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-rw-r--r--audio_utils/tests/spdif_tests.cpp152
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diff --git a/audio_utils/tests/spdif_tests.cpp b/audio_utils/tests/spdif_tests.cpp
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+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <array>
+#include <climits>
+#include <math.h>
+#include <memory>
+#include <string.h>
+
+#include <gtest/gtest.h>
+
+#include <audio_utils/spdif/SPDIFEncoder.h>
+
+using namespace android;
+
+class MySPDIFEncoder : public SPDIFEncoder {
+public:
+
+ explicit MySPDIFEncoder(audio_format_t format)
+ : SPDIFEncoder(format)
+ {
+ }
+ // Defaults to AC3 format. Was in original API.
+ MySPDIFEncoder() = default;
+
+ ssize_t writeOutput( const void* /* buffer */, size_t numBytes ) override {
+ mOutputSizeBytes = numBytes;
+ return numBytes;
+ }
+
+ FrameScanner *getFramer() const { return mFramer; }
+ size_t getByteCursor() const { return mByteCursor; }
+ size_t getPayloadBytesPending() const { return mPayloadBytesPending; }
+ size_t getBurstBufferSizeBytes() const { return mBurstBufferSizeBytes; }
+
+ size_t mOutputSizeBytes = 0;
+};
+
+// This is the beginning of the file voice1-48k-64kbps-15s.ac3
+static const uint8_t sVoice1ch48k_AC3[] = {
+ 0x0b, 0x77, 0x44, 0xcd, 0x08, 0x40, 0x2f, 0x84, 0x29, 0xca, 0x6e, 0x44, 0xa4, 0xfd, 0xce, 0xf7,
+ 0xc9, 0x9f, 0x3e, 0x74, 0xfa, 0x01, 0x0a, 0xda, 0xb3, 0x3e, 0xb0, 0x95, 0xf2, 0x5a, 0xef, 0x9e
+};
+
+// This is the beginning of the file channelcheck_48k6ch.eac3
+static const uint8_t sChannel6ch48k_EAC3[] = {
+ 0x0b, 0x77, 0x01, 0xbf, 0x3f, 0x85, 0x7f, 0xe8, 0x1e, 0x40, 0x82, 0x10, 0x00, 0x00, 0x00, 0x01,
+ 0x00, 0x00, 0x00, 0x03, 0xfc, 0x60, 0x80, 0x7e, 0x59, 0x00, 0xfc, 0xf3, 0xcf, 0x01, 0xf9, 0xe7
+};
+
+static const uint8_t sZeros[32] = { 0 };
+
+static constexpr int kBytesPerOutputFrame = 2 * sizeof(int16_t); // stereo
+
+TEST(audio_utils_spdif, SupportedFormats)
+{
+ ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_PCM_FLOAT));
+ ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_PCM_16_BIT));
+ ASSERT_FALSE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_MP3));
+
+ ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_AC3));
+ ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_E_AC3));
+ ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_DTS));
+ ASSERT_TRUE(SPDIFEncoder::isFormatSupported(AUDIO_FORMAT_DTS_HD));
+}
+
+TEST(audio_utils_spdif, ScanAC3)
+{
+ MySPDIFEncoder encoder(AUDIO_FORMAT_AC3);
+ FrameScanner *scanner = encoder.getFramer();
+ // It should recognize the valid AC3 header.
+ int i = 0;
+ while (i < 5) {
+ ASSERT_FALSE(scanner->scan(sVoice1ch48k_AC3[i++]));
+ }
+ ASSERT_TRUE(scanner->scan(sVoice1ch48k_AC3[i++]));
+ ASSERT_FALSE(scanner->scan(sVoice1ch48k_AC3[i++]));
+}
+
+TEST(audio_utils_spdif, WriteAC3)
+{
+ MySPDIFEncoder encoder(AUDIO_FORMAT_AC3);
+ encoder.write(sVoice1ch48k_AC3, sizeof(sVoice1ch48k_AC3));
+ ASSERT_EQ(48000, encoder.getFramer()->getSampleRate());
+ ASSERT_EQ(kBytesPerOutputFrame, encoder.getBytesPerOutputFrame());
+ ASSERT_EQ(1, encoder.getRateMultiplier());
+
+ // Check to make sure that the pending bytes calculation did not overflow.
+ size_t burstBufferSizeBytes = encoder.getBurstBufferSizeBytes(); // allocated maximum size
+ size_t pendingBytes = encoder.getPayloadBytesPending();
+ ASSERT_GE(burstBufferSizeBytes, pendingBytes);
+
+ // Write some fake compressed audio to force an output data burst.
+ for (int i = 0; i < 7; i++) {
+ auto result = encoder.write(sZeros, sizeof(sZeros));
+ ASSERT_EQ(sizeof(sZeros), result);
+ }
+ // This value is calculated in SPDIFEncoder::sendZeroPad()
+ // size_t burstSize = mFramer->getSampleFramesPerSyncFrame() * sizeof(uint16_t)
+ // * SPDIF_ENCODED_CHANNEL_COUNT;
+ // If it changes then there is probably a regression.
+ const int kExpectedBurstSize = 6144;
+ ASSERT_EQ(kExpectedBurstSize, encoder.mOutputSizeBytes);
+}
+
+TEST(audio_utils_spdif, ValidEAC3)
+{
+ MySPDIFEncoder encoder(AUDIO_FORMAT_E_AC3);
+ auto result = encoder.write(sChannel6ch48k_EAC3, sizeof(sChannel6ch48k_EAC3));
+ ASSERT_EQ(sizeof(sChannel6ch48k_EAC3), result);
+ ASSERT_EQ(4, encoder.getRateMultiplier()); // EAC3_RATE_MULTIPLIER
+ ASSERT_EQ(48000, encoder.getFramer()->getSampleRate());
+ ASSERT_EQ(kBytesPerOutputFrame, encoder.getBytesPerOutputFrame());
+
+ // Check to make sure that the pending bytes calculation did not overflow.
+ size_t bufferSize = encoder.getBurstBufferSizeBytes();
+ size_t pendingBytes = encoder.getPayloadBytesPending();
+ ASSERT_GE(bufferSize, pendingBytes);
+}
+
+TEST(audio_utils_spdif, InvalidLengthEAC3)
+{
+ MySPDIFEncoder encoder(AUDIO_FORMAT_E_AC3);
+ // Mangle a valid header and try to force a numeric overflow.
+ uint8_t mangled[sizeof(sChannel6ch48k_EAC3)] = {0};
+ memcpy(mangled, sChannel6ch48k_EAC3, sizeof(sChannel6ch48k_EAC3));
+
+ // force frmsiz to zero!
+ mangled[2] = mangled[2] & 0xF8;
+ mangled[3] = 0;
+ auto result = encoder.write(mangled, sizeof(mangled));
+ ASSERT_EQ(sizeof(mangled), result);
+
+ // Check to make sure that the pending bytes calculation did not overflow.
+ size_t bufferSize = encoder.getBurstBufferSizeBytes();
+ size_t pendingBytes = encoder.getPayloadBytesPending();
+ ASSERT_GE(bufferSize, pendingBytes);
+
+}