Android 音频设备信息加载
Android 系统的守护进程 audioserver 中运行着多个与音频相关的系统服务，主要包括 AudioFlinger 和 AudioPolicyService，当需要支持 AAudio 的 MMap 模式时，会运行 AAudioService。需要启动 MediaLogService 服务时，audioserver 程序会 fork 一个进程运行其它那些系统服务，在父进程中运行 MediaLogService，并将进程名修改为 media.log。audioserver 程序的主函数代码如下 (位于 frameworks/av/media/audioserver/main_audioserver.cpp) 所示：
int main(int argc __unused, char **argv)
{
// TODO: update with refined parameters
limitProcessMemory(
"audio.maxmem", /* "ro.audio.maxmem", property that defines limit */
(size_t)512 * (1 << 20), /* SIZE_MAX, upper limit in bytes */
20 /* upper limit as percentage of physical RAM */);
signal(SIGPIPE, SIG_IGN);
#if 1
// FIXME See bug 165702394 and bug 168511485
const bool doLog = false;
#else
bool doLog = (bool) property_get_bool("ro.test_harness", 0);
#endif
pid_t childPid;
// FIXME The advantage of making the process containing media.log service the parent process of
// the process that contains the other audio services, is that it allows us to collect more
// detailed information such as signal numbers, stop and continue, resource usage, etc.
// But it is also more complex. Consider replacing this by independent processes, and using
// binder on death notification instead.
if (doLog && (childPid = fork()) != 0) {
// media.log service
//prctl(PR_SET_NAME, (unsigned long) "media.log", 0, 0, 0);
// unfortunately ps ignores PR_SET_NAME for the main thread, so use this ugly hack
strcpy(argv[0], "media.log");
sp proc(ProcessState::self());
MediaLogService::instantiate();
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
for (;;) {
siginfo_t info;
int ret = waitid(P_PID, childPid, &info, WEXITED | WSTOPPED | WCONTINUED);
if (ret == EINTR) {
continue;
}
if (ret < 0) {
break;
}
char buffer[32];
const char *code;
switch (info.si_code) {
case CLD_EXITED:
code = "CLD_EXITED";
break;
case CLD_KILLED:
code = "CLD_KILLED";
break;
case CLD_DUMPED:
code = "CLD_DUMPED";
break;
case CLD_STOPPED:
code = "CLD_STOPPED";
break;
case CLD_TRAPPED:
code = "CLD_TRAPPED";
break;
case CLD_CONTINUED:
code = "CLD_CONTINUED";
break;
default:
snprintf(buffer, sizeof(buffer), "unknown (%d)", info.si_code);
code = buffer;
break;
}
struct rusage usage;
getrusage(RUSAGE_CHILDREN, &usage);
ALOG(LOG_ERROR, "media.log", "pid %d status %d code %s user %ld.%03lds sys %ld.%03lds",
info.si_pid, info.si_status, code,
usage.ru_utime.tv_sec, usage.ru_utime.tv_usec / 1000,
usage.ru_stime.tv_sec, usage.ru_stime.tv_usec / 1000);
sp sm = defaultServiceManager();
sp binder = sm->getService(String16("media.log"));
if (binder != 0) {
Vector args;
binder->dump(-1, args);
}
switch (info.si_code) {
case CLD_EXITED:
case CLD_KILLED:
case CLD_DUMPED: {
ALOG(LOG_INFO, "media.log", "exiting");
_exit(0);
// not reached
}
default:
break;
}
}
} else {
// all other services
if (doLog) {
prctl(PR_SET_PDEATHSIG, SIGKILL); // if parent media.log dies before me, kill me also
setpgid(0, 0); // but if I die first, don't kill my parent
}
android::hardware::configureRpcThreadpool(4, false /*callerWillJoin*/);
sp proc(ProcessState::self());
sp sm = defaultServiceManager();
ALOGI("ServiceManager: %p audio server", sm.get());
AudioFlinger::instantiate();
AudioPolicyService::instantiate();
// AAudioService should only be used in OC-MR1 and later.
// And only enable the AAudioService if the system MMAP policy explicitly allows it.
// This prevents a client from misusing AAudioService when it is not supported.
aaudio_policy_t mmapPolicy = property_get_int32(AAUDIO_PROP_MMAP_POLICY,
AAUDIO_POLICY_NEVER);
if (mmapPolicy == AAUDIO_POLICY_AUTO || mmapPolicy == AAUDIO_POLICY_ALWAYS) {
ALOGI("Start aaudio service");
AAudioService::instantiate();
} else {
ALOGI("Donnot start aaudio service");
}
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
}
}
AudioPolicyService 加载音频设备信息
AudioPolicyService 在创建及启动过程中会从 XML 文件中加载音频设备的信息，并请求 AudioFlinger 加载硬件模块，打开设备等。AudioPolicyService 的实现大概如下图这样：
更具体地说，AudioPolicyService 在初始化阶段，会创建并初始化 AudioPolicyManager 对象，默认的 AudioPolicyManager 对象在创建时，会加载音频策略配置信息，其中包含关于音频硬件模块和设备的配置信息，相关代码如下 (位于 frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp) 所示：
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface,
bool /*forTesting*/)
:
mUidCached(AID_AUDIOSERVER), // no need to call getuid(), there's only one of us running.
mpClientInterface(clientInterface),
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mA2dpSuspended(false),
mConfig(mHwModulesAll, mOutputDevicesAll, mInputDevicesAll, mDefaultOutputDevice),
mAudioPortGeneration(1),
mBeaconMuteRefCount(0),
mBeaconPlayingRefCount(0),
mBeaconMuted(false),
mTtsOutputAvailable(false),
mMasterMono(false),
mMusicEffectOutput(AUDIO_IO_HANDLE_NONE)
{
}
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface, false /*forTesting*/)
{
loadConfig();
}
void AudioPolicyManager::loadConfig() {
if (deserializeAudioPolicyXmlConfig(getConfig()) != NO_ERROR) {
ALOGE("could not load audio policy configuration file, setting defaults");
getConfig().setDefault();
}
//TODO: b/193496180 use spatializer flag at audio HAL when available
getConfig().convertSpatializerFlag();
}
在 AudioPolicyManager::loadConfig() 中：
- 反序列化音频策略 XML 配置文件，解析的结果保存在
AudioPolicyConfig对象中； - 如果解析音频策略 XML 配置文件失败，
AudioPolicyConfig的各项配置被设置为默认值； - 转换
AudioPolicyConfig中各个硬件模块的各个输出设备的空间音频标记。
AudioPolicyConfig::setDefault() 函数设置默认的音频策略配置，其定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h) 如下：
void setDefault(void)
{
mSource = "AudioPolicyConfig::setDefault";
mEngineLibraryNameSuffix = kDefaultEngineLibraryNameSuffix;
mDefaultOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
sp defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
sp micProfile = new AudioProfile(
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000);
defaultInputDevice->addAudioProfile(micProfile);
mOutputDevices.add(mDefaultOutputDevice);
mInputDevices.add(defaultInputDevice);
sp module = new HwModule(AUDIO_HARDWARE_MODULE_ID_PRIMARY, 2 /*halVersionMajor*/);
mHwModules.add(module);
sp outProfile = new OutputProfile("primary");
outProfile->addAudioProfile(
new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 44100));
outProfile->addSupportedDevice(mDefaultOutputDevice);
outProfile->setFlags(AUDIO_OUTPUT_FLAG_PRIMARY);
module->addOutputProfile(outProfile);
sp inProfile = new InputProfile("primary");
inProfile->addAudioProfile(micProfile);
inProfile->addSupportedDevice(defaultInputDevice);
module->addInputProfile(inProfile);
setDefaultSurroundFormats();
}
// Surround formats, with an optional list of subformats that are equivalent from users' POV.
using SurroundFormats = std::unordered_map>;
const SurroundFormats &getSurroundFormats() const
{
return mSurroundFormats;
}
void setSurroundFormats(const SurroundFormats &surroundFormats)
{
mSurroundFormats = surroundFormats;
}
void setDefaultSurroundFormats()
{
mSurroundFormats = {
{AUDIO_FORMAT_AC3, {}},
{AUDIO_FORMAT_E_AC3, {}},
{AUDIO_FORMAT_DTS, {}},
{AUDIO_FORMAT_DTS_HD, {}},
{AUDIO_FORMAT_AAC_LC, {
AUDIO_FORMAT_AAC_HE_V1, AUDIO_FORMAT_AAC_HE_V2, AUDIO_FORMAT_AAC_ELD,
AUDIO_FORMAT_AAC_XHE}},
{AUDIO_FORMAT_DOLBY_TRUEHD, {}},
{AUDIO_FORMAT_E_AC3_JOC, {}},
{AUDIO_FORMAT_AC4, {}}};
}
音频策略配置的结构如下图：
图中各个类之间的层次继承结构如下图：
转换 AudioPolicyConfig 中各个硬件模块的各个输出设备的空间音频标记的方法 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h) 如下：
void convertSpatializerFlag()
{
for (const auto& hwModule : mHwModules) {
for (const auto& curProfile : hwModule->getOutputProfiles()) {
if (curProfile->getFlags()
== (AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
curProfile->setFlags(AUDIO_OUTPUT_FLAG_SPATIALIZER);
}
}
}
}
deserializeAudioPolicyXmlConfig() 函数解析音频策略 XML 配置文件，填充 AudioPolicyConfig 对象，这个函数定义 (位于 frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp) 如下：
static status_t deserializeAudioPolicyXmlConfig(AudioPolicyConfig &config) {
if (std::string audioPolicyXmlConfigFile = audio_get_audio_policy_config_file();
!audioPolicyXmlConfigFile.empty()) {
status_t ret = deserializeAudioPolicyFile(audioPolicyXmlConfigFile.c_str(), &config);
if (ret == NO_ERROR) {
config.setSource(audioPolicyXmlConfigFile);
}
return ret;
}
return BAD_VALUE;
}
deserializeAudioPolicyXmlConfig() 函数首先获得音频策略 XML 配置文件的路径，然后解析文件，最后为 AudioPolicyConfig 对象设置内容来源。上面调用的 deserializeAudioPolicyFile() 函数定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp) 如下：
status_t PolicySerializer::deserialize(const char *configFile, AudioPolicyConfig *config,
bool ignoreVendorExtensions)
{
mIgnoreVendorExtensions = ignoreVendorExtensions;
auto doc = make_xmlUnique(xmlParseFile(configFile));
if (doc == nullptr) {
ALOGE("%s: Could not parse %s document.", __func__, configFile);
return BAD_VALUE;
}
xmlNodePtr root = xmlDocGetRootElement(doc.get());
if (root == NULL) {
ALOGE("%s: Could not parse %s document: empty.", __func__, configFile);
return BAD_VALUE;
}
if (xmlXIncludeProcess(doc.get()) < 0) {
ALOGE("%s: libxml failed to resolve XIncludes on %s document.", __func__, configFile);
}
if (xmlStrcmp(root->name, reinterpret_cast(rootName))) {
ALOGE("%s: No %s root element found in xml data %s.", __func__, rootName,
reinterpret_cast(root->name));
return BAD_VALUE;
}
std::string version = getXmlAttribute(root, versionAttribute);
if (version.empty()) {
ALOGE("%s: No version found in root node %s", __func__, rootName);
return BAD_VALUE;
}
if (version == "7.0") {
mChannelMasksSeparator = mSamplingRatesSeparator = mFlagsSeparator = " ";
} else if (version != "1.0") {
ALOGE("%s: Version does not match; expected \"1.0\" or \"7.0\" got \"%s\"",
__func__, version.c_str());
return BAD_VALUE;
}
// Let's deserialize children
// Modules
ModuleTraits::Collection modules;
status_t status = deserializeCollection(root, &modules, config);
if (status != NO_ERROR) {
return status;
}
config->setHwModules(modules);
// Global Configuration
deserialize(root, config);
// Surround configuration
deserialize(root, config);
return android::OK;
}
} // namespace
status_t deserializeAudioPolicyFile(const char *fileName, AudioPolicyConfig *config)
{
PolicySerializer serializer;
status_t status = serializer.deserialize(fileName, config);
if (status != OK) config->clear();
return status;
}
这里解析音频策略 XML 配置文件时，先解析音频硬件模块信息，再解析全局配置信息，最后解析环绕声信息。
audio_get_audio_policy_config_file() 函数用于获得音频策略 XML 配置文件的路径，其定义 (位于 system/media/audio/include/system/audio_config.h) 如下：
namespace android {
// Returns a vector of paths where audio configuration files
// must be searched, in the provided order.
static inline std::vector audio_get_configuration_paths() {
static const std::vector paths = []() {
char value[PROPERTY_VALUE_MAX] = {};
if (property_get("ro.boot.product.vendor.sku", value, "") <= 0) {
return std::vector({"/odm/etc", "/vendor/etc", "/system/etc"});
} else {
return std::vector({
"/odm/etc", std::string("/vendor/etc/audio/sku_") + value,
"/vendor/etc", "/system/etc"});
}
}();
return paths;
}
static inline std::string audio_find_readable_configuration_file(const char* fileName) {
for (const auto& path : audio_get_configuration_paths()) {
std::string tryPath = path + "/" + fileName;
if (access(tryPath.c_str(), R_OK) == 0) {
return tryPath;
}
}
return {};
}
static inline std::string audio_get_audio_policy_config_file() {
static constexpr const char *apmXmlConfigFileName = "audio_policy_configuration.xml";
static constexpr const char *apmA2dpOffloadDisabledXmlConfigFileName =
"audio_policy_configuration_a2dp_offload_disabled.xml";
static constexpr const char *apmBluetoothLegacyHalXmlConfigFileName =
"audio_policy_configuration_bluetooth_legacy_hal.xml";
std::string audioPolicyXmlConfigFile;
// First try alternative files if needed
if (property_get_bool("ro.bluetooth.a2dp_offload.supported", false)) {
if (property_get_bool("persist.bluetooth.bluetooth_audio_hal.disabled", false) &&
property_get_bool("persist.bluetooth.a2dp_offload.disabled", false)) {
// Both [email protected] and [email protected] (Offload) are disabled, and uses
// the legacy hardware module for A2DP and hearing aid.
audioPolicyXmlConfigFile = audio_find_readable_configuration_file(
apmBluetoothLegacyHalXmlConfigFileName);
} else if (property_get_bool("persist.bluetooth.a2dp_offload.disabled", false)) {
// A2DP offload supported but disabled: try to use special XML file
audioPolicyXmlConfigFile = audio_find_readable_configuration_file(
apmA2dpOffloadDisabledXmlConfigFileName);
}
} else if (property_get_bool("persist.bluetooth.bluetooth_audio_hal.disabled", false)) {
audioPolicyXmlConfigFile = audio_find_readable_configuration_file(
apmBluetoothLegacyHalXmlConfigFileName);
}
return audioPolicyXmlConfigFile.empty() ?
audio_find_readable_configuration_file(apmXmlConfigFileName) : audioPolicyXmlConfigFile;
}
} // namespace android
audio_get_audio_policy_config_file() 函数根据系统属性的配置，按一定的文件优先级，依次按优先级顺序在一组目录下查找音频策略 XML 配置文件。
对于 AAOS 版的模拟器，主要的音频策略 XML 配置文件为 device/generic/car/emulator/audio/audio_policy_configuration.xml，这个文件的主要内容为：
- bus0_media_out
- bus1_navigation_out
- bus2_voice_command_out
- bus3_call_ring_out
- bus4_call_out
- bus5_alarm_out
- bus6_notification_out
- bus7_system_sound_out
- bus100_audio_zone_1
- bus200_audio_zone_2
- Built-In Mic
- Built-In Back Mic
- Echo-Reference Mic
- FM Tuner
- Tone Generator 0
- Tone Generator 1
bus0_media_out
这个文件还包含了几个其它的配置文件，在 device/generic/car/emulator/audio/car_emulator_audio.mk 文件中可以看到这些文件在源码库及设备的文件系统中的位置：
PRODUCT_COPY_FILES += \
frameworks/native/data/etc/android.hardware.broadcastradio.xml:system/etc/permissions/android.hardware.broadcastradio.xml \
frameworks/av/services/audiopolicy/config/a2dp_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/usb_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/usb_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/r_submix_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/r_submix_audio_policy_configuration.xml \
frameworks/av/services/audiopolicy/config/audio_policy_volumes.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_volumes.xml \
frameworks/av/services/audiopolicy/config/default_volume_tables.xml:$(TARGET_COPY_OUT_VENDOR)/etc/default_volume_tables.xml \
device/generic/car/emulator/audio/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
device/generic/car/emulator/audio/car_audio_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/car_audio_configuration.xml \
加载的这些音频策略 XML 配置文件中，共描述了 4 个音频硬件模块。我们再来看音频策略 XML 配置文件模块信息解析这部分 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp)：
template
status_t PolicySerializer::deserializeCollection(const xmlNode *cur,
typename Trait::Collection *collection,
typename Trait::PtrSerializingCtx serializingContext)
{
for (cur = cur->xmlChildrenNode; cur != NULL; cur = cur->next) {
const xmlNode *child = NULL;
if (!xmlStrcmp(cur->name, reinterpret_cast(Trait::collectionTag))) {
child = cur->xmlChildrenNode;
} else if (!xmlStrcmp(cur->name, reinterpret_cast(Trait::tag))) {
child = cur;
}
for (; child != NULL; child = child->next) {
if (!xmlStrcmp(child->name, reinterpret_cast(Trait::tag))) {
auto maybeElement = deserialize(child, serializingContext);
if (maybeElement.index() == 1) {
status_t status = Trait::addElementToCollection(
std::get<1>(maybeElement), collection);
if (status != NO_ERROR) {
ALOGE("%s: could not add element to %s collection", __func__,
Trait::collectionTag);
return status;
}
} else if (mIgnoreVendorExtensions && std::get(maybeElement) == NO_INIT) {
// Skip a vendor extension element.
} else {
return BAD_VALUE;
}
}
}
if (!xmlStrcmp(cur->name, reinterpret_cast(Trait::tag))) {
return NO_ERROR;
}
}
return NO_ERROR;
}
. . . . . .
template<>
std::variant PolicySerializer::deserialize(
const xmlNode *cur, ModuleTraits::PtrSerializingCtx ctx)
{
using Attributes = ModuleTraits::Attributes;
auto& tag = ModuleTraits::tag;
auto& childAttachedDevicesTag = ModuleTraits::childAttachedDevicesTag;
auto& childAttachedDeviceTag = ModuleTraits::childAttachedDeviceTag;
auto& childDefaultOutputDeviceTag = ModuleTraits::childDefaultOutputDeviceTag;
std::string name = getXmlAttribute(cur, Attributes::name);
if (name.empty()) {
ALOGE("%s: No %s found", __func__, Attributes::name);
return BAD_VALUE;
}
uint32_t versionMajor = 0, versionMinor = 0;
std::string versionLiteral = getXmlAttribute(cur, Attributes::version);
if (!versionLiteral.empty()) {
sscanf(versionLiteral.c_str(), "%u.%u", &versionMajor, &versionMinor);
ALOGV("%s: mHalVersion = major %u minor %u", __func__,
versionMajor, versionMajor);
}
ALOGV("%s: %s %s=%s", __func__, ModuleTraits::tag, Attributes::name, name.c_str());
ModuleTraits::Element module = new HwModule(name.c_str(), versionMajor, versionMinor);
// Deserialize children: Audio Mix Port, Audio Device Ports (Source/Sink), Audio Routes
MixPortTraits::Collection mixPorts;
status_t status = deserializeCollection(cur, &mixPorts, NULL);
if (status != NO_ERROR) {
return status;
}
module->setProfiles(mixPorts);
DevicePortTraits::Collection devicePorts;
status = deserializeCollection(cur, &devicePorts, NULL);
if (status != NO_ERROR) {
return status;
}
module->setDeclaredDevices(devicePorts);
RouteTraits::Collection routes;
status = deserializeCollection(cur, &routes, module.get());
if (status != NO_ERROR) {
return status;
}
module->setRoutes(routes);
for (const xmlNode *children = cur->xmlChildrenNode; children != NULL;
children = children->next) {
if (!xmlStrcmp(children->name, reinterpret_cast(childAttachedDevicesTag))) {
ALOGV("%s: %s %s found", __func__, tag, childAttachedDevicesTag);
for (const xmlNode *child = children->xmlChildrenNode; child != NULL;
child = child->next) {
if (!xmlStrcmp(child->name,
reinterpret_cast(childAttachedDeviceTag))) {
auto attachedDevice = make_xmlUnique(xmlNodeListGetString(
child->doc, child->xmlChildrenNode, 1));
if (attachedDevice != nullptr) {
ALOGV("%s: %s %s=%s", __func__, tag, childAttachedDeviceTag,
reinterpret_cast(attachedDevice.get()));
sp device = module->getDeclaredDevices().
getDeviceFromTagName(std::string(reinterpret_cast(
attachedDevice.get())));
if (device == nullptr && mIgnoreVendorExtensions) {
ALOGW("Skipped attached device \"%s\" because it likely uses a vendor"
"extension type",
reinterpret_cast(attachedDevice.get()));
continue;
}
ctx->addDevice(device);
}
}
}
}
if (!xmlStrcmp(children->name,
reinterpret_cast(childDefaultOutputDeviceTag))) {
auto defaultOutputDevice = make_xmlUnique(xmlNodeListGetString(
children->doc, children->xmlChildrenNode, 1));
if (defaultOutputDevice != nullptr) {
ALOGV("%s: %s %s=%s", __func__, tag, childDefaultOutputDeviceTag,
reinterpret_cast(defaultOutputDevice.get()));
sp device = module->getDeclaredDevices().getDeviceFromTagName(
std::string(reinterpret_cast(defaultOutputDevice.get())));
if (device != 0 && ctx->getDefaultOutputDevice() == 0) {
ctx->setDefaultOutputDevice(device);
ALOGV("%s: default is %08x",
__func__, ctx->getDefaultOutputDevice()->type());
}
}
}
}
return module;
}
音频策略的音频硬件模块相关的这些 Traits 的定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp) 如下：
// A profile section contains a name, one audio format and the list of supported sampling rates
// and channel masks for this format
struct AudioProfileTraits : public AndroidCollectionTraits
{
static constexpr const char *tag = "profile";
static constexpr const char *collectionTag = "profiles";
struct Attributes
{
static constexpr const char *samplingRates = "samplingRates";
static constexpr const char *format = "format";
static constexpr const char *channelMasks = "channelMasks";
};
};
struct MixPortTraits : public AndroidCollectionTraits
{
static constexpr const char *tag = "mixPort";
static constexpr const char *collectionTag = "mixPorts";
struct Attributes
{
static constexpr const char *name = "name";
static constexpr const char *role = "role";
static constexpr const char *roleSource = "source"; /**< . */
static constexpr const char *flags = "flags";
static constexpr const char *maxOpenCount = "maxOpenCount";
static constexpr const char *maxActiveCount = "maxActiveCount";
};
// Children: GainTraits
};
struct DevicePortTraits : public AndroidCollectionTraits
{
static constexpr const char *tag = "devicePort";
static constexpr const char *collectionTag = "devicePorts";
struct Attributes
{
/** : any string without space. */
static constexpr const char *tagName = "tagName";
static constexpr const char *type = "type"; /**< . */
static constexpr const char *role = "role"; /**< . */
static constexpr const char *roleSource = "source"; /**< . */
/** optional: device address, char string less than 64. */
static constexpr const char *address = "address";
/** optional: the list of encoded audio formats that are known to be supported. */
static constexpr const char *encodedFormats = "encodedFormats";
};
// Children: GainTraits (optional)
};
struct RouteTraits : public AndroidCollectionTraits
{
static constexpr const char *tag = "route";
static constexpr const char *collectionTag = "routes";
struct Attributes
{
static constexpr const char *type = "type"; /**< : mix or mux. */
static constexpr const char *typeMix = "mix"; /**< type attribute mix value. */
static constexpr const char *sink = "sink"; /**< . */
/** sources: all source that can be involved in this route. */
static constexpr const char *sources = "sources";
};
typedef HwModule *PtrSerializingCtx;
};
struct ModuleTraits : public AndroidCollectionTraits
{
static constexpr const char *tag = "module";
static constexpr const char *collectionTag = "modules";
static constexpr const char *childAttachedDevicesTag = "attachedDevices";
static constexpr const char *childAttachedDeviceTag = "item";
static constexpr const char *childDefaultOutputDeviceTag = "defaultOutputDevice";
struct Attributes
{
static constexpr const char *name = "name";
static constexpr const char *version = "halVersion";
};
typedef AudioPolicyConfig *PtrSerializingCtx;
// Children: mixPortTraits, devicePortTraits, and routeTraits
// Need to call deserialize on each child
};
这样不难理解音频策略 XML 配置文件中的元素和 AudioPolicyConfig 包含的结构体之间具有这样的对应关系：
modules <--------------------> HwModuleCollection
module <--------------------> HwModule
mixPorts <--------------------> IOProfileCollection
mixPort <--------------------> IOProfile
gains <--------------------> AudioGains
gain <--------------------> AudioGain
devicePorts <--------------------> DeviceVector
devicePort <--------------------> DeviceDescriptor
profiles <--------------------> AudioProfileVector
profile <--------------------> AudioProfile
routes <--------------------> AudioRouteVector
route <--------------------> AudioRoute
音频策略 XML 配置文件中 attachedDevices 部分所列的设备，不直接解析为 AudioPolicyConfig 中的元素，它们主要用于对直接解析出来的，即声明的 DeviceDescriptor 做过滤。向 HwModule 添加 mixPorts 的方法如下：
status_t HwModule::addOutputProfile(const sp &profile)
{
profile->attach(this);
mOutputProfiles.add(profile);
mPorts.add(profile);
return NO_ERROR;
}
status_t HwModule::addInputProfile(const sp &profile)
{
profile->attach(this);
mInputProfiles.add(profile);
mPorts.add(profile);
return NO_ERROR;
}
status_t HwModule::addProfile(const sp &profile)
{
switch (profile->getRole()) {
case AUDIO_PORT_ROLE_SOURCE:
return addOutputProfile(profile);
case AUDIO_PORT_ROLE_SINK:
return addInputProfile(profile);
case AUDIO_PORT_ROLE_NONE:
return BAD_VALUE;
}
return BAD_VALUE;
}
void HwModule::setProfiles(const IOProfileCollection &profiles)
{
for (size_t i = 0; i < profiles.size(); i++) {
addProfile(profiles[i]);
}
}
mixPort 和 devicePort 都是既包含 profiles，又包含 gains，即 IOProfile 和 DeviceDescriptor 都是既包含 AudioProfileVector，又包含 AudioGains。
音频策略 XML 配置文件中的 route 元素的解析过程如下：
template<>
std::variant PolicySerializer::deserialize(
const xmlNode *cur, RouteTraits::PtrSerializingCtx ctx)
{
using Attributes = RouteTraits::Attributes;
std::string type = getXmlAttribute(cur, Attributes::type);
if (type.empty()) {
ALOGE("%s: No %s found", __func__, Attributes::type);
return BAD_VALUE;
}
audio_route_type_t routeType = (type == Attributes::typeMix) ?
AUDIO_ROUTE_MIX : AUDIO_ROUTE_MUX;
ALOGV("%s: %s %s=%s", __func__, RouteTraits::tag, Attributes::type, type.c_str());
RouteTraits::Element route = new AudioRoute(routeType);
std::string sinkAttr = getXmlAttribute(cur, Attributes::sink);
if (sinkAttr.empty()) {
ALOGE("%s: No %s found", __func__, Attributes::sink);
return BAD_VALUE;
}
// Convert Sink name to port pointer
sp sink = ctx->findPortByTagName(sinkAttr);
if (sink == NULL && !mIgnoreVendorExtensions) {
ALOGE("%s: no sink found with name=%s", __func__, sinkAttr.c_str());
return BAD_VALUE;
} else if (sink == NULL) {
ALOGW("Skipping route to sink \"%s\" as it likely has vendor extension type",
sinkAttr.c_str());
return NO_INIT;
}
route->setSink(sink);
std::string sourcesAttr = getXmlAttribute(cur, Attributes::sources);
if (sourcesAttr.empty()) {
ALOGE("%s: No %s found", __func__, Attributes::sources);
return BAD_VALUE;
}
// Tokenize and Convert Sources name to port pointer
PolicyAudioPortVector sources;
UniqueCPtr sourcesLiteral{strndup(
sourcesAttr.c_str(), strlen(sourcesAttr.c_str()))};
char *devTag = strtok(sourcesLiteral.get(), ",");
while (devTag != NULL) {
if (strlen(devTag) != 0) {
sp source = ctx->findPortByTagName(devTag);
if (source == NULL && !mIgnoreVendorExtensions) {
ALOGE("%s: no source found with name=%s", __func__, devTag);
return BAD_VALUE;
} else if (source == NULL) {
ALOGW("Skipping route source \"%s\" as it likely has vendor extension type",
devTag);
} else {
sources.add(source);
}
}
devTag = strtok(NULL, ",");
}
sink->addRoute(route);
for (size_t i = 0; i < sources.size(); i++) {
sp source = sources.itemAt(i);
source->addRoute(route);
}
route->setSources(sources);
return route;
}
AudioRoute 将 source 和 sink 连接起来，对于音频播放来说，devicePort 是 sink，mixPort 是 source，对于音频数据采集来说，devicePort 是 source，mixPort 是 sink。解析 route 元素时，会根据 source 和 sink 的名称，找出前面解析出来的匹配的 IOProfile 和 DeviceDescriptor 对象，source 可能会有多个。此时 AudioRoute 分别建立与 IOProfile 和 DeviceDescriptor 的关联。
routes 和所有 route 元素解析完成，设置给 HwModule 时，会建立 IOProfile 和 DeviceDescriptor 之间的关联，相关的 HwModule::setRoutes(const AudioRouteVector &routes) 函数的定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp) 如下：
void HwModule::setRoutes(const AudioRouteVector &routes)
{
mRoutes = routes;
// Now updating the streams (aka IOProfile until now) supported devices
refreshSupportedDevices();
}
void HwModule::refreshSupportedDevices()
{
// Now updating the streams (aka IOProfile until now) supported devices
for (const auto& stream : mInputProfiles) {
DeviceVector sourceDevices;
for (const auto& route : stream->getRoutes()) {
sp sink = route->getSink();
if (sink == 0 || stream != sink) {
ALOGE("%s: Invalid route attached to input stream", __FUNCTION__);
continue;
}
DeviceVector sourceDevicesForRoute = getRouteSourceDevices(route);
if (sourceDevicesForRoute.isEmpty()) {
ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
sourceDevices.add(sourceDevicesForRoute);
}
if (sourceDevices.isEmpty()) {
ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
stream->setSupportedDevices(sourceDevices);
}
for (const auto& stream : mOutputProfiles) {
DeviceVector sinkDevices;
for (const auto& route : stream->getRoutes()) {
sp source = findByTagName(route->getSources(), stream->getTagName());
if (source == 0 || stream != source) {
ALOGE("%s: Invalid route attached to output stream", __FUNCTION__);
continue;
}
sp sinkDevice = getRouteSinkDevice(route);
if (sinkDevice == 0) {
ALOGE("%s: invalid sink device for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
sinkDevices.add(sinkDevice);
}
stream->setSupportedDevices(sinkDevices);
}
}
对于采集输入 IOProfile (mixPort) 来说，它是与它关联的 AudioRoute 的 sink，设备是该 AudioRoute 的 source，这里找到与它关联的所有 AudioRoute，进而找到与该 AudioRoute 关联的所有 source，即为该 IOProfile (mixPort) 的所有支持的设备。
对于播放输出 IOProfile (mixPort) 来说，它是与它关联的 AudioRoute 的 source，设备是该 AudioRoute 的 sink，这里找到与它关联的所有 AudioRoute，进而找到与该 AudioRoute 关联的那个 sink，即为该 IOProfile (mixPort) 的支持的设备。
整个音频策略配置 XML 文件解析过程，为硬件模块 HwModule 中的 IOProfile、AudioRoute 和 DeviceDescriptor 这些对象建立了如下这样的关联：
默认的 AudioPolicyManager 对象在初始化时，会根据加载的音频策略配置信息，执行更多的初始化动作，AudioPolicyManager::initialize() 函数的定义 (位于 frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp) 如下：
status_t AudioPolicyManager::initialize() {
{
auto engLib = EngineLibrary::load(
"libaudiopolicyengine" + getConfig().getEngineLibraryNameSuffix() + ".so");
if (!engLib) {
ALOGE("%s: Failed to load the engine library", __FUNCTION__);
return NO_INIT;
}
mEngine = engLib->createEngine();
if (mEngine == nullptr) {
ALOGE("%s: Failed to instantiate the APM engine", __FUNCTION__);
return NO_INIT;
}
}
mEngine->setObserver(this);
status_t status = mEngine->initCheck();
if (status != NO_ERROR) {
LOG_FATAL("Policy engine not initialized(err=%d)", status);
return status;
}
mCommunnicationStrategy = mEngine->getProductStrategyForAttributes(
mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL));
// after parsing the config, mOutputDevicesAll and mInputDevicesAll contain all known devices;
// open all output streams needed to access attached devices
onNewAudioModulesAvailableInt(nullptr /*newDevices*/);
// make sure default device is reachable
if (mDefaultOutputDevice == 0 || !mAvailableOutputDevices.contains(mDefaultOutputDevice)) {
ALOGE_IF(mDefaultOutputDevice != 0, "Default device %s is unreachable",
mDefaultOutputDevice->toString().c_str());
status = NO_INIT;
}
// If microphones address is empty, set it according to device type
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->address().empty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
mAvailableInputDevices[i]->setAddress(AUDIO_BOTTOM_MICROPHONE_ADDRESS);
} else if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BACK_MIC) {
mAvailableInputDevices[i]->setAddress(AUDIO_BACK_MICROPHONE_ADDRESS);
}
}
}
ALOGW_IF(mPrimaryOutput == nullptr, "The policy configuration does not declare a primary output");
// Silence ALOGV statements
property_set("log.tag." LOG_TAG, "D");
updateDevicesAndOutputs();
return status;
}
AudioPolicyManager::initialize() 函数做了这样一些事情：
- 加载音频策略引擎，其中引擎库文件名后缀从音频策略配置获取；
- 创建音频策略引擎；
- 为音频策略引擎设置观察者，这里的 观察者 这个名称用的实际上不是很贴切，直观上会让人以为是
AudioPolicyManager要监听音频策略引擎的事件通知或状态变化，实际上这里是要让音频策略引擎可以获得设备相关的信息； - 初始化音频策略引擎；
- 初始化语音通话流类型的产品策略；
- 调用
onNewAudioModulesAvailableInt(nullptr /*newDevices*/)函数打开访问各个硬件模块的设备的输出音频流； - 确认默认输出设备可用；
- 如果麦克风地址为空，则根据设备类型设置它；
- 更新设备和输出。
上面为音频策略引擎注册的观察者接口 AudioPolicyManagerObserver 的定义 (位于 frameworks/av/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h) 如下：
class AudioPolicyManagerObserver
{
public:
virtual const AudioPatchCollection &getAudioPatches() const = 0;
virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const = 0;
virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const = 0;
virtual const SwAudioOutputCollection &getOutputs() const = 0;
virtual const AudioInputCollection &getInputs() const = 0;
virtual const DeviceVector getAvailableOutputDevices() const = 0;
virtual const DeviceVector getAvailableInputDevices() const = 0;
virtual const sp &getDefaultOutputDevice() const = 0;
protected:
virtual ~AudioPolicyManagerObserver() {}
};
这个 Observer 的名字用的，即使不能说是非常不贴切，也可以说是十分不贴切了。
onNewAudioModulesAvailableInt(nullptr /*newDevices*/) 函数的定义 (位于 frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp) 如下：
void AudioPolicyManager::onNewAudioModulesAvailableInt(DeviceVector *newDevices)
{
for (const auto& hwModule : mHwModulesAll) {
if (std::find(mHwModules.begin(), mHwModules.end(), hwModule) != mHwModules.end()) {
continue;
}
hwModule->setHandle(mpClientInterface->loadHwModule(hwModule->getName()));
if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
ALOGW("could not open HW module %s", hwModule->getName());
continue;
}
mHwModules.push_back(hwModule);
// open all output streams needed to access attached devices
// except for direct output streams that are only opened when they are actually
// required by an app.
// This also validates mAvailableOutputDevices list
for (const auto& outProfile : hwModule->getOutputProfiles()) {
if (!outProfile->canOpenNewIo()) {
ALOGE("Invalid Output profile max open count %u for profile %s",
outProfile->maxOpenCount, outProfile->getTagName().c_str());
continue;
}
if (!outProfile->hasSupportedDevices()) {
ALOGW("Output profile contains no device on module %s", hwModule->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
mTtsOutputAvailable = true;
}
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
DeviceVector availProfileDevices = supportedDevices.filter(mOutputDevicesAll);
sp supportedDevice = 0;
if (supportedDevices.contains(mDefaultOutputDevice)) {
supportedDevice = mDefaultOutputDevice;
} else {
// choose first device present in profile's SupportedDevices also part of
// mAvailableOutputDevices.
if (availProfileDevices.isEmpty()) {
continue;
}
supportedDevice = availProfileDevices.itemAt(0);
}
if (!mOutputDevicesAll.contains(supportedDevice)) {
continue;
}
sp outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
DeviceVector(supportedDevice),
AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for devices %s on hw module %s",
supportedDevice->toString().c_str(), hwModule->getName());
continue;
}
for (const auto &device : availProfileDevices) {
// give a valid ID to an attached device once confirmed it is reachable
if (!device->isAttached()) {
device->attach(hwModule);
mAvailableOutputDevices.add(device);
device->setEncapsulationInfoFromHal(mpClientInterface);
if (newDevices) newDevices->add(device);
setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
}
}
if (mPrimaryOutput == nullptr &&
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
outputDesc->close();
} else {
addOutput(output, outputDesc);
setOutputDevices(outputDesc,
DeviceVector(supportedDevice),
true,
0,
NULL);
}
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (const auto& inProfile : hwModule->getInputProfiles()) {
if (!inProfile->canOpenNewIo()) {
ALOGE("Invalid Input profile max open count %u for profile %s",
inProfile->maxOpenCount, inProfile->getTagName().c_str());
continue;
}
if (!inProfile->hasSupportedDevices()) {
ALOGW("Input profile contains no device on module %s", hwModule->getName());
continue;
}
// chose first device present in profile's SupportedDevices also part of
// available input devices
const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
DeviceVector availProfileDevices = supportedDevices.filter(mInputDevicesAll);
if (availProfileDevices.isEmpty()) {
ALOGV("%s: Input device list is empty! for profile %s",
__func__, inProfile->getTagName().c_str());
continue;
}
sp inputDesc =
new AudioInputDescriptor(inProfile, mpClientInterface);
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = inputDesc->open(nullptr,
availProfileDevices.itemAt(0),
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE,
&input);
if (status != NO_ERROR) {
ALOGW("Cannot open input stream for device %s on hw module %s",
availProfileDevices.toString().c_str(),
hwModule->getName());
continue;
}
for (const auto &device : availProfileDevices) {
// give a valid ID to an attached device once confirmed it is reachable
if (!device->isAttached()) {
device->attach(hwModule);
device->importAudioPortAndPickAudioProfile(inProfile, true);
mAvailableInputDevices.add(device);
if (newDevices) newDevices->add(device);
setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
}
}
inputDesc->close();
}
}
}
onNewAudioModulesAvailableInt(nullptr /*newDevices*/) 函数遍历前面加载的所有音频硬件模块的信息，针对每个音频硬件模块，执行如下这样的处理：
- 传入名称，请求加载音频硬件模块，这最终会请求
AudioFlinger， 继而请求 Audio HAL 服务加载音频硬件模块，返回音频硬件模块的句柄，更多相关内容可以参考 Android Audio HAL 服务； - 加载音频硬件模块失败，则继续处理下一个，成功则为音频硬件模块设置句柄，在一个专门的集合中保存音频硬件模块并继续；
- 遍历音频硬件模块的所有输出
IOProfile，针对其中的每一个IOProfile执行下面这样的操作：
3.1 检查是否可以打开新 IO，如果不行，则跳过当前的IOProfile，开始处理下一个，否则继续执行；
3.2 检查当前的IOProfile是否有支持的设备，如果没有，则跳过当前的IOProfile，开始处理下一个，否则继续执行；
3.3 检查IOProfile的标记，如果设置了 TTS 标记，则设置 TTS 可用；
3.4 获取IOProfile的支持的设备，并过滤一下得到可用设备列表，IOProfile的支持的设备是直接从音频策略配置文件里解析出来的，这里的mOutputDevicesAll中的设备是经过了音频策略配置文件中硬件模块的 attachedDevices 中的设备列表过滤的；
3.5 选择一个支持的设备，选择的方法为，如果IOProfile的支持的设备包含默认输出设备，则选择默认输出设备，否则选择可用设备中的第一个，这里多少给人一点脱裤子放屁的感觉，我们知道输出IOProfile的支持的设备一般最多有一个；
3.6 创建并打开音频输出描述符，这将会请求 AudioFlinger 打开一条音频流；
3.7 将设备与音频硬件模块挂接起来；
3.8 更新 primary 音频输出描述符；
3.7 如果IOProfile设置了 DIRECT 标记，则关闭音频输出描述符，否则，添加并设置它； - 遍历音频硬件模块的所有输入
IOProfile，针对其中的每一个IOProfile执行下面这样的操作：
4.1 检查是否可以打开新 IO，如果不行，则跳过当前的IOProfile，开始处理下一个，否则继续执行；
4.2 检查当前的IOProfile是否有支持的设备，如果没有，则跳过当前的IOProfile，开始处理下一个，否则继续执行；
4.3 获取IOProfile的支持的设备，并过滤一下得到可用设备列表，IOProfile的支持的设备是直接从音频策略配置文件里解析出来的，这里的mInputDevicesAll中的设备是经过了音频策略配置文件中硬件模块的 attachedDevices 中的设备列表过滤的；
4.4 如果可用设备列表为空，则跳过当前的IOProfile，开始处理下一个，否则继续执行；
4.5 创建并打开音频输入描述符，这将会请求 AudioFlinger 打开一条音频流；
4.6 将设备与音频硬件模块挂接起来；
4.7 关闭音频输入描述符。
可以看到，播放输出设备在 AudioPolicyManager 初始化期间被打开，并保持打开状态；输入设备则会做一次打开动作，但随后会被关闭。
音频输出描述符 SwAudioOutputDescriptor 的 open() 操作定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp) 如下：
status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
audio_io_handle_t *output)
{
mDevices = devices;
sp device = devices.getDeviceForOpening();
LOG_ALWAYS_FATAL_IF(device == nullptr,
"%s failed to get device descriptor for opening "
"with the requested devices, all device types: %s",
__func__, dumpDeviceTypes(devices.types()).c_str());
audio_config_t lHalConfig;
if (halConfig == nullptr) {
lHalConfig = AUDIO_CONFIG_INITIALIZER;
lHalConfig.sample_rate = mSamplingRate;
lHalConfig.channel_mask = mChannelMask;
lHalConfig.format = mFormat;
} else {
lHalConfig = *halConfig;
}
// if the selected profile is offloaded and no offload info was specified,
// create a default one
if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
lHalConfig.offload_info.format = lHalConfig.format;
lHalConfig.offload_info.stream_type = stream;
lHalConfig.offload_info.duration_us = -1;
lHalConfig.offload_info.has_video = true; // conservative
lHalConfig.offload_info.is_streaming = true; // likely
lHalConfig.offload_info.encapsulation_mode = lHalConfig.offload_info.encapsulation_mode;
lHalConfig.offload_info.content_id = lHalConfig.offload_info.content_id;
lHalConfig.offload_info.sync_id = lHalConfig.offload_info.sync_id;
}
audio_config_base_t lMixerConfig;
if (mixerConfig == nullptr) {
lMixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
lMixerConfig.sample_rate = lHalConfig.sample_rate;
lMixerConfig.channel_mask = lHalConfig.channel_mask;
lMixerConfig.format = lHalConfig.format;
} else {
lMixerConfig = *mixerConfig;
}
mFlags = (audio_output_flags_t)(mFlags | flags);
//TODO: b/193496180 use spatializer flag at audio HAL when available
audio_output_flags_t halFlags = mFlags;
if ((mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) {
halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
ALOGV("opening output for device %s profile %p name %s",
mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
&lHalConfig,
&lMixerConfig,
device,
&mLatency,
halFlags);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
"%s openOutput returned output handle %d for device %s, "
"selected device %s for opening",
__FUNCTION__, *output, devices.toString().c_str(),
device->toString().c_str());
mSamplingRate = lHalConfig.sample_rate;
mChannelMask = lHalConfig.channel_mask;
mFormat = lHalConfig.format;
mMixerChannelMask = lMixerConfig.channel_mask;
mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *output;
mProfile->curOpenCount++;
}
return status;
}
SwAudioOutputDescriptor 在 open() 操作中做的事情如下：
- 从可能的多个设备中选择一个；
- 构造打开输出设备时请求的参数，许多参数项会被配置一个默认值；
- 请求
AudioPolicyClientInterface打开输出设备，这最终将请求AudioFlinger打开输出设备，输出设备请求参数既是输入参数也是输出参数，在请求结束时，参数中将包含音频硬件输出流所使用的实际的参数； - 根据返回的参数，更新本地的一些参数配置。
从可能的多个设备中选择一个的方法 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp) 如下：
sp DeviceVector::getDevice(audio_devices_t type, const String8& address,
audio_format_t format) const
{
sp device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->type() == type) {
// If format is specified, match it and ignore address
// Otherwise if address is specified match it
// Otherwise always match
if (((address == "" || (itemAt(i)->address().compare(address.c_str()) == 0)) &&
format == AUDIO_FORMAT_DEFAULT) ||
(itemAt(i)->supportsFormat(format) && format != AUDIO_FORMAT_DEFAULT)) {
device = itemAt(i);
if (itemAt(i)->address().compare(address.c_str()) == 0) {
break;
}
}
}
}
ALOGV("DeviceVector::%s() for type %08x address \"%s\" found %p format %08x",
__func__, type, address.string(), device.get(), format);
return device;
}
. . . . . .
sp DeviceVector::getDeviceForOpening() const
{
if (isEmpty()) {
// Return nullptr if this collection is empty.
return nullptr;
} else if (areAllOfSameDeviceType(types(), audio_call_is_input_device)) {
// For input case, return the first one when there is only one device.
return size() > 1 ? nullptr : *begin();
} else if (areAllOfSameDeviceType(types(), audio_is_output_device)) {
// For output case, return the device descriptor according to apm strategy.
audio_devices_t deviceType = apm_extract_one_audio_device(types());
return deviceType == AUDIO_DEVICE_NONE ? nullptr :
getDevice(deviceType, String8(""), AUDIO_FORMAT_DEFAULT);
}
// Return null pointer if the devices are not all input/output device.
return nullptr;
}
对于输入设备来说，用列表中的第一个；对于输出设备来说，则先选择一个设备列表中包含的优先级最高的设备类型，再找到列表中第一个该设备类型的设备。输出设备类型的优先级排序从 apm_extract_one_audio_device() 函数的定义中可以看出来，这个函数定义 (位于 frameworks/av/services/audiopolicy/common/include/policy.h) 如下：
static inline audio_devices_t apm_extract_one_audio_device(
const android::DeviceTypeSet& deviceTypes) {
if (deviceTypes.empty()) {
return AUDIO_DEVICE_NONE;
} else if (deviceTypes.size() == 1) {
return *(deviceTypes.begin());
} else {
// Multiple device selection is either:
// - speaker + one other device: give priority to speaker in this case.
// - one A2DP device + another device: happens with duplicated output. In this case
// retain the device on the A2DP output as the other must not correspond to an active
// selection if not the speaker.
// - HDMI-CEC system audio mode only output: give priority to available item in order.
if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER) != 0) {
return AUDIO_DEVICE_OUT_SPEAKER;
} else if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER_SAFE) != 0) {
return AUDIO_DEVICE_OUT_SPEAKER_SAFE;
} else if (deviceTypes.count(AUDIO_DEVICE_OUT_HDMI_ARC) != 0) {
return AUDIO_DEVICE_OUT_HDMI_ARC;
} else if (deviceTypes.count(AUDIO_DEVICE_OUT_HDMI_EARC) != 0) {
return AUDIO_DEVICE_OUT_HDMI_EARC;
} else if (deviceTypes.count(AUDIO_DEVICE_OUT_AUX_LINE) != 0) {
return AUDIO_DEVICE_OUT_AUX_LINE;
} else if (deviceTypes.count(AUDIO_DEVICE_OUT_SPDIF) != 0) {
return AUDIO_DEVICE_OUT_SPDIF;
} else {
std::vector a2dpDevices = android::Intersection(
deviceTypes, android::getAudioDeviceOutAllA2dpSet());
if (a2dpDevices.empty() || a2dpDevices.size() > 1) {
ALOGW("%s invalid device combination: %s",
__func__, android::dumpDeviceTypes(deviceTypes).c_str());
}
return a2dpDevices.empty() ? AUDIO_DEVICE_NONE : a2dpDevices[0];
}
}
}
AudioPolicyClientInterface 请求 AudioFlinger 打开输出设备的详细定义 (位于 frameworks/av/services/audiopolicy/service/AudioPolicyClientImpl.cpp) 如下：
status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
audio_config_base_t *mixerConfig,
const sp& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
sp af = AudioSystem::get_audio_flinger();
if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
media::OpenOutputRequest request;
media::OpenOutputResponse response;
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*halConfig));
request.mixerConfig =
VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
status_t status = af->openOutput(request, &response);
if (status == OK) {
*output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
*halConfig =
VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
*latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral(response.latencyMs));
}
return status;
}
这个函数基本上就是构造输入，转换输出。
音频输入描述符 AudioInputDescriptor 的 open() 操作定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp) 大体类似：
status_t AudioInputDescriptor::open(const audio_config_t *config,
const sp &device,
audio_source_t source,
audio_input_flags_t flags,
audio_io_handle_t *input)
{
audio_config_t lConfig;
if (config == nullptr) {
lConfig = AUDIO_CONFIG_INITIALIZER;
lConfig.sample_rate = mSamplingRate;
lConfig.channel_mask = mChannelMask;
lConfig.format = mFormat;
} else {
lConfig = *config;
}
mDevice = device;
ALOGV("opening input for device %s profile %p name %s",
mDevice->toString().c_str(), mProfile.get(), mProfile->getName().c_str());
audio_devices_t deviceType = mDevice->type();
status_t status = mClientInterface->openInput(mProfile->getModuleHandle(),
input,
&lConfig,
&deviceType,
String8(mDevice->address().c_str()),
source,
flags);
LOG_ALWAYS_FATAL_IF(mDevice->type() != deviceType,
"%s openInput returned device %08x when given device %08x",
__FUNCTION__, mDevice->type(), deviceType);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*input == AUDIO_IO_HANDLE_NONE,
"%s openInput returned input handle %d for device %s",
__FUNCTION__, *input, mDevice->toString().c_str());
mSamplingRate = lConfig.sample_rate;
mChannelMask = lConfig.channel_mask;
mFormat = lConfig.format;
mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *input;
mProfile->curOpenCount++;
}
return status;
}
这里无需做过多解释。
AudioFlinger 加载硬件模块和打开输入输出设备
前面我们看到，AudioPolicyService 的 AudioPolicyManager 最终请求 AudioFlinger 加载音频硬件模块。AudioFlinger 加载音频硬件模块的操作定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp) 如下：
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
if (name == NULL) {
return AUDIO_MODULE_HANDLE_NONE;
}
if (!settingsAllowed()) {
return AUDIO_MODULE_HANDLE_NONE;
}
Mutex::Autolock _l(mLock);
AutoMutex lock(mHardwareLock);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock and AudioFlinger::mHardwareLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
sp dev;
int rc = mDevicesFactoryHal->openDevice(name, &dev);
if (rc) {
ALOGE("loadHwModule() error %d loading module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->initCheck();
mHardwareStatus = AUDIO_HW_IDLE;
if (rc) {
ALOGE("loadHwModule() init check error %d for module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
// Check and cache this HAL's level of support for master mute and master
// volume. If this is the first HAL opened, and it supports the get
// methods, use the initial values provided by the HAL as the current
// master mute and volume settings.
AudioHwDevice::Flags flags = static_cast(0);
if (0 == mAudioHwDevs.size()) {
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
float mv;
if (OK == dev->getMasterVolume(&mv)) {
mMasterVolume = mv;
}
mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
bool mm;
if (OK == dev->getMasterMute(&mm)) {
mMasterMute = mm;
}
}
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (OK == dev->setMasterVolume(mMasterVolume)) {
flags = static_cast(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
}
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (OK == dev->setMasterMute(mMasterMute)) {
flags = static_cast(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
}
mHardwareStatus = AUDIO_HW_IDLE;
if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_MSD) == 0) {
// An MSD module is inserted before hardware modules in order to mix encoded streams.
flags = static_cast(flags | AudioHwDevice::AHWD_IS_INSERT);
}
audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
AudioHwDevice *audioDevice = new AudioHwDevice(handle, name, dev, flags);
if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
mPrimaryHardwareDev = audioDevice;
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
mAudioHwDevs.add(handle, audioDevice);
ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle);
return handle;
}
这个操作的执行过程如下：
- 检查要打开的音频硬件模块是否已经打开，如果已经打开，则直接返回模块句柄，否则继续执行；
- 请求 Audio HAL 服务，打开音频硬件模块，获得设备 HAL 接口对象，关于 Audio HAL 更详细的内容，可以参考 Android Audio HAL 服务；
- 初始化设备 HAL 接口对象；
- 如果打开的音频硬件模块是第一个，则初始化主音量
mMasterVolume和主静音mMasterMute； - 为打开的设备 HAL 接口对象设置主音量和主静音；
- 基于设备 HAL 接口对象为要打开的音频硬件模块创建
AudioHwDevice对象，在AudioFlinger中，AudioHwDevice对象用来描述打开的音频硬件模块； - 如果要打开的音频硬件模块名称为
AUDIO_HARDWARE_MODULE_ID_PRIMARY，即 primary，则初始化主硬件设备等； - 保存为音频硬件模块创建的
AudioHwDevice对象，并返回模块句柄。
相同的实体在不同模块中的抽象不同，在不同模块间的概念不统一，AudioPolicyService 中的音频硬件模块在 AudioFlinger 中被称为设备 Device，AudioPolicyService 中的音频设备，在 AudioFlinger 中则被称为流 Stream。
音频输出描述符 SwAudioOutputDescriptor 在 open() 操作中请求 AudioFlinger 打开音频输出设备，打开音频输出设备的 AudioFlinger::openOutput() 函数定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp) 如下：
sp AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
audio_config_base_t *mixerConfig __unused,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType);
if (outHwDev == NULL) {
return 0;
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
} else {
// Audio Policy does not currently request a specific output handle.
// If this is ever needed, see openInput_l() for example code.
ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
return 0;
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
//halConfig->format = AUDIO_FORMAT_PCM_FLOAT;
//halConfig->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
//halConfig->format = AUDIO_FORMAT_PCM_32_BIT;
//halConfig->format = AUDIO_FORMAT_PCM_8_24_BIT;
// ALOGV("openOutput_l() upgrading format to %#08x", halConfig->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
//halConfig->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
//halConfig->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
AudioStreamOut *outputStream = NULL;
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
deviceType,
flags,
halConfig,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
if (status == NO_ERROR) {
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
sp thread =
new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady);
mMmapThreads.add(*output, thread);
ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
*output, thread.get());
return thread;
} else {
sp thread;
//TODO: b/193496180 use spatializer flag at audio HAL when available
if (flags == (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST
| AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
#ifdef MULTICHANNEL_EFFECT_CHAIN
thread = new SpatializerThread(this, outputStream, *output,
mSystemReady, mixerConfig);
ALOGD("openOutput_l() created spatializer output: ID %d thread %p",
*output, thread.get());
#else
ALOGE("openOutput_l() cannot create spatializer thread "
"without #define MULTICHANNEL_EFFECT_CHAIN");
#endif
} else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(halConfig->format)
|| !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
thread = new MixerThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
mPlaybackThreads.add(*output, thread);
struct audio_patch patch;
mPatchPanel.notifyStreamOpened(outHwDev, *output, &patch);
if (thread->isMsdDevice()) {
thread->setDownStreamPatch(&patch);
}
return thread;
}
}
return 0;
}
status_t AudioFlinger::openOutput(const media::OpenOutputRequest& request,
media::OpenOutputResponse* response)
{
audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_module_handle_t(request.module));
audio_config_t halConfig = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.halConfig));
audio_config_base_t mixerConfig = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfigBase_audio_config_base_t(request.mixerConfig));
sp device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_output_flags_t_mask(request.flags));
audio_io_handle_t output;
uint32_t latencyMs;
ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
halConfig.sample_rate,
halConfig.format,
halConfig.channel_mask,
flags);
audio_devices_t deviceType = device->type();
const String8 address = String8(device->address().c_str());
if (deviceType == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
sp thread = openOutput_l(module, &output, &halConfig,
&mixerConfig, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
latencyMs = playbackThread->latency();
// notify client processes of the new output creation
playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
// the first primary output opened designates the primary hw device if no HW module
// named "primary" was already loaded.
AutoMutex lock(mHardwareLock);
if ((mPrimaryHardwareDev == nullptr) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d as the primary audio interface", module);
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config =
VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(halConfig));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral(latencyMs));
response->flags = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
return NO_ERROR;
}
return NO_INIT;
}
这个操作的执行过程如下：
- 根据音频硬件模块句柄和设备类型查找适当的音频硬件模块；
- 构造音频输出设备 ID；
- 请求音频硬件模块打开音频输出流，
AudioFlinger中的流等价于AudioPolicyService中的设备； - 根据音频硬件设备的标记配置创建不同的 Thread，这些标记主要来自于音频策略 XML 配置文件。
根据音频硬件模块句柄和设备类型查找适当的音频硬件模块的 findSuitableHwDev_l() 函数定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp) 如下：
static const char * const audio_interfaces[] = {
AUDIO_HARDWARE_MODULE_ID_PRIMARY,
AUDIO_HARDWARE_MODULE_ID_A2DP,
AUDIO_HARDWARE_MODULE_ID_USB,
};
AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t deviceType)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
AutoMutex lock(mHardwareLock);
if (module == 0) {
ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
for (size_t i = 0; i < arraysize(audio_interfaces); i++) {
loadHwModule_l(audio_interfaces[i]);
}
// then try to find a module supporting the requested device.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
sp dev = audioHwDevice->hwDevice();
uint32_t supportedDevices;
if (dev->getSupportedDevices(&supportedDevices) == OK &&
(supportedDevices & deviceType) == deviceType) {
return audioHwDevice;
}
}
} else {
// check a match for the requested module handle
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
在这个函数中，如果音频硬件模块句柄为 0，则会先加载几个音频硬件模块，然后找到第一个支持请求的音频设备类型的音频硬件模块，并返回；如果音频硬件模块句柄不为 0，则根据句柄查找并返回。
加载音频硬件模块不只会在 AudioPolicyService 中发起，当向 AudioFlinger 请求打开输入输出音频设备，查找音频硬件模块时也可能发起。
在音频硬件模块 AudioHwDevice 中打开音频输出流的操作定义 (位于 frameworks/av/services/audioflinger/AudioHwDevice.cpp) 如下：
status_t AudioHwDevice::openOutputStream(
AudioStreamOut **ppStreamOut,
audio_io_handle_t handle,
audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address)
{
struct audio_config originalConfig = *config;
AudioStreamOut *outputStream = new AudioStreamOut(this, flags);
// Try to open the HAL first using the current format.
ALOGV("openOutputStream(), try "
" sampleRate %d, Format %#x, "
"channelMask %#x",
config->sample_rate,
config->format,
config->channel_mask);
status_t status = outputStream->open(handle, deviceType, config, address);
if (status != NO_ERROR) {
delete outputStream;
outputStream = NULL;
// FIXME Look at any modification to the config.
// The HAL might modify the config to suggest a wrapped format.
// Log this so we can see what the HALs are doing.
ALOGI("openOutputStream(), HAL returned"
" sampleRate %d, Format %#x, "
"channelMask %#x, status %d",
config->sample_rate,
config->format,
config->channel_mask,
status);
// If the data is encoded then try again using wrapped PCM.
bool wrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
&& ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
&& ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
if (wrapperNeeded) {
if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
status = outputStream->open(handle, deviceType, &originalConfig, address);
if (status != NO_ERROR) {
ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
status);
delete outputStream;
outputStream = NULL;
}
} else {
ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
originalConfig.format);
}
}
}
*ppStreamOut = outputStream;
return status;
}
这里创建 AudioStreamOut 对象并执行其 open() 操作。当 open() 执行失败时，则可能会根据配置的音频数据格式，尝试创建 SpdifStreamOut 对象并执行其 open() 操作。
AudioStreamOut 的 open() 操作定义 (位于 frameworks/av/services/audioflinger/AudioStreamOut.cpp) 如下：
status_t AudioStreamOut::open(
audio_io_handle_t handle,
audio_devices_t deviceType,
struct audio_config *config,
const char *address)
{
sp outStream;
audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
? (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
: flags;
int status = hwDev()->openOutputStream(
handle,
deviceType,
customFlags,
config,
address,
&outStream);
ALOGD("AudioStreamOut::open(), HAL returned "
" stream %p, sampleRate %d, Format %#x, "
"channelMask %#x, status %d",
outStream.get(),
config->sample_rate,
config->format,
config->channel_mask,
status);
// Some HALs may not recognize AUDIO_FORMAT_IEC61937. But if we declare
// it as PCM then it will probably work.
if (status != NO_ERROR && config->format == AUDIO_FORMAT_IEC61937) {
struct audio_config customConfig = *config;
customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
status = hwDev()->openOutputStream(
handle,
deviceType,
customFlags,
&customConfig,
address,
&outStream);
ALOGV("AudioStreamOut::open(), treat IEC61937 as PCM, status = %d", status);
}
if (status == NO_ERROR) {
stream = outStream;
mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
status = stream->getFrameSize(&mHalFrameSize);
LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
LOG_ALWAYS_FATAL_IF(mHalFrameSize <= 0, "Error frame size was %zu but must be greater than"
" zero", mHalFrameSize);
}
return status;
}
这里请求 Audio HAL 服务打开音频输出流，当失败时，根据配置的音频数据格式，则可能会尝试换一种音频数据格式再打开一次。
AudioFlinger 在打开音频输入输出设备时，创建的各种 Thread 具有如下这样的继承层次结构：
音频输入描述符 AudioInputDescriptor 在 open() 操作中请求 AudioFlinger 打开音频输入设备，打开音频输入设备的 AudioFlinger::openInput() 函数定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp) 如下：
status_t AudioFlinger::openInput(const media::OpenInputRequest& request,
media::OpenInputResponse* response)
{
Mutex::Autolock _l(mLock);
if (request.device.type == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
audio_io_handle_t input = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_io_handle_t(request.input));
audio_config_t config = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.config));
AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioDeviceTypeAddress(request.device));
sp thread = openInput_l(
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
&input,
&config,
device.mType,
device.address().c_str(),
VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSourceType_audio_source_t(request.source)),
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_input_flags_t_mask(request.flags)),
AUDIO_DEVICE_NONE,
String8{});
response->input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
response->device = request.device;
if (thread != 0) {
// notify client processes of the new input creation
thread->ioConfigChanged(AUDIO_INPUT_OPENED);
return NO_ERROR;
}
return NO_INIT;
}
sp AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
const char* address,
audio_source_t source,
audio_input_flags_t flags,
audio_devices_t outputDevice,
const String8& outputDeviceAddress)
{
AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
// Audio Policy can request a specific handle for hardware hotword.
// The goal here is not to re-open an already opened input.
// It is to use a pre-assigned I/O handle.
if (*input == AUDIO_IO_HANDLE_NONE) {
*input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
} else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
ALOGE("openInput_l() requested input handle %d is invalid", *input);
return 0;
} else if (mRecordThreads.indexOfKey(*input) >= 0) {
// This should not happen in a transient state with current design.
ALOGE("openInput_l() requested input handle %d is already assigned", *input);
return 0;
}
audio_config_t halconfig = *config;
sp inHwHal = inHwDev->hwDevice();
sp inStream;
status_t status = inHwHal->openInputStream(
*input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
", Format %#x, Channels %#x, flags %#x, status %d addr %s",
inStream.get(),
devices,
halconfig.sample_rate,
halconfig.format,
halconfig.channel_mask,
flags,
status, address);
// If the input could not be opened with the requested parameters and we can handle the
// conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
audio_is_linear_pcm(config->format) &&
audio_is_linear_pcm(halconfig.format) &&
(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream.clear();
status = inHwHal->openInputStream(
*input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
// FIXME log this new status; HAL should not propose any further changes
}
if (status == NO_ERROR && inStream != 0) {
AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
sp thread =
new MmapCaptureThread(this, *input, inHwDev, inputStream, mSystemReady);
mMmapThreads.add(*input, thread);
ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
thread.get());
return thread;
} else {
// Start record thread
// RecordThread requires both input and output device indication to forward to audio
// pre processing modules
sp thread = new RecordThread(this, inputStream, *input, mSystemReady);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
}
}
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
这个操作的执行过程如下：
- 根据音频硬件模块句柄和设备类型查找适当的音频硬件模块；
- 构造音频输入设备 ID；
- 直接请求设备 HAL 接口对象打开音频输入流；
- 上一步中以请求的参数打开音频输入流失败时，还会处理内部转换，并尝试再次以适当的参数打开；
- 为音频输入流创建
AudioStreamIn对象； - 根据音频硬件设备的标记配置创建不同的 Thread，这些标记主要来自于音频策略 XML 配置文件。