Miscellaneous code cleanup in audio framework
Changes:
- Move declaration of kClassPathName to top of file so it can be used
in more than one place, instead of "android/media/AudioSystem".
- Make private methods static.
- Add comment to stream_type, audio_mode, force_use types that they must match
values in AudioSystem.java.
- Add comment about unused types mp3_sub_format and vorbis_sub_format.
- Fix typos.
- Use @ in javadoc comments.
- Delete dead APIs setMode, getMode, setRouting, getRouting in AudioSystem.java
(they are all hidden, deprecated, and unused by rest of framework)
- Delete unused private log method.
- Fix pathname for android_media_AudioSystem.cpp.
- Improve code formatting for space after == and !=.
- Add logging of delta for changing audio policy manager ref count.
Change-Id: I18037c7beb8ab76d1fda08c11e589f6e591d36e1
diff --git a/core/jni/android_media_AudioSystem.cpp b/core/jni/android_media_AudioSystem.cpp
index 5f3fed2..5016bf9 100644
--- a/core/jni/android_media_AudioSystem.cpp
+++ b/core/jni/android_media_AudioSystem.cpp
@@ -34,6 +34,8 @@
using namespace android;
+static const char* const kClassPathName = "android/media/AudioSystem";
+
enum AudioError {
kAudioStatusOk = 0,
kAudioStatusError = 1,
@@ -96,14 +98,15 @@
return env->NewStringUTF(AudioSystem::getParameters(0, c_keys8).string());
}
-void android_media_AudioSystem_error_callback(status_t err)
+static void
+android_media_AudioSystem_error_callback(status_t err)
{
JNIEnv *env = AndroidRuntime::getJNIEnv();
if (env == NULL) {
return;
}
- jclass clazz = env->FindClass("android/media/AudioSystem");
+ jclass clazz = env->FindClass(kClassPathName);
int error;
@@ -218,12 +221,10 @@
{"getDevicesForStream", "(I)I", (void *)android_media_AudioSystem_getDevicesForStream},
};
-const char* const kClassPathName = "android/media/AudioSystem";
-
int register_android_media_AudioSystem(JNIEnv *env)
{
AudioSystem::setErrorCallback(android_media_AudioSystem_error_callback);
return AndroidRuntime::registerNativeMethods(env,
- "android/media/AudioSystem", gMethods, NELEM(gMethods));
+ kClassPathName, gMethods, NELEM(gMethods));
}
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 2dc4beb..edf4b8b7 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -33,6 +33,7 @@
{
public:
+ // must match android/media/AudioSystem.java STREAM_* constants
enum stream_type {
DEFAULT =-1,
VOICE_CALL = 0,
@@ -54,6 +55,8 @@
PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility
};
+ // FIXME These sub_format enums are currently unused
+
// MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
// bit rate, stereo mode, version...
enum mp3_sub_format {
@@ -100,7 +103,7 @@
};
- // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
+ // Channel mask definitions must be kept in sync with values in /media/java/android/media/AudioFormat.java
enum audio_channels {
// output channels
CHANNEL_OUT_FRONT_LEFT = 0x4,
@@ -150,6 +153,7 @@
CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
};
+ // must match android/media/AudioSystem.java MODE_* values
enum audio_mode {
MODE_INVALID = -2,
MODE_CURRENT = -1,
@@ -189,6 +193,7 @@
// set/get master volume
static status_t setMasterVolume(float value);
static status_t getMasterVolume(float* volume);
+
// mute/unmute audio outputs
static status_t setMasterMute(bool mute);
static status_t getMasterMute(bool* mute);
@@ -234,7 +239,7 @@
static status_t setVoiceVolume(float volume);
// return the number of audio frames written by AudioFlinger to audio HAL and
- // audio dsp to DAC since the output on which the specificed stream is playing
+ // audio dsp to DAC since the output on which the specified stream is playing
// has exited standby.
// returned status (from utils/Errors.h) can be:
// - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
@@ -321,7 +326,7 @@
FORCE_DEFAULT = FORCE_NONE
};
- // usages used for setForceUse()
+ // usages used for setForceUse(), must match AudioSystem.java
enum force_use {
FOR_COMMUNICATION,
FOR_MEDIA,
diff --git a/media/java/android/media/AudioSystem.java b/media/java/android/media/AudioSystem.java
index 2492d47..95d93b2 100644
--- a/media/java/android/media/AudioSystem.java
+++ b/media/java/android/media/AudioSystem.java
@@ -64,44 +64,20 @@
/*
* Sets the microphone mute on or off.
*
- * param on set <var>true</var> to mute the microphone;
+ * @param on set <var>true</var> to mute the microphone;
* <var>false</var> to turn mute off
- * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
+ * @return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
*/
public static native int muteMicrophone(boolean on);
/*
* Checks whether the microphone mute is on or off.
*
- * return true if microphone is muted, false if it's not
+ * @return true if microphone is muted, false if it's not
*/
public static native boolean isMicrophoneMuted();
- /*
- * Sets the audio mode.
- *
- * param mode the requested audio mode (NORMAL, RINGTONE, or IN_CALL).
- * Informs the HAL about the current audio state so that
- * it can route the audio appropriately.
- * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
- */
- /** @deprecated use {@link #setPhoneState(int)} */
- public static int setMode(int mode) {
- return AUDIO_STATUS_ERROR;
- }
- /*
- * Returns the current audio mode.
- *
- * return the current audio mode (NORMAL, RINGTONE, or IN_CALL).
- * Returns the current current audio state from the HAL.
- *
- */
- /** @deprecated Do not use. */
- public static int getMode() {
- return MODE_INVALID;
- }
-
- /* modes for setPhoneState */
+ /* modes for setPhoneState, must match AudioSystem.h audio_mode */
public static final int MODE_INVALID = -2;
public static final int MODE_CURRENT = -1;
public static final int MODE_NORMAL = 0;
@@ -111,7 +87,7 @@
public static final int NUM_MODES = 4;
- /* Routing bits for setRouting/getRouting API */
+ /* Routing bits for the former setRouting/getRouting API */
/** @deprecated */
@Deprecated public static final int ROUTE_EARPIECE = (1 << 0);
/** @deprecated */
@@ -128,33 +104,6 @@
@Deprecated public static final int ROUTE_ALL = 0xFFFFFFFF;
/*
- * Sets the audio routing for a specified mode
- *
- * param mode audio mode to change route. E.g., MODE_RINGTONE.
- * param routes bit vector of routes requested, created from one or
- * more of ROUTE_xxx types. Set bits indicate that route should be on
- * param mask bit vector of routes to change, created from one or more of
- * ROUTE_xxx types. Unset bits indicate the route should be left unchanged
- * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
- */
- /** @deprecated use {@link #setDeviceConnectionState(int,int,String)} */
- public static int setRouting(int mode, int routes, int mask) {
- return AUDIO_STATUS_ERROR;
- }
-
- /*
- * Returns the current audio routing bit vector for a specified mode.
- *
- * param mode audio mode to change route (e.g., MODE_RINGTONE)
- * return an audio route bit vector that can be compared with ROUTE_xxx
- * bits
- */
- /** @deprecated use {@link #getDeviceConnectionState(int,String)} */
- public static int getRouting(int mode) {
- return 0;
- }
-
- /*
* Checks whether the specified stream type is active.
*
* return true if any track playing on this stream is active.
@@ -163,7 +112,7 @@
/*
* Sets a group generic audio configuration parameters. The use of these parameters
- * are platform dependant, see libaudio
+ * are platform dependent, see libaudio
*
* param keyValuePairs list of parameters key value pairs in the form:
* key1=value1;key2=value2;...
@@ -172,7 +121,7 @@
/*
* Gets a group generic audio configuration parameters. The use of these parameters
- * are platform dependant, see libaudio
+ * are platform dependent, see libaudio
*
* param keys list of parameters
* return value: list of parameters key value pairs in the form:
@@ -180,15 +129,7 @@
*/
public static native String getParameters(String keys);
- /*
- private final static String TAG = "audio";
-
- private void log(String msg) {
- Log.d(TAG, "[AudioSystem] " + msg);
- }
- */
-
- // These match the enum in libs/android_runtime/android_media_AudioSystem.cpp
+ // These match the enum AudioError in frameworks/base/core/jni/android_media_AudioSystem.cpp
/* Command sucessful or Media server restarted. see ErrorCallback */
public static final int AUDIO_STATUS_OK = 0;
/* Command failed or unspecified audio error. see ErrorCallback */
@@ -215,7 +156,7 @@
/*
* Registers a callback to be invoked when an error occurs.
- * param cb the callback to run
+ * @param cb the callback to run
*/
public static void setErrorCallback(ErrorCallback cb)
{
@@ -272,16 +213,17 @@
public static final int DEVICE_IN_AUX_DIGITAL = 0x800000;
public static final int DEVICE_IN_DEFAULT = 0x80000000;
- // device states
+ // device states, must match AudioSystem::device_connection_state
public static final int DEVICE_STATE_UNAVAILABLE = 0;
public static final int DEVICE_STATE_AVAILABLE = 1;
+ private static final int NUM_DEVICE_STATES = 1;
- // phone state
+ // phone state, match audio_mode???
public static final int PHONE_STATE_OFFCALL = 0;
public static final int PHONE_STATE_RINGING = 1;
public static final int PHONE_STATE_INCALL = 2;
- // config for setForceUse
+ // device categories config for setForceUse, must match AudioSystem::forced_config
public static final int FORCE_NONE = 0;
public static final int FORCE_SPEAKER = 1;
public static final int FORCE_HEADPHONES = 2;
@@ -292,13 +234,15 @@
public static final int FORCE_BT_DESK_DOCK = 7;
public static final int FORCE_ANALOG_DOCK = 8;
public static final int FORCE_DIGITAL_DOCK = 9;
+ private static final int NUM_FORCE_CONFIG = 10;
public static final int FORCE_DEFAULT = FORCE_NONE;
- // usage for serForceUse
+ // usage for setForceUse, must match AudioSystem::force_use
public static final int FOR_COMMUNICATION = 0;
public static final int FOR_MEDIA = 1;
public static final int FOR_RECORD = 2;
public static final int FOR_DOCK = 3;
+ private static final int NUM_FORCE_USE = 4;
public static native int setDeviceConnectionState(int device, int state, String device_address);
public static native int getDeviceConnectionState(int device, String device_address);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2702242..52135e7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -5712,7 +5712,7 @@
const uint32_t i = 31 - __builtin_clz(device);
device &= ~(1 << i);
if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
- LOGE("device convertion error for AudioSystem device 0x%08x", device);
+ LOGE("device conversion error for AudioSystem device 0x%08x", device);
return 0;
}
deviceOut |= (uint32_t)sDeviceConvTable[i];
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
index f653dc5..c0ac669 100644
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ b/services/audioflinger/AudioPolicyManagerBase.cpp
@@ -542,7 +542,7 @@
}
- LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+ LOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
stream, samplingRate, format, channels, flags);
return output;
@@ -2107,7 +2107,7 @@
uint32_t device)
{
return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
- (format !=0 && !AudioSystem::isLinearPCM(format)));
+ (format != 0 && !AudioSystem::isLinearPCM(format)));
}
uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
@@ -2159,7 +2159,7 @@
return;
}
mRefCount[stream] += delta;
- LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+ LOGV("changeRefCount() delta %d, stream %d, refCount %d", delta, stream, mRefCount[stream]);
}
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
@@ -2215,7 +2215,8 @@
AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
: mSamplingRate(0), mFormat(0), mChannels(0),
- mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+ mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
+ mInputSource(0)
{
}