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/******************************************************************************
*
* 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_hearing_aid_hw/include/audio_hearing_aid_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_ha_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);
}
}
}
}