsigned height, unsigned fps, unsigned
 	y = ys[card_index % NUM_COLORS];
 	cb = cbs[card_index % NUM_COLORS];
 	cr = crs[card_index % NUM_COLORS];
+
+	if (has_audio) {
+		audio_ref_level = pow(10.0f, -23.0f / 20.0f) * (1u << 31);  // -23 dBFS (EBU R128 level).
+
+		float freq = 440.0 * pow(2.0, card_index / 12.0);
+		sincosf(2 * M_PI * freq / audio_sample_frequency, &audio_sin, &audio_cos);
+		audio_real = audio_ref_level;
+		audio_imag = 0.0f;
+	}
 }
 
 FakeCapture::~FakeCapture()
@@ -254,9 +263,16 @@ void FakeCapture::producer_thread_func()
 
 		FrameAllocator::Frame audio_frame = audio_frame_allocator->alloc_frame();
 		if (audio_frame.data != nullptr) {
-			assert(audio_frame.size >= 2 * sizeof(uint32_t) * audio_frequency / fps);
-			audio_frame.len = 2 * sizeof(uint32_t) * audio_frequency / fps;
-			memset(audio_frame.data, 0, audio_frame.len);
+			const unsigned num_stereo_samples = audio_sample_frequency / fps;
+			assert(audio_frame.size >= 2 * sizeof(int32_t) * num_stereo_samples);
+			audio_frame.len = 2 * sizeof(int32_t) * num_stereo_samples;
+
+			if (audio_sin == 0.0f) {
+				// Silence.
+				memset(audio_frame.data, 0, audio_frame.len);
+			} else {
+				make_tone((int32_t *)audio_frame.data, num_stereo_samples);
+			}
 		}
 
 		frame_callback(timecode++,
@@ -268,4 +284,26 @@ void FakeCapture::producer_thread_func()
 	}
 }
 
+void FakeCapture::make_tone(int32_t *out, unsigned num_stereo_samples)
+{
+	int32_t *ptr = out;
+	float r = audio_real, i = audio_imag;
+	for (unsigned sample_num = 0; sample_num < num_stereo_samples; ++sample_num) {
+		int32_t s = lrintf(r);
+		*ptr++ = s;
+		*ptr++ = s;
+
+		// Rotate the phaser by one sample.
+		float new_r = r * audio_cos - i * audio_sin;
+		float new_i = r * audio_sin + i * audio_cos;
+		r = new_r;
+		i = new_i;
+	}
+
+	// Periodically renormalize to counteract precision issues.
+	double corr = audio_ref_level / hypot(r, i);
+	audio_real = r * corr;
+	audio_imag = i * corr;
+}
+
 }  // namespace bmusb
-- 
2.39.2