在進行通道切換時,為什麼會在原通道上設定一回在去設定新的通道
一、Application framework
在Application framework層級是app層的code,是通過android.media提供的API來與audio硬體進行互動動作,這部分的代碼是通過audio JNI來調用native代碼進而達到影響硬體的效果;
二、JNI
JNI部分的代碼是位于 frameworks/base/core/jni/和frameworks/base/media/jni 目錄下的;
三、Native framework
四、Binder IPC
Binder IPC通信是跨程序通信的手段,audio的這部分代碼位于frameworks/av/media/libmedia目錄下,并且命名都是以I開頭的;
五、Media Server
Audio Service是隸屬Media Server的,其代碼位于 frameworks/av/services/audioflinger,它是真正的與HAL層的實作進行互動的;
六、HAL
HAL層定義了Audio Service調用的标準接口,不同的硬體必須根據自己的情況來實作這個接口來讓硬體在android中正常的工作,是以可以在不影響應用層系統調用的情況下,更換不同的硬體。大大減少了系統耦合性;
七、Kernel Driver
Audio驅動是與硬體進行互動,并且實作HAL層的接口供上層正常調用,這裡,廠商可以選擇ALSA,OSS以及自定義的音頻驅動; (NOTE:如果選擇ALSA,android建議使用 external/tinyalsa目錄下的實作); 接下來就來說說通話時音頻通道的切換,但是往下看之前必須知道,對于Audio Path的切換,android有一政策管理器來幫我們配置設定好輸入輸出的裝置,比如當手機播放音樂時,從Speaker播放出來,這時候插入耳機的話會從耳機裝置輸出;但是有時候我們想要自己去指定的話,就是我們接下來要說的了; 我們在通話時,要是開免提,實際上也就是Audio Path切換到了Speaker,也就是外方喇叭;代碼中的話調用一個函數即可,這是強制切換audio Path,不遵從系統的配置設定:
中間過程簡單不說,最終是調用到了JNI,android_media_AudioSystem中的android_media_AudioSystem_setForceUse()函數,來看下其具體實作:
android_media_AudioSystem_setForceUse(JNIEnv *env, jobject thiz, jint usage, jint config)
{
return check_AudioSystem_Command(AudioSystem::setForceUse(static_cast <audio_policy_force_use_t>(usage),
static_cast <audio_policy_forced_cfg_t>(config)));
}
顯而易見,它是調用了AudioSystem.cpp的setForceUse()函數,check_AudioSystem_Command()不說,重點看看audio_policy_force_use_t和audio_policy_forced_cfg_t這兩個結構體:
audio_policy_force_use_t 說明的是目前的Audio環境
audio_policy_forced_cfg_t 表示audio的輸入輸出裝置
它們是專門為setForceUse所用的;
/* usages used for audio_policy->set_force_use() */
typedef enum {
AUDIO_POLICY_FORCE_FOR_COMMUNICATION, //表示的是通話過程中
AUDIO_POLICY_FORCE_FOR_MEDIA, //媒體
AUDIO_POLICY_FORCE_FOR_RECORD, //錄音
AUDIO_POLICY_FORCE_FOR_DOCK,
AUDIO_POLICY_FORCE_FOR_SYSTEM,
AUDIO_POLICY_FORCE_USE_CNT,
AUDIO_POLICY_FORCE_USE_MAX = AUDIO_POLICY_FORCE_USE_CNT - ,
} audio_policy_force_use_t;
/* device categories used for audio_policy->set_force_use() */
typedef enum {
AUDIO_POLICY_FORCE_NONE,
AUDIO_POLICY_FORCE_SPEAKER,
AUDIO_POLICY_FORCE_HEADPHONES,
AUDIO_POLICY_FORCE_BT_SCO,
AUDIO_POLICY_FORCE_BT_A2DP,
AUDIO_POLICY_FORCE_WIRED_ACCESSORY,
AUDIO_POLICY_FORCE_BT_CAR_DOCK,
AUDIO_POLICY_FORCE_BT_DESK_DOCK,
AUDIO_POLICY_FORCE_ANALOG_DOCK,
AUDIO_POLICY_FORCE_DIGITAL_DOCK,
AUDIO_POLICY_FORCE_NO_BT_A2DP,
/* A2DP sink is not preferred to speaker or wired HS */
AUDIO_POLICY_FORCE_SYSTEM_ENFORCED,
AUDIO_POLICY_FORCE_CFG_CNT,
AUDIO_POLICY_FORCE_DEFAULT = AUDIO_POLICY_FORCE_NONE,
} audio_policy_forced_cfg_t;
這時候我們就應該知道,當我想要在通話時打開Speaker,傳遞的參數就是usage和config分别是AUDIO_POLICY_FORCE_FOR_COMMUNICATION和AUDIO_POLICY_FORCE_SPEAKER了,這兩個參數從上層一直到底層,還是很簡單的;
接着往下看就是調用的AudioSystem.cpp的setForceUse()函數了:
status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
SLOGE("setForceUse() usage = %d, config = %d" ,usage , config);
if (aps == ) return PERMISSION_DENIED; return aps->setForceUse(usage, config);
}
get_audio_policy_service()函數不做過多解釋,就是通過Native的ServiceManager來擷取audio policy的Service代理對象,進而實作與audio policy的程序間通訊;
.......
binder = sm->getService(String16("media.audio_policy"));
接下來就是調用frameworks/av/services/audioflinger/AudioPolicyService.cpp的setForceUse()函數了;
status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (usage < || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
return BAD_VALUE;
}
if (config < || config >= AUDIO_POLICY_FORCE_CFG_CNT) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config);
return NO_ERROR;
}
這個mpAudioPolicy是什麼呢?它的set_force_use函數在哪裡實作呢?這兩個問題需要了解就OK了; 首先mpAudioPolicy它是一個指針,在AudioServicePolicy.cpp的構造函數中被指派,來看看其指派過程:
......
const struct hw_module_t *module;
......
rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
......
rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
......
rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev,&aps_ops,this, &mpAudioPolicy);
......
首先AUDIO_POLICY_HARDWARE_MODULE_ID值是:
其次module是一個指針,指向的是一個hw_module_t結構體類型,它的作用是調用系統的哪個audio policy module,這個module可以是原始的,也可以由廠商自定義的
typedef struct hw_module_t {
/** tag must be initialized to HARDWARE_MODULE_TAG */
uint32_t tag;
uint16_t module_api_version;
#define version_major module_api_version
uint16_t hal_api_version;
#define version_minor hal_api_version
/** Identifier of module */
const char *id;
const char *name;
const char *author;
/** Modules methods */
struct hw_module_methods_t* methods;
/** module's dso */
void* dso;
/** padding to 128 bytes, reserved for future use */
uint32_t reserved[-];
} hw_module_t;
再來看看是如何給module指派的: hardware.c
int hw_get_module(const char *id, const struct hw_module_t **module)
{
return hw_get_module_by_class(id, NULL, module);
}
看看hw_get_module_by_class方法的實作: hardware.c
int hw_get_module_by_class(const char *class_id, const char *inst, const struct hw_module_t **module) {
int status;
int i; const struct hw_module_t *hmi = NULL;
char prop[PATH_MAX]; char path[PATH_MAX];
char name[PATH_MAX];
if (inst)
snprintf(name, PATH_MAX, "%s.%s", class_id, inst);
else
strlcpy(name, class_id, PATH_MAX);
/* * Here we rely on the fact that calling dlopen multiple times on
* the same .so will simply increment a refcount (and not load
* a new copy of the library).
* We also assume that dlopen() is thread-safe.
*/
/* Loop through the configuration variants looking for a module */
for (i= ; i<HAL_VARIANT_KEYS_COUNT+ ; i++)
{
if (i < HAL_VARIANT_KEYS_COUNT) {
if (property_get(variant_keys[i], prop, NULL) == ) {
continue;
}
snprintf(path, sizeof(path), "%s/%s.%s.so", HAL_LIBRARY_PATH2, name, prop);
if (access(path, R_OK) == )
break;
snprintf(path, sizeof(path), "%s/%s.%s.so", HAL_LIBRARY_PATH1, name, prop);
if (access(path, R_OK) == )
break;
}
else {
snprintf(path, sizeof(path), "%s/%s.default.so", HAL_LIBRARY_PATH1, name);
if (access(path, R_OK) == )
break;
}
}
status = -ENOENT;
if (i < HAL_VARIANT_KEYS_COUNT+) {
/* load the module, if this fails, we're doomed, and we should not try
* to load a different variant.
*/
status = load(class_id, path, module);
}
return status;
}
方法是找到指定的庫檔案并且加載;不做詳細介紹;這裡會得到audio_policy.default.so;這個庫正是編譯hardware/libhardware_legacy/audio出來的; 再跳回到AudioPolicyService的構造函數中來;接下來 :
java
它調用的是legacy_ap_dev_open()函數,不做詳細介紹:audio_policy_hal.cpp
static int legacy_ap_dev_open(const hw_module_t* module, const char* name, hw_device_t** device)
{
struct legacy_ap_device *dev;
if (strcmp(name, AUDIO_POLICY_INTERFACE) != )
return -EINVAL;
dev = (struct legacy_ap_device *)calloc(, sizeof(*dev));
if (!dev) return -ENOMEM;
dev->device.common.tag = HARDWARE_DEVICE_TAG;
dev->device.common.version = ;
dev->device.common.module = const_cast<hw_module_t*>(module);
dev->device.common.close = legacy_ap_dev_close;
dev->device.create_audio_policy = create_legacy_ap;
dev->device.destroy_audio_policy = destroy_legacy_ap;
*device = &dev->device.common; return ;
}
create_audio_policy()中的aps_ops參數指針代表的是,它是AudioPolicyService與外界互動的接口:
struct audio_policy_service_ops aps_ops = {
open_output : aps_open_output,
open_duplicate_output : aps_open_dup_output,
close_output : aps_close_output,
suspend_output : aps_suspend_output,
restore_output : aps_restore_output,
open_input : aps_open_input,
close_input : aps_close_input,
set_stream_volume : aps_set_stream_volume,
set_stream_output : aps_set_stream_output,
set_parameters : aps_set_parameters,
get_parameters : aps_get_parameters,
start_tone : aps_start_tone,
stop_tone : aps_stop_tone,
set_voice_volume : aps_set_voice_volume,
move_effects : aps_move_effects,
load_hw_module : aps_load_hw_module,
open_output_on_module : aps_open_output_on_module,
open_input_on_module : aps_open_input_on_module,
};
知道了這些,接下來看create_audio_policy(): create_audio_policy()這個函數作用是建立一個使用者自定義的policy_hal子產品的接口,因為我們使用的是qcom的晶片,qcom有自己的一套,android原生有自己的一套,就依照原生的來看吧;其實都是差不多的; 剛剛上面分析的legacy_ap_dev_open()函數有這樣一句:
......
dev->device.create_audio_policy = create_legacy_ap;
......
那這樣我們就來看看其create_legacy_ap()函數吧;我們隻需要關注的是其中的那麼幾小段:
static int create_legacy_ap(const struct audio_policy_device *device, struct audio_policy_service_ops *aps_ops, void *service, struct audio_policy **ap)
{
struct legacy_audio_policy *lap;
......
lap = (struct legacy_audio_policy *)calloc(, sizeof(*lap));
......
lap->policy.set_force_use = ap_set_force_use;
......
lap->service = service;
lap->aps_ops = aps_ops;
lap->service_client = new AudioPolicyCompatClient(aps_ops, service);
......
lap->apm = createAudioPolicyManager(lap->service_client);
......
*ap = &lap->policy;
......
}
就這樣,AudioPolicyService.cpp的set_force_use()函數就調用到了這裡: audio_policy_hal.cpp
/* force using a specific device category for the specified usage */
static void ap_set_force_use(struct audio_policy *pol, audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
struct legacy_audio_policy *lap = to_lap(pol);
lap->apm->setForceUse((AudioSystem::force_use)usage, (AudioSystem::forced_config)config);
}
從之前的create_legacy_ap()函數我們知道apm的由來,
java lap->apm = createAudioPolicyManager(lap->service_client);
createAudioPolicyManager()函數定義在AudioPolicyInterface.h接口中;
extern "C"
AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
而這個createAudioPolicyManager()由硬體廠商實作,傳回其AudioPolicyManager;qcom的實作是在AudioPolicyManagerALSA.cpp中;再往下不做具體分析了,主要是根據不同的政策來切換不同的Output和input裝置以及其他一些操作;如果想進一步分析的話,還需要關注AudioPolicyManagerBase.cpp; 其實準确的總結起來是AudioPolicyService是一個殼子,這個殼子的重要關鍵就是audio_policy,真正的實作可以由廠商來自己實作,當然android也有,就是AudioPolicyManagerDefault