| /* |
| * 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); |
| |
| } |