/****************************************************************************** * * Copyright 2017 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 <gtest/gtest.h> #include "audio_a2dp_hw/include/audio_a2dp_hw.h" namespace { static uint32_t codec_sample_rate2value( btav_a2dp_codec_sample_rate_t codec_sample_rate) { switch (codec_sample_rate) { case BTAV_A2DP_CODEC_SAMPLE_RATE_44100: return 44100; case BTAV_A2DP_CODEC_SAMPLE_RATE_48000: return 48000; case BTAV_A2DP_CODEC_SAMPLE_RATE_88200: return 88200; case BTAV_A2DP_CODEC_SAMPLE_RATE_96000: return 96000; case BTAV_A2DP_CODEC_SAMPLE_RATE_176400: return 176400; case BTAV_A2DP_CODEC_SAMPLE_RATE_192000: return 192000; case BTAV_A2DP_CODEC_SAMPLE_RATE_16000: return 16000; case BTAV_A2DP_CODEC_SAMPLE_RATE_24000: return 24000; case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE: break; } return 0; } static uint32_t codec_bits_per_sample2value( btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample) { switch (codec_bits_per_sample) { case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16: return 16; case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24: return 24; case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32: return 32; case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE: break; } return 0; } static uint32_t codec_channel_mode2value( btav_a2dp_codec_channel_mode_t codec_channel_mode) { switch (codec_channel_mode) { case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO: return 1; case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO: return 2; case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE: break; } return 0; } } // namespace class AudioA2dpHwTest : public ::testing::Test { protected: AudioA2dpHwTest() {} private: }; TEST_F(AudioA2dpHwTest, test_compute_buffer_size) { const btav_a2dp_codec_sample_rate_t codec_sample_rate_array[] = { BTAV_A2DP_CODEC_SAMPLE_RATE_NONE, BTAV_A2DP_CODEC_SAMPLE_RATE_44100, BTAV_A2DP_CODEC_SAMPLE_RATE_48000, BTAV_A2DP_CODEC_SAMPLE_RATE_88200, BTAV_A2DP_CODEC_SAMPLE_RATE_96000, BTAV_A2DP_CODEC_SAMPLE_RATE_176400, BTAV_A2DP_CODEC_SAMPLE_RATE_192000}; const btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample_array[] = { BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE, BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16, BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24, BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32}; const btav_a2dp_codec_channel_mode_t codec_channel_mode_array[] = { BTAV_A2DP_CODEC_CHANNEL_MODE_NONE, BTAV_A2DP_CODEC_CHANNEL_MODE_MONO, BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO}; for (const auto codec_sample_rate : codec_sample_rate_array) { for (const auto codec_bits_per_sample : codec_bits_per_sample_array) { for (const auto codec_channel_mode : codec_channel_mode_array) { size_t buffer_size = audio_a2dp_hw_stream_compute_buffer_size( codec_sample_rate, codec_bits_per_sample, codec_channel_mode); // Check for invalid input if ((codec_sample_rate == BTAV_A2DP_CODEC_SAMPLE_RATE_NONE) || (codec_bits_per_sample == BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE) || (codec_channel_mode == BTAV_A2DP_CODEC_CHANNEL_MODE_NONE)) { EXPECT_EQ(buffer_size, static_cast<size_t>(AUDIO_STREAM_OUTPUT_BUFFER_SZ)); continue; } uint32_t sample_rate = codec_sample_rate2value(codec_sample_rate); EXPECT_TRUE(sample_rate != 0); uint32_t bits_per_sample = codec_bits_per_sample2value(codec_bits_per_sample); EXPECT_TRUE(bits_per_sample != 0); uint32_t number_of_channels = codec_channel_mode2value(codec_channel_mode); EXPECT_TRUE(number_of_channels != 0); const uint64_t time_period_ms = 20; // TODO: Must be a parameter size_t expected_buffer_size = (time_period_ms * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * sample_rate * number_of_channels * (bits_per_sample / 8)) / 1000; // Compute the divisor and adjust the buffer size const size_t divisor = (AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * 16 * number_of_channels * bits_per_sample) / 8; const size_t remainder = expected_buffer_size % divisor; if (remainder != 0) { expected_buffer_size += divisor - remainder; } EXPECT_EQ(buffer_size, expected_buffer_size); } } } }