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-rw-r--r--dom/media/gtest/TestAudioPacketizer.cpp163
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diff --git a/dom/media/gtest/TestAudioPacketizer.cpp b/dom/media/gtest/TestAudioPacketizer.cpp
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+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <stdint.h>
+#include <math.h>
+#include <memory>
+#include "../AudioPacketizer.h"
+#include "gtest/gtest.h"
+
+using namespace mozilla;
+
+template <typename T>
+class AutoBuffer {
+ public:
+ explicit AutoBuffer(size_t aLength) { mStorage = new T[aLength]; }
+ ~AutoBuffer() { delete[] mStorage; }
+ T* Get() { return mStorage; }
+
+ private:
+ T* mStorage;
+};
+
+int16_t Sequence(int16_t* aBuffer, uint32_t aSize, uint32_t aStart = 0) {
+ uint32_t i;
+ for (i = 0; i < aSize; i++) {
+ aBuffer[i] = aStart + i;
+ }
+ return aStart + i;
+}
+
+void IsSequence(std::unique_ptr<int16_t[]> aBuffer, uint32_t aSize,
+ uint32_t aStart = 0) {
+ for (uint32_t i = 0; i < aSize; i++) {
+ ASSERT_TRUE(aBuffer[i] == static_cast<int64_t>(aStart + i))
+ << "Buffer is not a sequence at offset " << i << std::endl;
+ }
+ // Buffer is a sequence.
+}
+
+void Zero(std::unique_ptr<int16_t[]> aBuffer, uint32_t aSize) {
+ for (uint32_t i = 0; i < aSize; i++) {
+ ASSERT_TRUE(aBuffer[i] == 0)
+ << "Buffer is not null at offset " << i << std::endl;
+ }
+}
+
+double sine(uint32_t aPhase) { return sin(aPhase * 2 * M_PI * 440 / 44100); }
+
+TEST(AudioPacketizer, Test)
+{
+ for (int16_t channels = 1; channels < 2; channels++) {
+ // Test that the packetizer returns zero on underrun
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ for (int16_t i = 0; i < 10; i++) {
+ std::unique_ptr<int16_t[]> out(ap.Output());
+ Zero(std::move(out), 441);
+ }
+ }
+ // Simple test, with input/output buffer size aligned on the packet size,
+ // alternating Input and Output calls.
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ int16_t seqEnd = 0;
+ for (int16_t i = 0; i < 10; i++) {
+ AutoBuffer<int16_t> b(441 * channels);
+ int16_t prevEnd = seqEnd;
+ seqEnd = Sequence(b.Get(), channels * 441, prevEnd);
+ ap.Input(b.Get(), 441);
+ std::unique_ptr<int16_t[]> out(ap.Output());
+ IsSequence(std::move(out), 441 * channels, prevEnd);
+ }
+ }
+ // Simple test, with input/output buffer size aligned on the packet size,
+ // alternating two Input and Output calls.
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ int16_t seqEnd = 0;
+ for (int16_t i = 0; i < 10; i++) {
+ AutoBuffer<int16_t> b(441 * channels);
+ AutoBuffer<int16_t> b1(441 * channels);
+ int16_t prevEnd0 = seqEnd;
+ seqEnd = Sequence(b.Get(), 441 * channels, prevEnd0);
+ int16_t prevEnd1 = seqEnd;
+ seqEnd = Sequence(b1.Get(), 441 * channels, seqEnd);
+ ap.Input(b.Get(), 441);
+ ap.Input(b1.Get(), 441);
+ std::unique_ptr<int16_t[]> out(ap.Output());
+ std::unique_ptr<int16_t[]> out2(ap.Output());
+ IsSequence(std::move(out), 441 * channels, prevEnd0);
+ IsSequence(std::move(out2), 441 * channels, prevEnd1);
+ }
+ }
+ // Input/output buffer size not aligned on the packet size,
+ // alternating two Input and Output calls.
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ int16_t prevEnd = 0;
+ int16_t prevSeq = 0;
+ for (int16_t i = 0; i < 10; i++) {
+ AutoBuffer<int16_t> b(480 * channels);
+ AutoBuffer<int16_t> b1(480 * channels);
+ prevSeq = Sequence(b.Get(), 480 * channels, prevSeq);
+ prevSeq = Sequence(b1.Get(), 480 * channels, prevSeq);
+ ap.Input(b.Get(), 480);
+ ap.Input(b1.Get(), 480);
+ std::unique_ptr<int16_t[]> out(ap.Output());
+ std::unique_ptr<int16_t[]> out2(ap.Output());
+ IsSequence(std::move(out), 441 * channels, prevEnd);
+ prevEnd += 441 * channels;
+ IsSequence(std::move(out2), 441 * channels, prevEnd);
+ prevEnd += 441 * channels;
+ }
+ printf("Available: %d\n", ap.PacketsAvailable());
+ }
+
+ // "Real-life" test case: streaming a sine wave through a packetizer, and
+ // checking that we have the right output.
+ // 128 is, for example, the size of a Web Audio API block, and 441 is the
+ // size of a webrtc.org packet when the sample rate is 44100 (10ms)
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ AutoBuffer<int16_t> b(128 * channels);
+ uint32_t phase = 0;
+ uint32_t outPhase = 0;
+ for (int16_t i = 0; i < 1000; i++) {
+ for (int32_t j = 0; j < 128; j++) {
+ for (int32_t c = 0; c < channels; c++) {
+ // int16_t sinewave at 440Hz/44100Hz sample rate
+ b.Get()[j * channels + c] = (2 << 14) * sine(phase);
+ }
+ phase++;
+ }
+ ap.Input(b.Get(), 128);
+ while (ap.PacketsAvailable()) {
+ std::unique_ptr<int16_t[]> packet(ap.Output());
+ for (uint32_t k = 0; k < ap.mPacketSize; k++) {
+ for (int32_t c = 0; c < channels; c++) {
+ ASSERT_TRUE(packet[k * channels + c] ==
+ static_cast<int16_t>(((2 << 14) * sine(outPhase))));
+ }
+ outPhase++;
+ }
+ }
+ }
+ }
+ // Test that clearing the packetizer empties it and starts returning zeros.
+ {
+ AudioPacketizer<int16_t, int16_t> ap(441, channels);
+ AutoBuffer<int16_t> b(440 * channels);
+ Sequence(b.Get(), 440 * channels);
+ ap.Input(b.Get(), 440);
+ EXPECT_EQ(ap.FramesAvailable(), 440U);
+ ap.Clear();
+ EXPECT_EQ(ap.FramesAvailable(), 0U);
+ EXPECT_TRUE(ap.Empty());
+ std::unique_ptr<int16_t[]> out(ap.Output());
+ Zero(std::move(out), 441);
+ }
+ }
+}