Merge "Create unit test framework for audio_a2dp_hw am: 431414efe3 am:... (bbee2baf) · Commits · e / os / android_packages_modules_Bluetooth

system/audio_a2dp_hw/Android.bp

+30 −9

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		cc_defaults {
		name: "audio_a2dp_hw_defaults",
		defaults: ["fluoride_defaults"],
		include_dirs: [
		"packages/modules/Bluetooth/system",
		"packages/modules/Bluetooth/system/audio_a2dp_hw/include",
		]
		}

		// Audio A2DP shared library for target
		// ========================================================
		cc_library_shared {
		cc_library {
		name: "audio.a2dp.default",
		defaults: ["fluoride_defaults"],
		defaults: ["audio_a2dp_hw_defaults"],
		relative_install_path: "hw",
		srcs: [
		"audio_a2dp_hw.cc",
		"audio_a2dp_hw_utils.cc",
		],
		include_dirs: [
		"packages/modules/Bluetooth/system",
		"src/audio_a2dp_hw.cc",
		"src/audio_a2dp_hw_utils.cc",
		],
		shared_libs: [
		"liblog",
		@@ -19,8 +25,23 @@ cc_library_shared {

		cc_library_static {
		name: "libaudio-a2dp-hw-utils",
		defaults: ["fluoride_defaults"],
		defaults: ["audio_a2dp_hw_defaults"],
		srcs: [
		"audio_a2dp_hw_utils.cc",
		"src/audio_a2dp_hw_utils.cc",
		],
		}

		// Audio A2DP library unit tests for target and host
		// ========================================================
		cc_test {
		name: "net_test_audio_a2dp_hw",
		test_suites: ["device-tests"],
		defaults: ["audio_a2dp_hw_defaults"],
		srcs: [
		"test/audio_a2dp_hw_test.cc",
		],
		shared_libs: [
		"liblog",
		],
		static_libs: ["audio.a2dp.default"],
		}

system/audio_a2dp_hw/audio_a2dp_hw.h→system/audio_a2dp_hw/include/audio_a2dp_hw.h

+33 −3

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		@@ -29,6 +29,8 @@

		#include <stdint.h>

		#include <hardware/bt_av.h>

		/*****************************************************************************
		* Constants & Macros
		*****************************************************************************/
		@@ -41,9 +43,6 @@
		// If one assumes the write buffer is always full during normal BT playback,
		// then increasing this value increases our playback latency.
		//
		// FIXME: AUDIO_STREAM_OUTPUT_BUFFER_SZ should be controlled by the actual audio
		// sample rate rather than being constant.
		//
		// FIXME: The BT HAL should consume data at a constant rate.
		// AudioFlinger assumes that the HAL draws data at a constant rate, which is
		// true for most audio devices; however, the BT engine reads data at a variable
		@@ -115,6 +114,37 @@ typedef uint8_t tA2DP_BITS_PER_SAMPLE;
		* Functions
		*****************************************************************************/

		// Computes the Audio A2DP HAL output buffer size.
		// \|codec_sample_rate\| is the sample rate of the output stream.
		// \|codec_bits_per_sample\| is the number of bits per sample of the output
		// stream.
		// \|codec_channel_mode\| is the channel mode of the output stream.
		//
		// The buffer size is computed by using the following formula:
		//
		// AUDIO_STREAM_OUTPUT_BUFFER_SIZE =
		// (TIME_PERIOD_MS * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *
		// SAMPLE_RATE_HZ * NUMBER_OF_CHANNELS * (BITS_PER_SAMPLE / 8)) / 1000
		//
		// AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is
		// divided for AudioFlinger data delivery. The AudioFlinger mixer delivers
		// data in chunks of
		// (AUDIO_STREAM_OUTPUT_BUFFER_SIZE / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS) .
		// If the number of periods is 2, the socket buffer represents "double
		// buffering" of the AudioFlinger mixer buffer.
		//
		// Furthermore, the AudioFlinger expects the buffer size to be a multiple
		// of 16 frames.
		//
		// NOTE: Currently, the computation uses the conservative 20ms time period.
		//
		// Returns the computed buffer size. If any of the input parameters is
		// invalid, the return value is the default \|AUDIO_STREAM_OUTPUT_BUFFER_SZ\|.
		extern size_t audio_a2dp_hw_stream_compute_buffer_size(
		btav_a2dp_codec_sample_rate_t codec_sample_rate,
		btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,
		btav_a2dp_codec_channel_mode_t codec_channel_mode);

		// Returns a string representation of \|event\|.
		extern const char* audio_a2dp_hw_dump_ctrl_event(tA2DP_CTRL_CMD event);

system/audio_a2dp_hw/audio_a2dp_hw.cc→system/audio_a2dp_hw/src/audio_a2dp_hw.cc

+3 −13

Original line number	Diff line number	Diff line
		@@ -40,7 +40,6 @@
		#include <mutex>

		#include <hardware/audio.h>
		#include <hardware/bt_av.h>
		#include <hardware/hardware.h>
		#include <system/audio.h>

		@@ -150,10 +149,6 @@ struct a2dp_stream_in {
		*****************************************************************************/

		static size_t out_get_buffer_size(const struct audio_stream* stream);
		static size_t audio_stream_compute_buffer_size(
		btav_a2dp_codec_sample_rate_t codec_sample_rate,
		btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,
		btav_a2dp_codec_channel_mode_t codec_channel_mode);

		/*****************************************************************************
		* Externs
		@@ -611,7 +606,7 @@ static int a2dp_read_output_audio_config(
		common->cfg.rate = stream_config.rate;
		common->cfg.channel_mask = stream_config.channel_mask;
		common->cfg.format = stream_config.format;
		common->buffer_sz = audio_stream_compute_buffer_size(
		common->buffer_sz = audio_a2dp_hw_stream_compute_buffer_size(
		codec_config->sample_rate, codec_config->bits_per_sample,
		codec_config->channel_mode);
		}
		@@ -922,24 +917,19 @@ static size_t out_get_buffer_size(const struct audio_stream* stream) {
		// period_size is the AudioFlinger mixer buffer size.
		const size_t period_size =
		out->common.buffer_sz / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS;
		const size_t mixer_unit_size = 16 /* frames / 4 /* framesize */;

		DEBUG("socket buffer size: %zu period size: %zu", out->common.buffer_sz,
		period_size);
		if (period_size % mixer_unit_size != 0) {
		ERROR("period size %zu not a multiple of %zu", period_size,
		mixer_unit_size);
		}

		return period_size;
		}

		static size_t audio_stream_compute_buffer_size(
		size_t audio_a2dp_hw_stream_compute_buffer_size(
		btav_a2dp_codec_sample_rate_t codec_sample_rate,
		btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,
		btav_a2dp_codec_channel_mode_t codec_channel_mode) {
		size_t buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ; // Default value
		const uint32_t time_period_ms = 20; // Conservative 20ms
		const uint64_t time_period_ms = 20; // Conservative 20ms
		uint32_t sample_rate;
		uint32_t bits_per_sample;
		uint32_t number_of_channels;

system/audio_a2dp_hw/audio_a2dp_hw_utils.cc→system/audio_a2dp_hw/src/audio_a2dp_hw_utils.cc

+0 −0

File moved.

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system/audio_a2dp_hw/test/audio_a2dp_hw_test.cc

0 → 100644

+142 −0

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		/******************************************************************************
		*
		* Copyright (C) 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_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);
		}
		}
		}
		}