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authorEric Laurent <elaurent@google.com>2009-07-17 12:17:14 -0700
committerEric Laurent <elaurent@google.com>2009-07-23 06:03:39 -0700
commita553c25b33c99b345cf1c8688f8df0ed8df14e5a (patch)
tree025c461b13e66ad0ceac8d0f8d9b13fd88ae168a /libs/audioflinger
parentebd7bc54028949619bbf3fa5ed6c1188f588c230 (diff)
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Fix issue 1795088 Improve audio routing code
Initial commit for review. Integrated comments after patch set 1 review. Fixed lockup in AudioFlinger::ThreadBase::exit() Fixed lockup when playing tone with AudioPlocyService startTone()
Diffstat (limited to 'libs/audioflinger')
-rw-r--r--libs/audioflinger/A2dpAudioInterface.cpp224
-rw-r--r--libs/audioflinger/A2dpAudioInterface.h50
-rw-r--r--libs/audioflinger/Android.mk64
-rw-r--r--libs/audioflinger/AudioDumpInterface.cpp382
-rw-r--r--libs/audioflinger/AudioDumpInterface.h116
-rw-r--r--libs/audioflinger/AudioFlinger.cpp3236
-rw-r--r--libs/audioflinger/AudioFlinger.h525
-rw-r--r--libs/audioflinger/AudioHardwareGeneric.cpp161
-rw-r--r--libs/audioflinger/AudioHardwareGeneric.h49
-rw-r--r--libs/audioflinger/AudioHardwareInterface.cpp109
-rw-r--r--libs/audioflinger/AudioHardwareStub.cpp52
-rw-r--r--libs/audioflinger/AudioHardwareStub.h37
-rw-r--r--libs/audioflinger/AudioMixer.cpp1
-rw-r--r--libs/audioflinger/AudioMixer.h3
-rw-r--r--libs/audioflinger/AudioPolicyManagerGeneric.cpp764
-rw-r--r--libs/audioflinger/AudioPolicyManagerGeneric.h189
-rw-r--r--libs/audioflinger/AudioPolicyService.cpp677
-rw-r--r--libs/audioflinger/AudioPolicyService.h201
18 files changed, 5193 insertions, 1647 deletions
diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp
index 16a4f2d..6f9e934 100644
--- a/libs/audioflinger/A2dpAudioInterface.cpp
+++ b/libs/audioflinger/A2dpAudioInterface.cpp
@@ -29,25 +29,41 @@ namespace android {
// ----------------------------------------------------------------------------
-A2dpAudioInterface::A2dpAudioInterface() :
- mOutput(0)
+//AudioHardwareInterface* A2dpAudioInterface::createA2dpInterface()
+//{
+// AudioHardwareInterface* hw = 0;
+//
+// hw = AudioHardwareInterface::create();
+// LOGD("new A2dpAudioInterface(hw: %p)", hw);
+// hw = new A2dpAudioInterface(hw);
+// return hw;
+//}
+
+A2dpAudioInterface::A2dpAudioInterface(AudioHardwareInterface* hw) :
+ mOutput(0), mHardwareInterface(hw), mBluetoothEnabled(true)
{
}
A2dpAudioInterface::~A2dpAudioInterface()
{
- delete mOutput;
+ closeOutputStream((AudioStreamOut *)mOutput);
+ delete mHardwareInterface;
}
status_t A2dpAudioInterface::initCheck()
{
- return 0;
+ if (mHardwareInterface == 0) return NO_INIT;
+ return mHardwareInterface->initCheck();
}
AudioStreamOut* A2dpAudioInterface::openOutputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status)
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
{
- LOGD("A2dpAudioInterface::openOutputStream %d, %d, %d\n", format, channelCount, sampleRate);
+ if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
+ LOGV("A2dpAudioInterface::openOutputStream() open HW device: %x", devices);
+ return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
+ }
+
status_t err = 0;
// only one output stream allowed
@@ -59,71 +75,127 @@ AudioStreamOut* A2dpAudioInterface::openOutputStream(
// create new output stream
A2dpAudioStreamOut* out = new A2dpAudioStreamOut();
- if ((err = out->set(format, channelCount, sampleRate)) == NO_ERROR) {
+ if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
mOutput = out;
+ mOutput->setBluetoothEnabled(mBluetoothEnabled);
} else {
delete out;
}
-
+
if (status)
*status = err;
return mOutput;
}
+void A2dpAudioInterface::closeOutputStream(AudioStreamOut* out) {
+ if (mOutput == 0 || mOutput != out) {
+ LOGW("Attempt to close invalid output stream");
+ }
+ else {
+ delete mOutput;
+ mOutput = 0;
+ }
+}
+
+
AudioStreamIn* A2dpAudioInterface::openInputStream(
- int inputSource, int format, int channelCount, uint32_t sampleRate,
- status_t *status, AudioSystem::audio_in_acoustics acoustics)
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
+ AudioSystem::audio_in_acoustics acoustics)
{
- if (status)
- *status = -1;
- return NULL;
+ return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
+}
+
+void A2dpAudioInterface::closeInputStream(AudioStreamIn* in)
+{
+ return mHardwareInterface->closeInputStream(in);
+}
+
+status_t A2dpAudioInterface::setMode(int mode)
+{
+ return mHardwareInterface->setMode(mode);
}
status_t A2dpAudioInterface::setMicMute(bool state)
{
- return 0;
+ return mHardwareInterface->setMicMute(state);
}
status_t A2dpAudioInterface::getMicMute(bool* state)
{
- return 0;
+ return mHardwareInterface->getMicMute(state);
}
-status_t A2dpAudioInterface::setParameter(const char *key, const char *value)
+status_t A2dpAudioInterface::setParameters(const String8& keyValuePairs)
{
- LOGD("setParameter %s,%s\n", key, value);
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 value;
+ String8 key;
+ status_t status = NO_ERROR;
+
+ LOGV("setParameters() %s", keyValuePairs.string());
+
+ key = "bluetooth_enabled";
+ if (param.get(key, value) == NO_ERROR) {
+ mBluetoothEnabled = (value == "true");
+ if (mOutput) {
+ mOutput->setBluetoothEnabled(mBluetoothEnabled);
+ }
+ param.remove(key);
+ }
- if (!key || !value)
- return -EINVAL;
+ if (param.size()) {
+ status_t hwStatus = mHardwareInterface->setParameters(param.toString());
+ if (status == NO_ERROR) {
+ status = hwStatus;
+ }
+ }
- if (strcmp(key, "a2dp_sink_address") == 0) {
- return mOutput->setAddress(value);
+ return status;
+}
+
+String8 A2dpAudioInterface::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ AudioParameter a2dpParam = AudioParameter();
+ String8 value;
+ String8 key;
+
+ key = "bluetooth_enabled";
+ if (param.get(key, value) == NO_ERROR) {
+ value = mBluetoothEnabled ? "true" : "false";
+ a2dpParam.add(key, value);
+ param.remove(key);
}
- if (strcmp(key, "bluetooth_enabled") == 0) {
- mOutput->setBluetoothEnabled(strcmp(value, "true") == 0);
+
+ String8 keyValuePairs = a2dpParam.toString();
+
+ if (param.size()) {
+ keyValuePairs += ";";
+ keyValuePairs += mHardwareInterface->getParameters(param.toString());
}
- return 0;
+ LOGV("getParameters() %s", keyValuePairs.string());
+ return keyValuePairs;
}
-status_t A2dpAudioInterface::setVoiceVolume(float v)
+size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
{
- return 0;
+ return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
}
-status_t A2dpAudioInterface::setMasterVolume(float v)
+status_t A2dpAudioInterface::setVoiceVolume(float v)
{
- return 0;
+ return mHardwareInterface->setVoiceVolume(v);
}
-status_t A2dpAudioInterface::doRouting()
+status_t A2dpAudioInterface::setMasterVolume(float v)
{
- return 0;
+ return mHardwareInterface->setMasterVolume(v);
}
status_t A2dpAudioInterface::dump(int fd, const Vector<String16>& args)
{
- return 0;
+ return mHardwareInterface->dumpState(fd, args);
}
// ----------------------------------------------------------------------------
@@ -132,7 +204,7 @@ A2dpAudioInterface::A2dpAudioStreamOut::A2dpAudioStreamOut() :
mFd(-1), mStandby(true), mStartCount(0), mRetryCount(0), mData(NULL),
// assume BT enabled to start, this is safe because its only the
// enabled->disabled transition we are worried about
- mBluetoothEnabled(true)
+ mBluetoothEnabled(true), mDevice(0)
{
// use any address by default
strcpy(mA2dpAddress, "00:00:00:00:00:00");
@@ -140,27 +212,43 @@ A2dpAudioInterface::A2dpAudioStreamOut::A2dpAudioStreamOut() :
}
status_t A2dpAudioInterface::A2dpAudioStreamOut::set(
- int format, int channels, uint32_t rate)
+ uint32_t device, int *pFormat, uint32_t *pChannels, uint32_t *pRate)
{
- LOGD("A2dpAudioStreamOut::set %d, %d, %d\n", format, channels, rate);
+ int lFormat = pFormat ? *pFormat : 0;
+ uint32_t lChannels = pChannels ? *pChannels : 0;
+ uint32_t lRate = pRate ? *pRate : 0;
+
+ LOGD("A2dpAudioStreamOut::set %x, %d, %d, %d\n", device, lFormat, lChannels, lRate);
// fix up defaults
- if (format == 0) format = AudioSystem::PCM_16_BIT;
- if (channels == 0) channels = channelCount();
- if (rate == 0) rate = sampleRate();
+ if (lFormat == 0) lFormat = format();
+ if (lChannels == 0) lChannels = channels();
+ if (lRate == 0) lRate = sampleRate();
// check values
- if ((format != AudioSystem::PCM_16_BIT) ||
- (channels != channelCount()) ||
- (rate != sampleRate()))
+ if ((lFormat != format()) ||
+ (lChannels != channels()) ||
+ (lRate != sampleRate())){
+ if (pFormat) *pFormat = format();
+ if (pChannels) *pChannels = channels();
+ if (pRate) *pRate = sampleRate();
return BAD_VALUE;
+ }
+ if (pFormat) *pFormat = lFormat;
+ if (pChannels) *pChannels = lChannels;
+ if (pRate) *pRate = lRate;
+
+ mDevice = device;
return NO_ERROR;
}
A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut()
{
+ LOGV("A2dpAudioStreamOut destructor");
+ standby();
close();
+ LOGV("A2dpAudioStreamOut destructor returning from close()");
}
ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes)
@@ -230,6 +318,59 @@ status_t A2dpAudioInterface::A2dpAudioStreamOut::standby()
return result;
}
+status_t A2dpAudioInterface::A2dpAudioStreamOut::setParameters(const String8& keyValuePairs)
+{
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 value;
+ String8 key = String8("a2dp_sink_address");
+ status_t status = NO_ERROR;
+ int device;
+ LOGV("A2dpAudioStreamOut::setParameters() %s", keyValuePairs.string());
+
+ if (param.get(key, value) == NO_ERROR) {
+ if (value.length() != strlen("00:00:00:00:00:00")) {
+ status = BAD_VALUE;
+ } else {
+ setAddress(value.string());
+ }
+ param.remove(key);
+ }
+ key = AudioParameter::keyRouting;
+ if (param.getInt(key, device) == NO_ERROR) {
+ if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device)) {
+ mDevice = device;
+ status = NO_ERROR;
+ } else {
+ status = BAD_VALUE;
+ }
+ param.remove(key);
+ }
+
+ if (param.size()) {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
+String8 A2dpAudioInterface::A2dpAudioStreamOut::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ String8 value;
+ String8 key = String8("a2dp_sink_address");
+
+ if (param.get(key, value) == NO_ERROR) {
+ value = mA2dpAddress;
+ param.add(key, value);
+ }
+ key = AudioParameter::keyRouting;
+ if (param.get(key, value) == NO_ERROR) {
+ param.addInt(key, (int)mDevice);
+ }
+
+ LOGV("A2dpAudioStreamOut::getParameters() %s", param.toString().string());
+ return param.toString();
+}
+
status_t A2dpAudioInterface::A2dpAudioStreamOut::setAddress(const char* address)
{
Mutex::Autolock lock(mLock);
@@ -260,12 +401,14 @@ status_t A2dpAudioInterface::A2dpAudioStreamOut::setBluetoothEnabled(bool enable
status_t A2dpAudioInterface::A2dpAudioStreamOut::close()
{
Mutex::Autolock lock(mLock);
+ LOGV("A2dpAudioStreamOut::close() calling close_l()");
return close_l();
}
status_t A2dpAudioInterface::A2dpAudioStreamOut::close_l()
{
if (mData) {
+ LOGV("A2dpAudioStreamOut::close_l() calling a2dp_cleanup(mData)");
a2dp_cleanup(mData);
mData = NULL;
}
@@ -277,5 +420,4 @@ status_t A2dpAudioInterface::A2dpAudioStreamOut::dump(int fd, const Vector<Strin
return NO_ERROR;
}
-
}; // namespace android
diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h
index 091e775..d6709e2 100644
--- a/libs/audioflinger/A2dpAudioInterface.h
+++ b/libs/audioflinger/A2dpAudioInterface.h
@@ -32,38 +32,44 @@ class A2dpAudioInterface : public AudioHardwareBase
class A2dpAudioStreamOut;
public:
- A2dpAudioInterface();
+ A2dpAudioInterface(AudioHardwareInterface* hw);
virtual ~A2dpAudioInterface();
virtual status_t initCheck();
virtual status_t setVoiceVolume(float volume);
virtual status_t setMasterVolume(float volume);
+ virtual status_t setMode(int mode);
+
// mic mute
virtual status_t setMicMute(bool state);
virtual status_t getMicMute(bool* state);
- // Temporary interface, do not use
- // TODO: Replace with a more generic key:value get/set mechanism
- virtual status_t setParameter(const char *key, const char *value);
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
+
+ virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
// create I/O streams
virtual AudioStreamOut* openOutputStream(
- int format=0,
- int channelCount=0,
- uint32_t sampleRate=0,
+ uint32_t devices,
+ int *format=0,
+ uint32_t *channels=0,
+ uint32_t *sampleRate=0,
status_t *status=0);
+ virtual void closeOutputStream(AudioStreamOut* out);
virtual AudioStreamIn* openInputStream(
- int inputSource,
- int format,
- int channelCount,
- uint32_t sampleRate,
+ uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
status_t *status,
AudioSystem::audio_in_acoustics acoustics);
+ virtual void closeInputStream(AudioStreamIn* in);
+// static AudioHardwareInterface* createA2dpInterface();
protected:
- virtual status_t doRouting();
virtual status_t dump(int fd, const Vector<String16>& args);
private:
@@ -71,19 +77,22 @@ private:
public:
A2dpAudioStreamOut();
virtual ~A2dpAudioStreamOut();
- status_t set(int format,
- int channelCount,
- uint32_t sampleRate);
+ status_t set(uint32_t device,
+ int *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pRate);
virtual uint32_t sampleRate() const { return 44100; }
// SBC codec wants a multiple of 512
virtual size_t bufferSize() const { return 512 * 20; }
- virtual int channelCount() const { return 2; }
+ virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
virtual int format() const { return AudioSystem::PCM_16_BIT; }
virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
- virtual status_t setVolume(float volume) { return INVALID_OPERATION; }
+ virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; }
virtual ssize_t write(const void* buffer, size_t bytes);
status_t standby();
virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
private:
friend class A2dpAudioInterface;
@@ -102,11 +111,18 @@ private:
void* mData;
Mutex mLock;
bool mBluetoothEnabled;
+ uint32_t mDevice;
};
+ friend class A2dpAudioStreamOut;
+
A2dpAudioStreamOut* mOutput;
+ AudioHardwareInterface *mHardwareInterface;
+ char mA2dpAddress[20];
+ bool mBluetoothEnabled;
};
+
// ----------------------------------------------------------------------------
}; // namespace android
diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk
index bb224be..7ed6a5f 100644
--- a/libs/audioflinger/Android.mk
+++ b/libs/audioflinger/Android.mk
@@ -1,13 +1,26 @@
LOCAL_PATH:= $(call my-dir)
+#AUDIO_POLICY_TEST := true
+#ENABLE_AUDIO_DUMP := true
+
include $(CLEAR_VARS)
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+ ENABLE_AUDIO_DUMP := true
+endif
+
+
LOCAL_SRC_FILES:= \
AudioHardwareGeneric.cpp \
AudioHardwareStub.cpp \
- AudioDumpInterface.cpp \
AudioHardwareInterface.cpp
+ifeq ($(ENABLE_AUDIO_DUMP),true)
+ LOCAL_SRC_FILES += AudioDumpInterface.cpp
+ LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP
+endif
+
LOCAL_SHARED_LIBRARIES := \
libcutils \
libutils \
@@ -21,8 +34,40 @@ endif
LOCAL_MODULE:= libaudiointerface
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+ LOCAL_SRC_FILES += A2dpAudioInterface.cpp
+ LOCAL_SHARED_LIBRARIES += liba2dp
+ LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
+ LOCAL_C_INCLUDES += $(call include-path-for, bluez)
+endif
+
include $(BUILD_STATIC_LIBRARY)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ AudioPolicyManagerGeneric.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ libmedia
+
+LOCAL_MODULE:= libaudiopolicygeneric
+
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+ LOCAL_CFLAGS += -DWITH_A2DP
+endif
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+ LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
+endif
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
+
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
@@ -30,7 +75,8 @@ LOCAL_SRC_FILES:= \
AudioMixer.cpp.arm \
AudioResampler.cpp.arm \
AudioResamplerSinc.cpp.arm \
- AudioResamplerCubic.cpp.arm
+ AudioResamplerCubic.cpp.arm \
+ AudioPolicyService.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -41,17 +87,25 @@ LOCAL_SHARED_LIBRARIES := \
ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
LOCAL_STATIC_LIBRARIES += libaudiointerface
+ LOCAL_CFLAGS += -DGENERIC_AUDIO
else
LOCAL_SHARED_LIBRARIES += libaudio
endif
+ifeq ($(TARGET_SIMULATOR),true)
+ LOCAL_LDLIBS += -ldl
+else
+ LOCAL_SHARED_LIBRARIES += libdl
+endif
+
LOCAL_MODULE:= libaudioflinger
ifeq ($(BOARD_HAVE_BLUETOOTH),true)
- LOCAL_SRC_FILES += A2dpAudioInterface.cpp
- LOCAL_SHARED_LIBRARIES += liba2dp
LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
- LOCAL_C_INCLUDES += $(call include-path-for, bluez)
+endif
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+ LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
endif
ifeq ($(TARGET_SIMULATOR),true)
diff --git a/libs/audioflinger/AudioDumpInterface.cpp b/libs/audioflinger/AudioDumpInterface.cpp
index b4940cb..87bb014 100644
--- a/libs/audioflinger/AudioDumpInterface.cpp
+++ b/libs/audioflinger/AudioDumpInterface.cpp
@@ -16,6 +16,7 @@
*/
#define LOG_TAG "AudioFlingerDump"
+//#define LOG_NDEBUG 0
#include <stdint.h>
#include <sys/types.h>
@@ -28,68 +29,209 @@
namespace android {
-bool gFirst = true; // true if first write after a standby
-
// ----------------------------------------------------------------------------
AudioDumpInterface::AudioDumpInterface(AudioHardwareInterface* hw)
+ : mFirstHwOutput(true), mPolicyCommands(String8("")), mFileName(String8(""))
{
if(hw == 0) {
LOGE("Dump construct hw = 0");
}
mFinalInterface = hw;
- mStreamOut = 0;
+ LOGV("Constructor %p, mFinalInterface %p", this, mFinalInterface);
}
AudioDumpInterface::~AudioDumpInterface()
{
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ closeOutputStream((AudioStreamOut *)mOutputs[i]);
+ }
if(mFinalInterface) delete mFinalInterface;
- if(mStreamOut) delete mStreamOut;
}
AudioStreamOut* AudioDumpInterface::openOutputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status)
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
+{
+ AudioStreamOut* outFinal = NULL;
+ int lFormat = AudioSystem::PCM_16_BIT;
+ uint32_t lChannels = AudioSystem::CHANNEL_OUT_STEREO;
+ uint32_t lRate = 44100;
+
+
+ if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices) || mFirstHwOutput) {
+ outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status);
+ if (outFinal != 0) {
+ lFormat = outFinal->format();
+ lChannels = outFinal->channels();
+ lRate = outFinal->sampleRate();
+ if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
+ mFirstHwOutput = false;
+ }
+ }
+ } else {
+ if (format != 0 && *format != 0) lFormat = *format;
+ if (channels != 0 && *channels != 0) lChannels = *channels;
+ if (sampleRate != 0 && *sampleRate != 0) lRate = *sampleRate;
+ if (status) *status = NO_ERROR;
+ }
+ LOGV("openOutputStream(), outFinal %p", outFinal);
+
+ AudioStreamOutDump *dumOutput = new AudioStreamOutDump(this, mOutputs.size(), outFinal,
+ devices, lFormat, lChannels, lRate);
+ mOutputs.add(dumOutput);
+
+ return dumOutput;
+}
+
+void AudioDumpInterface::closeOutputStream(AudioStreamOut* out)
+{
+ AudioStreamOutDump *dumpOut = (AudioStreamOutDump *)out;
+
+ if (mOutputs.indexOf(dumpOut) < 0) {
+ LOGW("Attempt to close invalid output stream");
+ return;
+ }
+ dumpOut->standby();
+ if (dumpOut->finalStream() != NULL) {
+ mFinalInterface->closeOutputStream(dumpOut->finalStream());
+ }
+
+ mOutputs.remove(dumpOut);
+ delete dumpOut;
+}
+
+AudioStreamIn* AudioDumpInterface::openInputStream(uint32_t devices, int *format, uint32_t *channels,
+ uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
- AudioStreamOut* outFinal = mFinalInterface->openOutputStream(format, channelCount, sampleRate, status);
+ AudioStreamIn* inFinal = NULL;
+ int lFormat = AudioSystem::PCM_16_BIT;
+ uint32_t lChannels = AudioSystem::CHANNEL_IN_MONO;
+ uint32_t lRate = 8000;
+
- if(outFinal) {
- mStreamOut = new AudioStreamOutDump(outFinal);
- return mStreamOut;
+ if (mInputs.size() == 0) {
+ inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
+ if (inFinal == 0) return 0;
+
+ lFormat = inFinal->format();
+ lChannels = inFinal->channels();
+ lRate = inFinal->sampleRate();
} else {
- LOGE("Dump outFinal=0");
- return 0;
+ if (format != 0 && *format != 0) lFormat = *format;
+ if (channels != 0 && *channels != 0) lChannels = *channels;
+ if (sampleRate != 0 && *sampleRate != 0) lRate = *sampleRate;
+ if (status) *status = NO_ERROR;
+ }
+ LOGV("openInputStream(), inFinal %p", inFinal);
+
+ AudioStreamInDump *dumInput = new AudioStreamInDump(this, mInputs.size(), inFinal,
+ devices, lFormat, lChannels, lRate);
+ mInputs.add(dumInput);
+
+ return dumInput;
+}
+void AudioDumpInterface::closeInputStream(AudioStreamIn* in)
+{
+ AudioStreamInDump *dumpIn = (AudioStreamInDump *)in;
+
+ if (mInputs.indexOf(dumpIn) < 0) {
+ LOGW("Attempt to close invalid input stream");
+ return;
+ }
+ dumpIn->standby();
+ if (dumpIn->finalStream() != NULL) {
+ mFinalInterface->closeInputStream(dumpIn->finalStream());
}
+
+ mInputs.remove(dumpIn);
+ delete dumpIn;
}
+
+status_t AudioDumpInterface::setParameters(const String8& keyValuePairs)
+{
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 value;
+ int valueInt;
+ LOGV("setParameters %s", keyValuePairs.string());
+
+ if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
+ mFileName = value;
+ return NO_ERROR;
+ }
+ if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
+ Mutex::Autolock _l(mLock);
+ param.remove(String8("test_cmd_policy"));
+ mPolicyCommands = param.toString();
+ LOGV("test_cmd_policy command %s written", mPolicyCommands.string());
+ return NO_ERROR;
+ }
+
+ if (mFinalInterface != 0 ) return mFinalInterface->setParameters(keyValuePairs);
+ return NO_ERROR;
+}
+
+String8 AudioDumpInterface::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ String8 value;
+
+// LOGV("getParameters %s", keys.string());
+
+ if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
+ return mFileName;
+ }
+ if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
+ Mutex::Autolock _l(mLock);
+// LOGV("test_cmd_policy command %s read", mPolicyCommands.string());
+ return mPolicyCommands;
+ }
+
+ if (mFinalInterface != 0 ) return mFinalInterface->getParameters(keys);
+ return String8("");
+}
+
+
// ----------------------------------------------------------------------------
-AudioStreamOutDump::AudioStreamOutDump( AudioStreamOut* finalStream)
+AudioStreamOutDump::AudioStreamOutDump(AudioDumpInterface *interface,
+ int id,
+ AudioStreamOut* finalStream,
+ uint32_t devices,
+ int format,
+ uint32_t channels,
+ uint32_t sampleRate)
+ : mInterface(interface), mId(id),
+ mSampleRate(sampleRate), mFormat(format), mChannels(channels), mLatency(0), mDevice(devices),
+ mBufferSize(1024), mFinalStream(finalStream), mOutFile(0), mFileCount(0)
{
- mFinalStream = finalStream;
- mOutFile = 0;
+ LOGV("AudioStreamOutDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
}
AudioStreamOutDump::~AudioStreamOutDump()
{
Close();
- delete mFinalStream;
}
ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes)
{
ssize_t ret;
- ret = mFinalStream->write(buffer, bytes);
- if(!mOutFile && gFirst) {
- gFirst = false;
- // check if dump file exist
- mOutFile = fopen(FLINGER_DUMP_NAME, "r");
- if(mOutFile) {
- fclose(mOutFile);
- mOutFile = fopen(FLINGER_DUMP_NAME, "ab");
+ if (mFinalStream) {
+ ret = mFinalStream->write(buffer, bytes);
+ } else {
+ usleep((bytes * 1000000) / frameSize() / sampleRate());
+ ret = bytes;
+ }
+ if(!mOutFile) {
+ if (mInterface->fileName() != "") {
+ char name[255];
+ sprintf(name, "%s_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
+ mOutFile = fopen(name, "wb");
+ LOGV("Opening dump file %s, fh %p", name, mOutFile);
}
}
if (mOutFile) {
@@ -100,13 +242,65 @@ ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes)
status_t AudioStreamOutDump::standby()
{
+ LOGV("AudioStreamOutDump standby(), mOutFile %p, mFinalStream %p", mOutFile, mFinalStream);
+
Close();
- gFirst = true;
- return mFinalStream->standby();
+ if (mFinalStream != 0 ) return mFinalStream->standby();
+ return NO_ERROR;
+}
+
+uint32_t AudioStreamOutDump::sampleRate() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->sampleRate();
+ return mSampleRate;
+}
+
+size_t AudioStreamOutDump::bufferSize() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->bufferSize();
+ return mBufferSize;
+}
+
+uint32_t AudioStreamOutDump::channels() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->channels();
+ return mChannels;
+}
+int AudioStreamOutDump::format() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->format();
+ return mFormat;
+}
+uint32_t AudioStreamOutDump::latency() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->latency();
+ return 0;
+}
+status_t AudioStreamOutDump::setVolume(float left, float right)
+{
+ if (mFinalStream != 0 ) return mFinalStream->setVolume(left, right);
+ return NO_ERROR;
+}
+status_t AudioStreamOutDump::setParameters(const String8& keyValuePairs)
+{
+ LOGV("AudioStreamOutDump::setParameters()");
+ if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
+ return NO_ERROR;
+}
+String8 AudioStreamOutDump::getParameters(const String8& keys)
+{
+ String8 result = String8("");
+ if (mFinalStream != 0 ) result = mFinalStream->getParameters(keys);
+ return result;
}
+status_t AudioStreamOutDump::dump(int fd, const Vector<String16>& args)
+{
+ if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
+ return NO_ERROR;
+}
-void AudioStreamOutDump::Close(void)
+void AudioStreamOutDump::Close()
{
if(mOutFile) {
fclose(mOutFile);
@@ -114,4 +308,140 @@ void AudioStreamOutDump::Close(void)
}
}
+// ----------------------------------------------------------------------------
+
+AudioStreamInDump::AudioStreamInDump(AudioDumpInterface *interface,
+ int id,
+ AudioStreamIn* finalStream,
+ uint32_t devices,
+ int format,
+ uint32_t channels,
+ uint32_t sampleRate)
+ : mInterface(interface), mId(id),
+ mSampleRate(sampleRate), mFormat(format), mChannels(channels), mDevice(devices),
+ mBufferSize(1024), mFinalStream(finalStream), mInFile(0)
+{
+ LOGV("AudioStreamInDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
+}
+
+
+AudioStreamInDump::~AudioStreamInDump()
+{
+ Close();
+}
+
+ssize_t AudioStreamInDump::read(void* buffer, ssize_t bytes)
+{
+ if (mFinalStream) {
+ return mFinalStream->read(buffer, bytes);
+ }
+
+ usleep((bytes * 1000000) / frameSize() / sampleRate());
+
+ if(!mInFile) {
+ char name[255];
+ strcpy(name, "/sdcard/music/sine440");
+ if (channels() == AudioSystem::CHANNEL_IN_MONO) {
+ strcat(name, "_mo");
+ } else {
+ strcat(name, "_st");
+ }
+ if (format() == AudioSystem::PCM_16_BIT) {
+ strcat(name, "_16b");
+ } else {
+ strcat(name, "_8b");
+ }
+ if (sampleRate() < 16000) {
+ strcat(name, "_8k");
+ } else if (sampleRate() < 32000) {
+ strcat(name, "_22k");
+ } else if (sampleRate() < 48000) {
+ strcat(name, "_44k");
+ } else {
+ strcat(name, "_48k");
+ }
+ strcat(name, ".wav");
+ mInFile = fopen(name, "rb");
+ LOGV("Opening dump file %s, fh %p", name, mInFile);
+ if (mInFile) {
+ fseek(mInFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
+ }
+
+ }
+ if (mInFile) {
+ ssize_t bytesRead = fread(buffer, bytes, 1, mInFile);
+ if (bytesRead != bytes) {
+ fseek(mInFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
+ fread((uint8_t *)buffer+bytesRead, bytes-bytesRead, 1, mInFile);
+ }
+ }
+ return bytes;
+}
+
+status_t AudioStreamInDump::standby()
+{
+ LOGV("AudioStreamInDump standby(), mInFile %p, mFinalStream %p", mInFile, mFinalStream);
+
+ Close();
+ if (mFinalStream != 0 ) return mFinalStream->standby();
+ return NO_ERROR;
+}
+
+status_t AudioStreamInDump::setGain(float gain)
+{
+ if (mFinalStream != 0 ) return mFinalStream->setGain(gain);
+ return NO_ERROR;
+}
+
+uint32_t AudioStreamInDump::sampleRate() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->sampleRate();
+ return mSampleRate;
+}
+
+size_t AudioStreamInDump::bufferSize() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->bufferSize();
+ return mBufferSize;
+}
+
+uint32_t AudioStreamInDump::channels() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->channels();
+ return mChannels;
+}
+
+int AudioStreamInDump::format() const
+{
+ if (mFinalStream != 0 ) return mFinalStream->format();
+ return mFormat;
+}
+
+status_t AudioStreamInDump::setParameters(const String8& keyValuePairs)
+{
+ LOGV("AudioStreamInDump::setParameters()");
+ if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
+ return NO_ERROR;
+}
+
+String8 AudioStreamInDump::getParameters(const String8& keys)
+{
+ String8 result = String8("");
+ if (mFinalStream != 0 ) result = mFinalStream->getParameters(keys);
+ return result;
+}
+
+status_t AudioStreamInDump::dump(int fd, const Vector<String16>& args)
+{
+ if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
+ return NO_ERROR;
+}
+
+void AudioStreamInDump::Close()
+{
+ if(mInFile) {
+ fclose(mInFile);
+ mInFile = 0;
+ }
+}
}; // namespace android
diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h
index b72c94e..4de4a16 100644
--- a/libs/audioflinger/AudioDumpInterface.h
+++ b/libs/audioflinger/AudioDumpInterface.h
@@ -20,35 +20,94 @@
#include <stdint.h>
#include <sys/types.h>
+#include <utils/String8.h>
+#include <utils/SortedVector.h>
#include <hardware_legacy/AudioHardwareBase.h>
namespace android {
-#define FLINGER_DUMP_NAME "/data/FlingerOut.pcm" // name of file used for dump
+#define AUDIO_DUMP_WAVE_HDR_SIZE 44
+
+class AudioDumpInterface;
class AudioStreamOutDump : public AudioStreamOut {
public:
- AudioStreamOutDump( AudioStreamOut* FinalStream);
+ AudioStreamOutDump(AudioDumpInterface *interface,
+ int id,
+ AudioStreamOut* finalStream,
+ uint32_t devices,
+ int format,
+ uint32_t channels,
+ uint32_t sampleRate);
~AudioStreamOutDump();
- virtual ssize_t write(const void* buffer, size_t bytes);
-
- virtual uint32_t sampleRate() const { return mFinalStream->sampleRate(); }
- virtual size_t bufferSize() const { return mFinalStream->bufferSize(); }
- virtual int channelCount() const { return mFinalStream->channelCount(); }
- virtual int format() const { return mFinalStream->format(); }
- virtual uint32_t latency() const { return mFinalStream->latency(); }
- virtual status_t setVolume(float volume)
- { return mFinalStream->setVolume(volume); }
+
+ virtual ssize_t write(const void* buffer, size_t bytes);
+ virtual uint32_t sampleRate() const;
+ virtual size_t bufferSize() const;
+ virtual uint32_t channels() const;
+ virtual int format() const;
+ virtual uint32_t latency() const;
+ virtual status_t setVolume(float left, float right);
virtual status_t standby();
- virtual status_t dump(int fd, const Vector<String16>& args) { return mFinalStream->dump(fd, args); }
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
+ virtual status_t dump(int fd, const Vector<String16>& args);
void Close(void);
+ AudioStreamOut* finalStream() { return mFinalStream; }
+ uint32_t device() { return mDevice; }
private:
+ AudioDumpInterface *mInterface;
+ int mId;
+ uint32_t mSampleRate; //
+ uint32_t mFormat; //
+ uint32_t mChannels; // output configuration
+ uint32_t mLatency; //
+ uint32_t mDevice; // current device this output is routed to
+ size_t mBufferSize;
AudioStreamOut *mFinalStream;
- FILE *mOutFile; // output file
+ FILE *mOutFile; // output file
+ int mFileCount;
};
+class AudioStreamInDump : public AudioStreamIn {
+public:
+ AudioStreamInDump(AudioDumpInterface *interface,
+ int id,
+ AudioStreamIn* finalStream,
+ uint32_t devices,
+ int format,
+ uint32_t channels,
+ uint32_t sampleRate);
+ ~AudioStreamInDump();
+
+ virtual uint32_t sampleRate() const;
+ virtual size_t bufferSize() const;
+ virtual uint32_t channels() const;
+ virtual int format() const;
+
+ virtual status_t setGain(float gain);
+ virtual ssize_t read(void* buffer, ssize_t bytes);
+ virtual status_t standby();
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
+ virtual status_t dump(int fd, const Vector<String16>& args);
+ void Close(void);
+ AudioStreamIn* finalStream() { return mFinalStream; }
+ uint32_t device() { return mDevice; }
+
+private:
+ AudioDumpInterface *mInterface;
+ int mId;
+ uint32_t mSampleRate; //
+ uint32_t mFormat; //
+ uint32_t mChannels; // output configuration
+ uint32_t mDevice; // current device this output is routed to
+ size_t mBufferSize;
+ AudioStreamIn *mFinalStream;
+ FILE *mInFile; // output file
+};
class AudioDumpInterface : public AudioHardwareBase
{
@@ -56,10 +115,13 @@ class AudioDumpInterface : public AudioHardwareBase
public:
AudioDumpInterface(AudioHardwareInterface* hw);
virtual AudioStreamOut* openOutputStream(
- int format=0,
- int channelCount=0,
- uint32_t sampleRate=0,
+ uint32_t devices,
+ int *format=0,
+ uint32_t *channels=0,
+ uint32_t *sampleRate=0,
status_t *status=0);
+ virtual void closeOutputStream(AudioStreamOut* out);
+
virtual ~AudioDumpInterface();
virtual status_t initCheck()
@@ -75,21 +137,25 @@ public:
virtual status_t getMicMute(bool* state)
{return mFinalInterface->getMicMute(state);}
- virtual status_t setParameter(const char* key, const char* value)
- {return mFinalInterface->setParameter(key, value);}
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
- virtual AudioStreamIn* openInputStream(int inputSource, int format, int channelCount,
- uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
- { return mFinalInterface->openInputStream(inputSource, format, channelCount, sampleRate, status, acoustics); }
+ virtual AudioStreamIn* openInputStream(uint32_t devices, int *format, uint32_t *channels,
+ uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
+ virtual void closeInputStream(AudioStreamIn* in);
virtual status_t dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
+ String8 fileName() const { return mFileName; }
protected:
- virtual status_t doRouting() {return mFinalInterface->setRouting(mMode, mRoutes[mMode]);}
-
- AudioHardwareInterface *mFinalInterface;
- AudioStreamOutDump *mStreamOut;
+ AudioHardwareInterface *mFinalInterface;
+ SortedVector<AudioStreamOutDump *> mOutputs;
+ bool mFirstHwOutput;
+ SortedVector<AudioStreamInDump *> mInputs;
+ Mutex mLock;
+ String8 mPolicyCommands;
+ String8 mFileName;
};
}; // namespace android
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index ffc0278..c05ab77 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -71,15 +71,9 @@ static const float MAX_GAIN = 4096.0f;
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
-static const int kStartSleepTime = 30000;
-static const int kStopSleepTime = 30000;
-
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
-// Maximum number of pending buffers allocated by OutputTrack::write()
-static const uint8_t kMaxOutputTrackBuffers = 5;
-
#define AUDIOFLINGER_SECURITY_ENABLED 1
@@ -121,132 +115,32 @@ static bool settingsAllowed() {
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false),
- mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0),
- mRouteRestoreTime(0), mMusicMuteSaved(false)
+ mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false)
{
mHardwareStatus = AUDIO_HW_IDLE;
+
mAudioHardware = AudioHardwareInterface::create();
+
mHardwareStatus = AUDIO_HW_INIT;
if (mAudioHardware->initCheck() == NO_ERROR) {
// open 16-bit output stream for s/w mixer
- mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
- status_t status;
- AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
- mHardwareStatus = AUDIO_HW_IDLE;
- if (hwOutput) {
- mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
- } else {
- LOGE("Failed to initialize hardware output stream, status: %d", status);
- }
-
-#ifdef WITH_A2DP
- // Create A2DP interface
- mA2dpAudioInterface = new A2dpAudioInterface();
- AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
- if (a2dpOutput) {
- mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
- if (hwOutput) {
- uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
- MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
- hwOutput->sampleRate(),
- AudioSystem::PCM_16_BIT,
- hwOutput->channelCount(),
- frameCount);
- mHardwareMixerThread->setOuputTrack(a2dpOutTrack);
- }
- } else {
- LOGE("Failed to initialize A2DP output stream, status: %d", status);
- }
-#endif
-
- // FIXME - this should come from settings
- setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
- setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
- setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+
setMode(AudioSystem::MODE_NORMAL);
setMasterVolume(1.0f);
setMasterMute(false);
-
- // Start record thread
- mAudioRecordThread = new AudioRecordThread(mAudioHardware, this);
- if (mAudioRecordThread != 0) {
- mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
- }
- } else {
+ } else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
}
AudioFlinger::~AudioFlinger()
{
- if (mAudioRecordThread != 0) {
- mAudioRecordThread->exit();
- mAudioRecordThread.clear();
- }
- mHardwareMixerThread.clear();
- delete mAudioHardware;
- // deleting mA2dpAudioInterface also deletes mA2dpOutput;
-#ifdef WITH_A2DP
- mA2dpMixerThread.clear();
- delete mA2dpAudioInterface;
-#endif
+ mRecordThreads.clear();
+ mPlaybackThreads.clear();
}
-#ifdef WITH_A2DP
-// setA2dpEnabled_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::setA2dpEnabled_l(bool enable)
-{
- SortedVector < sp<MixerThread::Track> > tracks;
- SortedVector < wp<MixerThread::Track> > activeTracks;
-
- LOGV_IF(enable, "set output to A2DP\n");
- LOGV_IF(!enable, "set output to hardware audio\n");
-
- // Transfer tracks playing on MUSIC stream from one mixer to the other
- if (enable) {
- mHardwareMixerThread->getTracks_l(tracks, activeTracks);
- mA2dpMixerThread->putTracks_l(tracks, activeTracks);
- } else {
- mA2dpMixerThread->getTracks_l(tracks, activeTracks);
- mHardwareMixerThread->putTracks_l(tracks, activeTracks);
- mA2dpMixerThread->mOutput->standby();
- }
- mA2dpEnabled = enable;
- mNotifyA2dpChange = true;
- mWaitWorkCV.broadcast();
-}
-
-// checkA2dpEnabledChange_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::checkA2dpEnabledChange_l()
-{
- if (mNotifyA2dpChange) {
- // Notify AudioSystem of the A2DP activation/deactivation
- size_t size = mNotificationClients.size();
- for (size_t i = 0; i < size; i++) {
- sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
- if (binder != NULL) {
- LOGV("Notifying output change to client %p", binder.get());
- sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
- client->a2dpEnabledChanged(mA2dpEnabled);
- }
- }
- mNotifyA2dpChange = false;
- }
-}
-#endif // WITH_A2DP
-
-bool AudioFlinger::streamForcedToSpeaker(int streamType)
-{
- // NOTE that streams listed here must not be routed to A2DP by default:
- // AudioSystem::routedToA2dpOutput(streamType) == false
- return (streamType == AudioSystem::RING ||
- streamType == AudioSystem::ALARM ||
- streamType == AudioSystem::NOTIFICATION ||
- streamType == AudioSystem::ENFORCED_AUDIBLE);
-}
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
@@ -276,10 +170,7 @@ status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
char buffer[SIZE];
String8 result;
int hardwareStatus = mHardwareStatus;
-
- if (hardwareStatus == AUDIO_HW_IDLE && mHardwareMixerThread->mStandby) {
- hardwareStatus = AUDIO_HW_STANDBY;
- }
+
snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
result.append(buffer);
write(fd, result.string(), result.size());
@@ -337,13 +228,16 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
dumpClients(fd, args);
dumpInternals(fd, args);
- mHardwareMixerThread->dump(fd, args);
-#ifdef WITH_A2DP
- mA2dpMixerThread->dump(fd, args);
-#endif
- // dump record client
- if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
+ // dump playback threads
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads[i]->dump(fd, args);
+ }
+
+ // dump record threads
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mRecordThreads[i]->dump(fd, args);
+ }
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
@@ -353,6 +247,7 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
return NO_ERROR;
}
+
// IAudioFlinger interface
@@ -365,9 +260,10 @@ sp<IAudioTrack> AudioFlinger::createTrack(
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
+ void *output,
status_t *status)
{
- sp<MixerThread::Track> track;
+ sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
@@ -381,6 +277,12 @@ sp<IAudioTrack> AudioFlinger::createTrack(
{
Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGE("unknown output thread");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
wclient = mClients.valueFor(pid);
@@ -390,16 +292,8 @@ sp<IAudioTrack> AudioFlinger::createTrack(
client = new Client(this, pid);
mClients.add(pid, client);
}
-#ifdef WITH_A2DP
- if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
- track = mA2dpMixerThread->createTrack_l(client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, &lStatus);
- } else
-#endif
- {
- track = mHardwareMixerThread->createTrack_l(client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, &lStatus);
- }
+ track = thread->createTrack_l(client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer, &lStatus);
}
if (lStatus == NO_ERROR) {
trackHandle = new TrackHandle(track);
@@ -414,54 +308,59 @@ Exit:
return trackHandle;
}
-uint32_t AudioFlinger::sampleRate(int output) const
+uint32_t AudioFlinger::sampleRate(void *output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->sampleRate();
- }
-#endif
- return mHardwareMixerThread->sampleRate();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("sampleRate() unknown thread %p", output);
+ return 0;
+ }
+ return thread->sampleRate();
}
-int AudioFlinger::channelCount(int output) const
+int AudioFlinger::channelCount(void *output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->channelCount();
- }
-#endif
- return mHardwareMixerThread->channelCount();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("channelCount() unknown thread %p", output);
+ return 0;
+ }
+ return thread->channelCount();
}
-int AudioFlinger::format(int output) const
+int AudioFlinger::format(void *output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->format();
- }
-#endif
- return mHardwareMixerThread->format();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("format() unknown thread %p", output);
+ return 0;
+ }
+ return thread->format();
}
-size_t AudioFlinger::frameCount(int output) const
+size_t AudioFlinger::frameCount(void *output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->frameCount();
- }
-#endif
- return mHardwareMixerThread->frameCount();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("frameCount() unknown thread %p", output);
+ return 0;
+ }
+ return thread->frameCount();
}
-uint32_t AudioFlinger::latency(int output) const
+uint32_t AudioFlinger::latency(void *output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->latency();
- }
-#endif
- return mHardwareMixerThread->latency();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("latency() unknown thread %p", output);
+ return 0;
+ }
+ return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
@@ -478,94 +377,12 @@ status_t AudioFlinger::setMasterVolume(float value)
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
- mHardwareMixerThread->setMasterVolume(value);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setMasterVolume(value);
-#endif
-
- return NO_ERROR;
-}
-
-status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
-{
- status_t err = NO_ERROR;
-
- // check calling permissions
- if (!settingsAllowed()) {
- return PERMISSION_DENIED;
- }
- if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
- LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
- return BAD_VALUE;
- }
-
-#ifdef WITH_A2DP
- LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
- IPCThreadState::self()->getCallingPid());
- if (mode == AudioSystem::MODE_NORMAL &&
- (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
- AutoMutex lock(&mLock);
-
- bool enableA2dp = false;
- if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
- enableA2dp = true;
- }
- if (mA2dpDisableCount > 0) {
- mA2dpSuppressed = enableA2dp;
- } else {
- setA2dpEnabled_l(enableA2dp);
- }
- LOGV("setOutput done\n");
- }
- // setRouting() is always called at least for mode == AudioSystem::MODE_IN_CALL when
- // SCO is enabled, whatever current mode is so we can safely handle A2DP disabling only
- // in this case to avoid doing it several times.
- if (mode == AudioSystem::MODE_IN_CALL &&
- (mask & AudioSystem::ROUTE_BLUETOOTH_SCO)) {
- AutoMutex lock(&mLock);
- handleRouteDisablesA2dp_l(routes);
- }
-#endif
- // do nothing if only A2DP routing is affected
- mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
- if (mask) {
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_HW_GET_ROUTING;
- uint32_t r;
- err = mAudioHardware->getRouting(mode, &r);
- if (err == NO_ERROR) {
- r = (r & ~mask) | (routes & mask);
- if (mode == AudioSystem::MODE_NORMAL ||
- (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
- mSavedRoute = r;
- r |= mForcedRoute;
- LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
- }
- mHardwareStatus = AUDIO_HW_SET_ROUTING;
- err = mAudioHardware->setRouting(mode, r);
- }
- mHardwareStatus = AUDIO_HW_IDLE;
- }
- return err;
-}
+ mMasterVolume = value;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads[i]->setMasterVolume(value);
-uint32_t AudioFlinger::getRouting(int mode) const
-{
- uint32_t routes = 0;
- if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
- if (mode == AudioSystem::MODE_NORMAL ||
- (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
- routes = mSavedRoute;
- } else {
- mHardwareStatus = AUDIO_HW_GET_ROUTING;
- mAudioHardware->getRouting(mode, &routes);
- mHardwareStatus = AUDIO_HW_IDLE;
- }
- } else {
- LOGW("Illegal value: getRouting(%d)", mode);
- }
- return routes;
+ return NO_ERROR;
}
status_t AudioFlinger::setMode(int mode)
@@ -586,15 +403,6 @@ status_t AudioFlinger::setMode(int mode)
return ret;
}
-int AudioFlinger::getMode() const
-{
- int mode = AudioSystem::MODE_INVALID;
- mHardwareStatus = AUDIO_HW_SET_MODE;
- mAudioHardware->getMode(&mode);
- mHardwareStatus = AUDIO_HW_IDLE;
- return mode;
-}
-
status_t AudioFlinger::setMicMute(bool state)
{
// check calling permissions
@@ -624,37 +432,46 @@ status_t AudioFlinger::setMasterMute(bool muted)
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- mHardwareMixerThread->setMasterMute(muted);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setMasterMute(muted);
-#endif
+
+ mMasterMute = muted;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads[i]->setMasterMute(muted);
+
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
- return mHardwareMixerThread->masterVolume();
+ return mMasterVolume;
}
bool AudioFlinger::masterMute() const
{
- return mHardwareMixerThread->masterMute();
+ return mMasterMute;
}
-status_t AudioFlinger::setStreamVolume(int stream, float value)
+status_t AudioFlinger::setStreamVolume(int stream, float value, void *output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
- uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+ if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return BAD_VALUE;
}
+ AutoMutex lock(mLock);
+ PlaybackThread *thread = NULL;
+ if (output) {
+ thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+ }
+
status_t ret = NO_ERROR;
-
+
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
float hwValue;
@@ -671,18 +488,18 @@ status_t AudioFlinger::setStreamVolume(int stream, float value)
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
ret = mAudioHardware->setVoiceVolume(hwValue);
mHardwareStatus = AUDIO_HW_IDLE;
-
+
}
-
- mHardwareMixerThread->setStreamVolume(stream, value);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setStreamVolume(stream, value);
-#endif
- mHardwareMixerThread->setStreamVolume(stream, value);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setStreamVolume(stream, value);
-#endif
+ mStreamTypes[stream].volume = value;
+
+ if (thread == NULL) {
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads[i]->setStreamVolume(stream, value);
+
+ } else {
+ thread->setStreamVolume(stream, value);
+ }
return ret;
}
@@ -694,82 +511,116 @@ status_t AudioFlinger::setStreamMute(int stream, bool muted)
return PERMISSION_DENIED;
}
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
+ if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
-#ifdef WITH_A2DP
- mA2dpMixerThread->setStreamMute(stream, muted);
-#endif
- if (stream == AudioSystem::MUSIC)
- {
- AutoMutex lock(&mHardwareLock);
- if (mForcedRoute != 0)
- mMusicMuteSaved = muted;
- else
- mHardwareMixerThread->setStreamMute(stream, muted);
- } else {
- mHardwareMixerThread->setStreamMute(stream, muted);
- }
+ mStreamTypes[stream].mute = muted;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads[i]->setStreamMute(stream, muted);
return NO_ERROR;
}
-float AudioFlinger::streamVolume(int stream) const
+float AudioFlinger::streamVolume(int stream, void *output) const
{
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return 0.0f;
}
-
- float volume = mHardwareMixerThread->streamVolume(stream);
+
+ AutoMutex lock(mLock);
+ float volume;
+ if (output) {
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return 0.0f;
+ }
+ volume = thread->streamVolume(stream);
+ } else {
+ volume = mStreamTypes[stream].volume;
+ }
+
// remove correction applied by setStreamVolume()
if (stream == AudioSystem::VOICE_CALL) {
volume = (volume - 0.01) / 0.99 ;
}
-
+
return volume;
}
bool AudioFlinger::streamMute(int stream) const
{
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
return true;
}
-
- if (stream == AudioSystem::MUSIC && mForcedRoute != 0)
- {
- return mMusicMuteSaved;
- }
- return mHardwareMixerThread->streamMute(stream);
+
+ return mStreamTypes[stream].mute;
}
bool AudioFlinger::isMusicActive() const
{
Mutex::Autolock _l(mLock);
- #ifdef WITH_A2DP
- if (isA2dpEnabled()) {
- return mA2dpMixerThread->isMusicActive_l();
- }
- #endif
- return mHardwareMixerThread->isMusicActive_l();
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads[i]->isMusicActive()) {
+ return true;
+ }
+ }
+ return false;
}
-status_t AudioFlinger::setParameter(const char* key, const char* value)
+status_t AudioFlinger::setParameters(void *ioHandle, const String8& keyValuePairs)
{
- status_t result, result2;
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_SET_PARAMETER;
-
- LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
- result = mAudioHardware->setParameter(key, value);
- if (mA2dpAudioInterface) {
- result2 = mA2dpAudioInterface->setParameter(key, value);
- if (result2)
- result = result2;
+ status_t result;
+
+ LOGV("setParameters(): io %p, keyvalue %s, tid %d, calling tid %d",
+ ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
}
- mHardwareStatus = AUDIO_HW_IDLE;
- return result;
+
+ // ioHandle == 0 means the parameters are global to the audio hardware interface
+ if (ioHandle == 0) {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_PARAMETER;
+ result = mAudioHardware->setParameters(keyValuePairs);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return result;
+ }
+
+ // Check if parameters are for an output
+ PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+ if (playbackThread != NULL) {
+ return playbackThread->setParameters(keyValuePairs);
+ }
+
+ // Check if parameters are for an input
+ RecordThread *recordThread = checkRecordThread_l(ioHandle);
+ if (recordThread != NULL) {
+ return recordThread->setParameters(keyValuePairs);
+ }
+
+ return BAD_VALUE;
+}
+
+String8 AudioFlinger::getParameters(void *ioHandle, const String8& keys)
+{
+// LOGV("getParameters() io %p, keys %s, tid %d, calling tid %d",
+// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+
+ if (ioHandle == 0) {
+ return mAudioHardware->getParameters(keys);
+ }
+ PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+ if (playbackThread != NULL) {
+ return playbackThread->getParameters(keys);
+ }
+ RecordThread *recordThread = checkRecordThread_l(ioHandle);
+ if (recordThread != NULL) {
+ return recordThread->getParameters(keys);
+ }
+ return String8("");
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
@@ -779,7 +630,7 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int cha
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
-
+
LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
@@ -788,12 +639,21 @@ void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
LOGV("Adding notification client %p", binder.get());
binder->linkToDeath(this);
mNotificationClients.add(binder);
- client->a2dpEnabledChanged(isA2dpEnabled());
+ }
+
+ // the config change is always sent from playback or record threads to avoid deadlock
+ // with AudioSystem::gLock
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads[i]->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
+ }
+
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ mRecordThreads[i]->sendConfigEvent(AudioSystem::INPUT_OPENED);
}
}
void AudioFlinger::binderDied(const wp<IBinder>& who) {
-
+
LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
@@ -808,6 +668,17 @@ void AudioFlinger::binderDied(const wp<IBinder>& who) {
}
}
+void AudioFlinger::audioConfigChanged(int event, void *param1, void *param2) {
+ Mutex::Autolock _l(mLock);
+ size_t size = mNotificationClients.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<IBinder> binder = mNotificationClients.itemAt(i);
+ LOGV("audioConfigChanged() Notifying change to client %p", binder.get());
+ sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
+ client->ioConfigChanged(event, param1, param2);
+ }
+}
+
void AudioFlinger::removeClient(pid_t pid)
{
LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
@@ -815,147 +686,140 @@ void AudioFlinger::removeClient(pid_t pid)
mClients.removeItem(pid);
}
-bool AudioFlinger::isA2dpEnabled() const
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger)
+ : Thread(false),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
+ mFormat(0), mFrameSize(1), mNewParameters(String8("")), mStandby(false)
{
- return mA2dpEnabled;
}
-void AudioFlinger::handleForcedSpeakerRoute(int command)
+AudioFlinger::ThreadBase::~ThreadBase()
{
- switch(command) {
- case ACTIVE_TRACK_ADDED:
- {
- AutoMutex lock(mHardwareLock);
- if (mForcedSpeakerCount++ == 0) {
- if (mForcedRoute == 0) {
- mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
- LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
- if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
- LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
- usleep(mHardwareMixerThread->latency()*1000);
- mHardwareStatus = AUDIO_HW_SET_ROUTING;
- mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareStatus = AUDIO_HW_IDLE;
- // delay track start so that audio hardware has time to siwtch routes
- usleep(kStartSleepTime);
- }
- }
- mForcedRoute = AudioSystem::ROUTE_SPEAKER;
- mRouteRestoreTime = 0;
- }
- LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
- }
- break;
- case ACTIVE_TRACK_REMOVED:
- {
- AutoMutex lock(mHardwareLock);
- if (mForcedSpeakerCount > 0){
- if (--mForcedSpeakerCount == 0) {
- mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
- }
- LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);
- } else {
- LOGE("mForcedSpeakerCount is already zero");
- }
- }
- break;
- case CHECK_ROUTE_RESTORE_TIME:
- case FORCE_ROUTE_RESTORE:
- if (mRouteRestoreTime) {
- AutoMutex lock(mHardwareLock);
- if (mRouteRestoreTime &&
- (systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
- mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
- mForcedRoute = 0;
- if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
- mHardwareStatus = AUDIO_HW_SET_ROUTING;
- mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
- mHardwareStatus = AUDIO_HW_IDLE;
- LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
- }
- mRouteRestoreTime = 0;
- }
- }
- break;
- }
}
-#ifdef WITH_A2DP
-// handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
-{
- if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) {
- if (mA2dpDisableCount++ == 0) {
- if (mA2dpEnabled) {
- setA2dpEnabled_l(false);
- mA2dpSuppressed = true;
- }
- }
- LOGV("mA2dpDisableCount incremented to %d", mA2dpDisableCount);
- } else {
- if (mA2dpDisableCount > 0) {
- if (--mA2dpDisableCount == 0) {
- if (mA2dpSuppressed) {
- setA2dpEnabled_l(true);
- mA2dpSuppressed = false;
- }
- }
- LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
- } else {
- LOGV("mA2dpDisableCount is already zero");
- }
+void AudioFlinger::ThreadBase::exit()
+{
+ // keep a strong ref on ourself so that we want get
+ // destroyed in the middle of requestExitAndWait()
+ sp <ThreadBase> strongMe = this;
+
+ LOGV("ThreadBase::exit");
+ {
+ AutoMutex lock(&mLock);
+ requestExit();
+ mWaitWorkCV.signal();
}
+ requestExitAndWait();
}
-#endif
-// ----------------------------------------------------------------------------
+uint32_t AudioFlinger::ThreadBase::sampleRate() const
+{
+ return mSampleRate;
+}
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
- : Thread(false),
- mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType),
- mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
- mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
- mInWrite(false)
+int AudioFlinger::ThreadBase::channelCount() const
{
- mSampleRate = output->sampleRate();
- mChannelCount = output->channelCount();
+ return mChannelCount;
+}
- // FIXME - Current mixer implementation only supports stereo output
- if (mChannelCount == 1) {
- LOGE("Invalid audio hardware channel count");
+int AudioFlinger::ThreadBase::format() const
+{
+ return mFormat;
+}
+
+size_t AudioFlinger::ThreadBase::frameCount() const
+{
+ return mFrameCount;
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+ status_t result;
+
+ Mutex::Autolock _l(mLock);
+ mNewParameters = keyValuePairs;
+
+ mWaitWorkCV.signal();
+ mParamCond.wait(mLock);
+
+ return mParamStatus;
+}
+
+void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
+{
+ Mutex::Autolock _l(mLock);
+ ConfigEvent *configEvent = new ConfigEvent();
+ configEvent->mEvent = event;
+ configEvent->mParam = param;
+ mConfigEvents.add(configEvent);
+ LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+ mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+ mLock.lock();
+ while(!mConfigEvents.isEmpty()) {
+ LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+ ConfigEvent *configEvent = mConfigEvents[0];
+ mConfigEvents.removeAt(0);
+ // release mLock because audioConfigChanged() will call
+ // Audioflinger::audioConfigChanged() which locks AudioFlinger mLock thus creating
+ // potential cross deadlock between AudioFlinger::mLock and mLock
+ mLock.unlock();
+ audioConfigChanged(configEvent->mEvent, configEvent->mParam);
+ delete configEvent;
+ mLock.lock();
}
+ mLock.unlock();
+}
- mFormat = output->format();
- mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
- mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
- // FIXME - Current mixer implementation only supports stereo output: Always
- // Allocate a stereo buffer even if HW output is mono.
- mMixBuffer = new int16_t[mFrameCount * 2];
- memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : ThreadBase(audioFlinger),
+ mOutput(output),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0),
+ mInWrite(false), mMixBuffer(0), mSuspended(false), mBytesWritten(0)
+{
+ readOutputParameters();
+
+ mMasterVolume = mAudioFlinger->masterVolume();
+ mMasterMute = mAudioFlinger->masterMute();
+
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
+ }
+ // notify client processes that a new input has been opened
+ sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
-AudioFlinger::MixerThread::~MixerThread()
+AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
- delete mAudioMixer;
+ if (mType != DUPLICATING) {
+ mAudioFlinger->mAudioHardware->closeOutputStream(mOutput);
+ }
}
-status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
+ snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
@@ -966,7 +830,7 @@ status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& a
}
}
- snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
+ snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
@@ -983,15 +847,13 @@ status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& a
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
- result.append(buffer);
- snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+ snprintf(buffer, SIZE, "Output thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
@@ -1008,238 +870,354 @@ status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>
}
// Thread virtuals
+status_t AudioFlinger::PlaybackThread::readyToRun()
+{
+ if (mSampleRate == 0) {
+ LOGE("No working audio driver found.");
+ return NO_INIT;
+ }
+ LOGI("AudioFlinger's thread %p ready to run", this);
+ return NO_ERROR;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Playback Thread %p", this);
+
+ run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+ const sp<AudioFlinger::Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ status_t *status)
+{
+ sp<Track> track;
+ status_t lStatus;
+
+ if (mType == DIRECT) {
+ if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) {
+ LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
+ sampleRate, format, channelCount, mOutput);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ } else {
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (sampleRate > mSampleRate*2) {
+ LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+
+ if (mOutput == 0) {
+ LOGE("Audio driver not initialized.");
+ lStatus = NO_INIT;
+ goto Exit;
+ }
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer);
+ if (track->getCblk() == NULL) {
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+ mTracks.add(track);
+ }
+ lStatus = NO_ERROR;
+
+Exit:
+ if(status) {
+ *status = lStatus;
+ }
+ return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+ if (mOutput) {
+ return mOutput->latency();
+ }
+ else {
+ return 0;
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+ mMasterVolume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+ mMasterMute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::PlaybackThread::masterVolume() const
+{
+ return mMasterVolume;
+}
+
+bool AudioFlinger::PlaybackThread::masterMute() const
+{
+ return mMasterMute;
+}
+
+status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
+{
+ mStreamTypes[stream].volume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
+{
+ mStreamTypes[stream].mute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::PlaybackThread::streamVolume(int stream) const
+{
+ return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::PlaybackThread::streamMute(int stream) const
+{
+ return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::PlaybackThread::isMusicActive() const
+{
+ Mutex::Autolock _l(mLock);
+ size_t count = mActiveTracks.size();
+ for (size_t i = 0 ; i < count ; ++i) {
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ if (t->mStreamType == AudioSystem::MUSIC)
+ return true;
+ }
+ return false;
+}
+
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+{
+ status_t status = ALREADY_EXISTS;
+
+ // here the track could be either new, or restarted
+ // in both cases "unstop" the track
+ if (track->isPaused()) {
+ track->mState = TrackBase::RESUMING;
+ LOGV("PAUSED => RESUMING (%d)", track->name());
+ } else {
+ track->mState = TrackBase::ACTIVE;
+ LOGV("? => ACTIVE (%d)", track->name());
+ }
+ // set retry count for buffer fill
+ track->mRetryCount = kMaxTrackStartupRetries;
+ if (mActiveTracks.indexOf(track) < 0) {
+ // the track is newly added, make sure it fills up all its
+ // buffers before playing. This is to ensure the client will
+ // effectively get the latency it requested.
+ track->mFillingUpStatus = Track::FS_FILLING;
+ track->mResetDone = false;
+ mActiveTracks.add(track);
+ status = NO_ERROR;
+ }
+
+ LOGV("mWaitWorkCV.broadcast");
+ mWaitWorkCV.broadcast();
+
+ return status;
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+{
+ track->mState = TrackBase::TERMINATED;
+ if (mActiveTracks.indexOf(track) < 0) {
+ LOGV("remove track (%d) and delete from mixer", track->name());
+ mTracks.remove(track);
+ deleteTrackName_l(track->name());
+ }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+ return mOutput->getParameters(keys);
+}
+
+void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = 0;
+
+ LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
+
+ switch (event) {
+ case AudioSystem::OUTPUT_OPENED:
+ case AudioSystem::OUTPUT_CONFIG_CHANGED:
+ desc.channels = mChannelCount;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = latency();
+ param2 = &desc;
+ break;
+
+ case AudioSystem::STREAM_CONFIG_CHANGED:
+ param2 = &param;
+ case AudioSystem::OUTPUT_CLOSED:
+ default:
+ break;
+ }
+ mAudioFlinger->audioConfigChanged(event, this, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+ mSampleRate = mOutput->sampleRate();
+ mChannelCount = AudioSystem::popCount(mOutput->channels());
+
+ mFormat = mOutput->format();
+ mFrameSize = mOutput->frameSize();
+ mFrameCount = mOutput->bufferSize() / mFrameSize;
+
+ mMinBytesToWrite = (mOutput->latency() * mSampleRate * mFrameSize) / 1000;
+ // FIXME - Current mixer implementation only supports stereo output: Always
+ // Allocate a stereo buffer even if HW output is mono.
+ if (mMixBuffer != NULL) delete mMixBuffer;
+ mMixBuffer = new int16_t[mFrameCount * 2];
+ memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : PlaybackThread(audioFlinger, output),
+ mAudioMixer(0)
+{
+ mType = PlaybackThread::MIXER;
+ mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+
+ // FIXME - Current mixer implementation only supports stereo output
+ if (mChannelCount == 1) {
+ LOGE("Invalid audio hardware channel count");
+ }
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+ delete mAudioMixer;
+}
+
bool AudioFlinger::MixerThread::threadLoop()
{
unsigned long sleepTime = kBufferRecoveryInUsecs;
int16_t* curBuf = mMixBuffer;
Vector< sp<Track> > tracksToRemove;
size_t enabledTracks = 0;
- nsecs_t standbyTime = systemTime();
- size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
+ nsecs_t standbyTime = systemTime();
+ size_t mixBufferSize = mFrameCount * mFrameSize;
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
-#ifdef WITH_A2DP
- bool outputTrackActive = false;
-#endif
+ while (!exitPending())
+ {
+ processConfigEvents();
- do {
enabledTracks = 0;
- { // scope for the AudioFlinger::mLock
-
- Mutex::Autolock _l(mAudioFlinger->mLock);
+ { // scope for mLock
-#ifdef WITH_A2DP
- if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
- if (outputTrackActive) {
- mAudioFlinger->mLock.unlock();
- mOutputTrack->stop();
- mAudioFlinger->mLock.lock();
- outputTrackActive = false;
- }
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount * mFrameSize;
+ maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
}
- mAudioFlinger->checkA2dpEnabledChange_l();
-#endif
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
- if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
- // wait until we have something to do...
- LOGV("Audio hardware entering standby, output %d\n", mOutputType);
+ if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
mOutput->standby();
mStandby = true;
+ mBytesWritten = 0;
}
-
-#ifdef WITH_A2DP
- if (outputTrackActive) {
- mAudioFlinger->mLock.unlock();
- mOutputTrack->stop();
- mAudioFlinger->mLock.lock();
- outputTrackActive = false;
- }
-#endif
- if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
- mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
- }
- // we're about to wait, flush the binder command buffer
- IPCThreadState::self()->flushCommands();
- mAudioFlinger->mWaitWorkCV.wait(mAudioFlinger->mLock);
- LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
-
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- LOGD("Silence is golden");
- setMasterMute(true);
- }
- }
-
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- continue;
- }
- // Forced route to speaker is handled by hardware mixer thread
- if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
- mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
- }
+ if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
- // find out which tracks need to be processed
- size_t count = activeTracks.size();
- for (size_t i=0 ; i<count ; i++) {
- sp<Track> t = activeTracks[i].promote();
- if (t == 0) continue;
+ if (exitPending()) break;
- Track* const track = t.get();
- audio_track_cblk_t* cblk = track->cblk();
+ // wait until we have something to do...
+ LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("MixerThread %p TID %d waking up\n", this, gettid());
- // The first time a track is added we wait
- // for all its buffers to be filled before processing it
- mAudioMixer->setActiveTrack(track->name());
- if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
- !track->isPaused())
- {
- //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
-
- // compute volume for this track
- int16_t left, right;
- if (track->isMuted() || mMasterMute || track->isPausing()) {
- left = right = 0;
- if (track->isPausing()) {
- LOGV("paused(%d)", track->name());
- track->setPaused();
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
}
- } else {
- float typeVolume = mStreamTypes[track->type()].volume;
- float v = mMasterVolume * typeVolume;
- float v_clamped = v * cblk->volume[0];
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- left = int16_t(v_clamped);
- v_clamped = v * cblk->volume[1];
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- right = int16_t(v_clamped);
}
- // XXX: these things DON'T need to be done each time
- mAudioMixer->setBufferProvider(track);
- mAudioMixer->enable(AudioMixer::MIXING);
-
- int param;
- if ( track->mFillingUpStatus == Track::FS_FILLED) {
- // no ramp for the first volume setting
- track->mFillingUpStatus = Track::FS_ACTIVE;
- if (track->mState == TrackBase::RESUMING) {
- track->mState = TrackBase::ACTIVE;
- param = AudioMixer::RAMP_VOLUME;
- } else {
- param = AudioMixer::VOLUME;
- }
- } else {
- param = AudioMixer::RAMP_VOLUME;
- }
- mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
- mAudioMixer->setParameter(
- AudioMixer::TRACK,
- AudioMixer::FORMAT, track->format());
- mAudioMixer->setParameter(
- AudioMixer::TRACK,
- AudioMixer::CHANNEL_COUNT, track->channelCount());
- mAudioMixer->setParameter(
- AudioMixer::RESAMPLE,
- AudioMixer::SAMPLE_RATE,
- int(cblk->sampleRate));
-
- // reset retry count
- track->mRetryCount = kMaxTrackRetries;
- enabledTracks++;
- } else {
- //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
- if (track->isStopped()) {
- track->reset();
- }
- if (track->isTerminated() || track->isStopped() || track->isPaused()) {
- // We have consumed all the buffers of this track.
- // Remove it from the list of active tracks.
- LOGV("remove(%d) from active list", track->name());
- tracksToRemove.add(track);
- } else {
- // No buffers for this track. Give it a few chances to
- // fill a buffer, then remove it from active list.
- if (--(track->mRetryCount) <= 0) {
- LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
- tracksToRemove.add(track);
- }
- }
- // LOGV("disable(%d)", track->name());
- mAudioMixer->disable(AudioMixer::MIXING);
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ continue;
}
}
- // remove all the tracks that need to be...
- count = tracksToRemove.size();
- if (UNLIKELY(count)) {
- for (size_t i=0 ; i<count ; i++) {
- const sp<Track>& track = tracksToRemove[i];
- removeActiveTrack_l(track);
- if (track->isTerminated()) {
- mTracks.remove(track);
- deleteTrackName_l(track->mName);
- }
- }
- }
+ enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
}
-
+
if (LIKELY(enabledTracks)) {
// mix buffers...
mAudioMixer->process(curBuf);
-#ifdef WITH_A2DP
- if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
- if (!outputTrackActive) {
- LOGV("starting output track in mixer for output %d", mOutputType);
- mOutputTrack->start();
- outputTrackActive = true;
- }
- mOutputTrack->write(curBuf, mFrameCount);
- }
-#endif
-
// output audio to hardware
- mLastWriteTime = systemTime();
- mInWrite = true;
- mOutput->write(curBuf, mixBufferSize);
- mNumWrites++;
- mInWrite = false;
- mStandby = false;
- nsecs_t temp = systemTime();
- standbyTime = temp + kStandbyTimeInNsecs;
- nsecs_t delta = temp - mLastWriteTime;
- if (delta > maxPeriod) {
- LOGW("write blocked for %llu msecs", ns2ms(delta));
- mNumDelayedWrites++;
- }
- sleepTime = kBufferRecoveryInUsecs;
- } else {
-#ifdef WITH_A2DP
- if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
- if (outputTrackActive) {
- mOutputTrack->write(curBuf, 0);
- if (mOutputTrack->bufferQueueEmpty()) {
- mOutputTrack->stop();
- outputTrackActive = false;
- } else {
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- }
+ if (mSuspended) {
+ usleep(kMaxBufferRecoveryInUsecs);
+ } else {
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize);
+ if (bytesWritten > 0) mBytesWritten += bytesWritten;
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ nsecs_t temp = systemTime();
+ standbyTime = temp + kStandbyTimeInNsecs;
+ nsecs_t delta = temp - mLastWriteTime;
+ if (delta > maxPeriod) {
+ LOGW("write blocked for %llu msecs", ns2ms(delta));
+ mNumDelayedWrites++;
}
+ sleepTime = kBufferRecoveryInUsecs;
}
-#endif
+ } else {
// There was nothing to mix this round, which means all
// active tracks were late. Sleep a little bit to give
// them another chance. If we're too late, the audio
// hardware will zero-fill for us.
- //LOGV("no buffers - usleep(%lu)", sleepTime);
+ // LOGV("thread %p no buffers - usleep(%lu)", this, sleepTime);
usleep(sleepTime);
if (sleepTime < kMaxBufferRecoveryInUsecs) {
sleepTime += kBufferRecoveryInUsecs;
@@ -1250,101 +1228,165 @@ bool AudioFlinger::MixerThread::threadLoop()
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
- } while (true);
-
- return false;
-}
+ }
-status_t AudioFlinger::MixerThread::readyToRun()
-{
- if (mSampleRate == 0) {
- LOGE("No working audio driver found.");
- return NO_INIT;
+ if (!mStandby) {
+ mOutput->standby();
}
- LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
- return NO_ERROR;
+ sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+ processConfigEvents();
+
+ LOGV("MixerThread %p exiting", this);
+ return false;
}
-void AudioFlinger::MixerThread::onFirstRef()
+// prepareTracks_l() must be called with ThreadBase::mLock held
+size_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
{
- const size_t SIZE = 256;
- char buffer[SIZE];
-
- snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
- run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
-}
+ size_t enabledTracks = 0;
+ // find out which tracks need to be processed
+ size_t count = activeTracks.size();
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = activeTracks[i].promote();
+ if (t == 0) continue;
-// MixerThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
- const sp<AudioFlinger::Client>& client,
- int streamType,
- uint32_t sampleRate,
- int format,
- int channelCount,
- int frameCount,
- const sp<IMemory>& sharedBuffer,
- status_t *status)
-{
- sp<Track> track;
- status_t lStatus;
-
- // Resampler implementation limits input sampling rate to 2 x output sampling rate.
- if (sampleRate > mSampleRate*2) {
- LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
- lStatus = BAD_VALUE;
- goto Exit;
- }
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ mAudioMixer->setActiveTrack(track->name());
+ if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+ !track->isPaused())
+ {
+ //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ // compute volume for this track
+ int16_t left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing() ||
+ mStreamTypes[track->type()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = int16_t(v_clamped);
+ v_clamped = v * cblk->volume[1];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = int16_t(v_clamped);
+ }
- if (mSampleRate == 0) {
- LOGE("Audio driver not initialized.");
- lStatus = NO_INIT;
- goto Exit;
+ // XXX: these things DON'T need to be done each time
+ mAudioMixer->setBufferProvider(track);
+ mAudioMixer->enable(AudioMixer::MIXING);
+
+ int param;
+ if ( track->mFillingUpStatus == Track::FS_FILLED) {
+ // no ramp for the first volume setting
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ param = AudioMixer::RAMP_VOLUME;
+ } else {
+ param = AudioMixer::VOLUME;
+ }
+ } else {
+ param = AudioMixer::RAMP_VOLUME;
+ }
+ mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
+ mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
+ mAudioMixer->setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::FORMAT, track->format());
+ mAudioMixer->setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::CHANNEL_COUNT, track->channelCount());
+ mAudioMixer->setParameter(
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ int(cblk->sampleRate));
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ enabledTracks++;
+ } else {
+ //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if (track->isStopped()) {
+ track->reset();
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ tracksToRemove->add(track);
+ mAudioMixer->disable(AudioMixer::MIXING);
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ tracksToRemove->add(track);
+ }
+ // For tracks using static shared memry buffer, make sure that we have
+ // written enough data to audio hardware before disabling the track
+ // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+ // don't care about code removing track from active list above.
+ if ((track->mSharedBuffer == 0) || (mBytesWritten >= mMinBytesToWrite)) {
+ mAudioMixer->disable(AudioMixer::MIXING);
+ } else {
+ enabledTracks++;
+ }
+ }
+ }
}
- track = new Track(this, client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer);
- if (track->getCblk() == NULL) {
- lStatus = NO_MEMORY;
- goto Exit;
+ // remove all the tracks that need to be...
+ count = tracksToRemove->size();
+ if (UNLIKELY(count)) {
+ for (size_t i=0 ; i<count ; i++) {
+ const sp<Track>& track = tracksToRemove->itemAt(i);
+ mActiveTracks.remove(track);
+ if (track->isTerminated()) {
+ mTracks.remove(track);
+ deleteTrackName_l(track->mName);
+ }
+ }
}
- mTracks.add(track);
- lStatus = NO_ERROR;
-Exit:
- if(status) {
- *status = lStatus;
- }
- return track;
+ return enabledTracks;
}
-// getTracks_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::getTracks_l(
+void AudioFlinger::MixerThread::getTracks(
SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks)
+ SortedVector < wp<Track> >& activeTracks,
+ int streamType)
{
+ LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size());
+ Mutex::Autolock _l(mLock);
size_t size = mTracks.size();
- LOGV ("MixerThread::getTracks_l() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType, mTracks.size(), mActiveTracks.size());
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
- if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
+ if (t->type() == streamType) {
tracks.add(t);
int j = mActiveTracks.indexOf(t);
if (j >= 0) {
t = mActiveTracks[j].promote();
if (t != NULL) {
- activeTracks.add(t);
- }
+ activeTracks.add(t);
+ }
}
}
}
size = activeTracks.size();
for (size_t i = 0; i < size; i++) {
- removeActiveTrack_l(activeTracks[i]);
+ mActiveTracks.remove(activeTracks[i]);
}
-
+
size = tracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = tracks[i];
@@ -1353,219 +1395,554 @@ void AudioFlinger::MixerThread::getTracks_l(
}
}
-// putTracks_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::putTracks_l(
+void AudioFlinger::MixerThread::putTracks(
SortedVector < sp<Track> >& tracks,
SortedVector < wp<Track> >& activeTracks)
{
-
- LOGV ("MixerThread::putTracks_l() for output %d, tracks.size %d, activeTracks.size %d", mOutputType, tracks.size(), activeTracks.size());
-
+ LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size());
+ Mutex::Autolock _l(mLock);
size_t size = tracks.size();
for (size_t i = 0; i < size ; i++) {
sp<Track> t = tracks[i];
int name = getTrackName_l();
if (name < 0) return;
-
+
t->mName = name;
- t->mMixerThread = this;
+ t->mThread = this;
mTracks.add(t);
int j = activeTracks.indexOf(t);
if (j >= 0) {
- addActiveTrack_l(t);
- }
+ mActiveTracks.add(t);
+ }
}
}
-uint32_t AudioFlinger::MixerThread::sampleRate() const
-{
- return mSampleRate;
-}
-
-int AudioFlinger::MixerThread::channelCount() const
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::MixerThread::getTrackName_l()
{
- return mChannelCount;
+ return mAudioMixer->getTrackName();
}
-int AudioFlinger::MixerThread::format() const
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::MixerThread::deleteTrackName_l(int name)
{
- return mFormat;
+ mAudioMixer->deleteTrackName(name);
}
-size_t AudioFlinger::MixerThread::frameCount() const
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
{
- return mFrameCount;
-}
+ bool reconfig = false;
-uint32_t AudioFlinger::MixerThread::latency() const
-{
- if (mOutput) {
- return mOutput->latency();
- }
- else {
- return 0;
+ if (mNewParameters != "") {
+ status_t status = NO_ERROR;
+ AudioParameter param = AudioParameter(mNewParameters);
+ int value;
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ if (value != AudioSystem::PCM_16_BIT) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ if (value != AudioSystem::CHANNEL_OUT_STEREO) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mOutput->setParameters(mNewParameters);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(mNewParameters);
+ }
+ if (status == NO_ERROR && reconfig) {
+ delete mAudioMixer;
+ readOutputParameters();
+ mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+ for (size_t i = 0; i < mTracks.size() ; i++) {
+ int name = getTrackName_l();
+ if (name < 0) break;
+ mTracks[i]->mName = name;
+ }
+ sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+ mParamStatus = status;
+ mNewParameters = "";
+ mParamCond.signal();
}
+ return reconfig;
}
-status_t AudioFlinger::MixerThread::setMasterVolume(float value)
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
{
- mMasterVolume = value;
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ PlaybackThread::dumpInternals(fd, args);
+
+ snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
+// ----------------------------------------------------------------------------
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : PlaybackThread(audioFlinger, output),
+ mLeftVolume (1.0), mRightVolume(1.0)
{
- mMasterMute = muted;
- return NO_ERROR;
+ mType = PlaybackThread::DIRECT;
}
-float AudioFlinger::MixerThread::masterVolume() const
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
- return mMasterVolume;
}
-bool AudioFlinger::MixerThread::masterMute() const
+
+bool AudioFlinger::DirectOutputThread::threadLoop()
{
- return mMasterMute;
+ unsigned long sleepTime = kBufferRecoveryInUsecs;
+ sp<Track> trackToRemove;
+ sp<Track> activeTrack;
+ nsecs_t standbyTime = systemTime();
+ int8_t *curBuf;
+ size_t mixBufferSize = mFrameCount*mFrameSize;
+
+ while (!exitPending())
+ {
+ processConfigEvents();
+
+ { // scope for the mLock
+
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount*mFrameSize;
+ }
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ // wait until we have something to do...
+ if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p\n", this);
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ }
+
+ if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+
+ if (exitPending()) break;
+
+ LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ continue;
+ }
+ }
+
+ // find out which tracks need to be processed
+ if (mActiveTracks.size() != 0) {
+ sp<Track> t = mActiveTracks[0].promote();
+ if (t == 0) continue;
+
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+ !track->isPaused())
+ {
+ //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ // compute volume for this track
+ float left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing() ||
+ mStreamTypes[track->type()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = v_clamped/MAX_GAIN;
+ v_clamped = v * cblk->volume[1];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = v_clamped/MAX_GAIN;
+ }
+
+ if (left != mLeftVolume || right != mRightVolume) {
+ mOutput->setVolume(left, right);
+ left = mLeftVolume;
+ right = mRightVolume;
+ }
+
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ }
+ }
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ activeTrack = t;
+ } else {
+ //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if (track->isStopped()) {
+ track->reset();
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ trackToRemove = track;
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ trackToRemove = track;
+ }
+
+ // For tracks using static shared memry buffer, make sure that we have
+ // written enough data to audio hardware before disabling the track
+ // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+ // don't care about code removing track from active list above.
+ if ((track->mSharedBuffer != 0) && (mBytesWritten < mMinBytesToWrite)) {
+ activeTrack = t;
+ }
+ }
+ }
+ }
+
+ // remove all the tracks that need to be...
+ if (UNLIKELY(trackToRemove != 0)) {
+ mActiveTracks.remove(trackToRemove);
+ if (trackToRemove->isTerminated()) {
+ mTracks.remove(trackToRemove);
+ deleteTrackName_l(trackToRemove->mName);
+ }
+ }
+ }
+
+ if (activeTrack != 0) {
+ AudioBufferProvider::Buffer buffer;
+ size_t frameCount = mFrameCount;
+ curBuf = (int8_t *)mMixBuffer;
+ // output audio to hardware
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ while(frameCount) {
+ buffer.frameCount = frameCount;
+ activeTrack->getNextBuffer(&buffer);
+ if (UNLIKELY(buffer.raw == 0)) {
+ memset(curBuf, 0, frameCount * mFrameSize);
+ break;
+ }
+ memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+ frameCount -= buffer.frameCount;
+ curBuf += buffer.frameCount * mFrameSize;
+ activeTrack->releaseBuffer(&buffer);
+ }
+ if (mSuspended) {
+ usleep(kMaxBufferRecoveryInUsecs);
+ } else {
+ int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+ if (bytesWritten) mBytesWritten += bytesWritten;
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ nsecs_t temp = systemTime();
+ standbyTime = temp + kStandbyTimeInNsecs;
+ sleepTime = kBufferRecoveryInUsecs;
+ }
+ } else {
+ // There was nothing to mix this round, which means all
+ // active tracks were late. Sleep a little bit to give
+ // them another chance. If we're too late, the audio
+ // hardware will zero-fill for us.
+ //LOGV("no buffers - usleep(%lu)", sleepTime);
+ usleep(sleepTime);
+ if (sleepTime < kMaxBufferRecoveryInUsecs) {
+ sleepTime += kBufferRecoveryInUsecs;
+ }
+ }
+
+ // finally let go of removed track, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ trackToRemove.clear();
+ activeTrack.clear();
+ }
+
+ if (!mStandby) {
+ mOutput->standby();
+ }
+ sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+ processConfigEvents();
+
+ LOGV("DirectOutputThread %p exiting", this);
+ return false;
}
-status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l()
{
- mStreamTypes[stream].volume = value;
- return NO_ERROR;
+ return 0;
}
-status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
{
- mStreamTypes[stream].mute = muted;
- return NO_ERROR;
}
-float AudioFlinger::MixerThread::streamVolume(int stream) const
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
{
- return mStreamTypes[stream].volume;
+ bool reconfig = false;
+
+ if (mNewParameters != "") {
+ status_t status = NO_ERROR;
+ AudioParameter param = AudioParameter(mNewParameters);
+ int value;
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mOutput->setParameters(mNewParameters);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(mNewParameters);
+ }
+ if (status == NO_ERROR && reconfig) {
+ readOutputParameters();
+ sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+ mParamStatus = status;
+ mNewParameters = "";
+ mParamCond.signal();
+ }
+ return reconfig;
}
-bool AudioFlinger::MixerThread::streamMute(int stream) const
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread)
+ : MixerThread(audioFlinger, mainThread->getOutput())
{
- return mStreamTypes[stream].mute;
+ mType = PlaybackThread::DUPLICATING;
+ addOutputTrack(mainThread);
}
-// isMusicActive_l() must be called with AudioFlinger::mLock held
-bool AudioFlinger::MixerThread::isMusicActive_l() const
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
{
- size_t count = mActiveTracks.size();
- for (size_t i = 0 ; i < count ; ++i) {
- sp<Track> t = mActiveTracks[i].promote();
- if (t == 0) continue;
- Track* const track = t.get();
- if (t->mStreamType == AudioSystem::MUSIC)
- return true;
- }
- return false;
+ mOutputTracks.clear();
}
-// addTrack_l() must be called with AudioFlinger::mLock held
-status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
+bool AudioFlinger::DuplicatingThread::threadLoop()
{
- status_t status = ALREADY_EXISTS;
+ unsigned long sleepTime = kBufferRecoveryInUsecs;
+ int16_t* curBuf = mMixBuffer;
+ Vector< sp<Track> > tracksToRemove;
+ size_t enabledTracks = 0;
+ nsecs_t standbyTime = systemTime();
+ size_t mixBufferSize = mFrameCount*mFrameSize;
+ SortedVector< sp<OutputTrack> > outputTracks;
- // here the track could be either new, or restarted
- // in both cases "unstop" the track
- if (track->isPaused()) {
- track->mState = TrackBase::RESUMING;
- LOGV("PAUSED => RESUMING (%d)", track->name());
- } else {
- track->mState = TrackBase::ACTIVE;
- LOGV("? => ACTIVE (%d)", track->name());
- }
- // set retry count for buffer fill
- track->mRetryCount = kMaxTrackStartupRetries;
- if (mActiveTracks.indexOf(track) < 0) {
- // the track is newly added, make sure it fills up all its
- // buffers before playing. This is to ensure the client will
- // effectively get the latency it requested.
- track->mFillingUpStatus = Track::FS_FILLING;
- track->mResetDone = false;
- addActiveTrack_l(track);
- status = NO_ERROR;
- }
-
- LOGV("mWaitWorkCV.broadcast");
- mAudioFlinger->mWaitWorkCV.broadcast();
+ while (!exitPending())
+ {
+ processConfigEvents();
- return status;
-}
+ enabledTracks = 0;
+ { // scope for the mLock
-// destroyTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
-{
- track->mState = TrackBase::TERMINATED;
- if (mActiveTracks.indexOf(track) < 0) {
- LOGV("remove track (%d) and delete from mixer", track->name());
- mTracks.remove(track);
- deleteTrackName_l(track->name());
- }
-}
+ Mutex::Autolock _l(mLock);
-// addActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::addActiveTrack_l(const wp<Track>& t)
-{
- mActiveTracks.add(t);
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount*mFrameSize;
+ }
- // Force routing to speaker for certain stream types
- // The forced routing to speaker is managed by hardware mixer
- if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
- sp<Track> track = t.promote();
- if (track == NULL) return;
-
- if (streamForcedToSpeaker(track->type())) {
- mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
- }
- }
-}
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
-// removeActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::removeActiveTrack_l(const wp<Track>& t)
-{
- mActiveTracks.remove(t);
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ outputTracks.add(mOutputTracks[i]);
+ }
- // Force routing to speaker for certain stream types
- // The forced routing to speaker is managed by hardware mixer
- if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
- sp<Track> track = t.promote();
- if (track == NULL) return;
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ if (!mStandby) {
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ mLock.unlock();
+ outputTracks[i]->stop();
+ mLock.lock();
+ }
+ mStandby = true;
+ mBytesWritten = 0;
+ }
- if (streamForcedToSpeaker(track->type())) {
- mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
+ if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+ outputTracks.clear();
+
+ if (exitPending()) break;
+
+ LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ sleepTime = kBufferRecoveryInUsecs;
+ continue;
+ }
+ }
+
+ enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
+ }
+
+ bool mustSleep = true;
+ if (LIKELY(enabledTracks)) {
+ // mix buffers...
+ mAudioMixer->process(curBuf);
+ if (!mSuspended) {
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->write(curBuf, mFrameCount);
+ }
+ mStandby = false;
+ mustSleep = false;
+ mBytesWritten += mixBufferSize;
+ }
+ } else {
+ // flush remaining overflow buffers in output tracks
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ if (outputTracks[i]->isActive()) {
+ outputTracks[i]->write(curBuf, 0);
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ mustSleep = false;
+ }
+ }
}
+ if (mustSleep) {
+// LOGV("threadLoop() sleeping %d", sleepTime);
+ usleep(sleepTime);
+ if (sleepTime < kMaxBufferRecoveryInUsecs) {
+ sleepTime += kBufferRecoveryInUsecs;
+ }
+ } else {
+ sleepTime = kBufferRecoveryInUsecs;
+ }
+
+ // finally let go of all our tracks, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ tracksToRemove.clear();
+ outputTracks.clear();
}
-}
-// getTrackName_l() must be called with AudioFlinger::mLock held
-int AudioFlinger::MixerThread::getTrackName_l()
-{
- return mAudioMixer->getTrackName();
+ if (!mStandby) {
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ mLock.unlock();
+ outputTracks[i]->stop();
+ mLock.lock();
+ }
+ }
+
+ sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+ processConfigEvents();
+
+ return false;
}
-// deleteTrackName_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::deleteTrackName_l(int name)
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
- mAudioMixer->deleteTrackName(name);
+ int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
+ OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+ mSampleRate,
+ mFormat,
+ mChannelCount,
+ frameCount);
+ thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+ mOutputTracks.add(outputTrack);
+ LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
}
-size_t AudioFlinger::MixerThread::getOutputFrameCount()
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
{
- return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+ mOutputTracks.removeAt(i);
+ return;
+ }
+ }
+ LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
}
+
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::TrackBase::TrackBase(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
@@ -1574,7 +1951,7 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
uint32_t flags,
const sp<IMemory>& sharedBuffer)
: RefBase(),
- mMixerThread(mixerThread),
+ mThread(thread),
mClient(client),
mFrameCount(0),
mState(IDLE),
@@ -1582,13 +1959,6 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK)
{
- mName = mixerThread->getTrackName_l();
- LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
- if (mName < 0) {
- LOGE("no more track names availlable");
- return;
- }
-
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
@@ -1642,16 +2012,19 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
}
}
-AudioFlinger::MixerThread::TrackBase::~TrackBase()
+AudioFlinger::PlaybackThread::TrackBase::~TrackBase()
{
if (mCblk) {
- mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ if (mClient == NULL) {
+ delete mCblk;
+ }
}
mCblkMemory.clear(); // and free the shared memory
mClient.clear();
}
-void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void AudioFlinger::PlaybackThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
@@ -1659,7 +2032,7 @@ void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Bu
buffer->frameCount = 0;
}
-bool AudioFlinger::MixerThread::TrackBase::step() {
+bool AudioFlinger::PlaybackThread::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
@@ -1671,7 +2044,7 @@ bool AudioFlinger::MixerThread::TrackBase::step() {
return result;
}
-void AudioFlinger::MixerThread::TrackBase::reset() {
+void AudioFlinger::PlaybackThread::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
@@ -1682,27 +2055,27 @@ void AudioFlinger::MixerThread::TrackBase::reset() {
LOGV("TrackBase::reset");
}
-sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
+sp<IMemory> AudioFlinger::PlaybackThread::TrackBase::getCblk() const
{
return mCblkMemory;
}
-int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
+int AudioFlinger::PlaybackThread::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
-int AudioFlinger::MixerThread::TrackBase::channelCount() const {
+int AudioFlinger::PlaybackThread::TrackBase::channelCount() const {
return (int)mCblk->channels;
}
-void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+void* AudioFlinger::PlaybackThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
- int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
- int16_t *bufferEnd = bufferStart + frames * cblk->channels;
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
+ int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
- if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- (cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
+ if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
+ ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channels %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
@@ -1715,9 +2088,9 @@ void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t
// ----------------------------------------------------------------------------
-// Track constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::Track::Track(
- const sp<MixerThread>& mixerThread,
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
@@ -1725,40 +2098,58 @@ AudioFlinger::MixerThread::Track::Track(
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
- : TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+ : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer),
+ mMute(false), mSharedBuffer(sharedBuffer), mName(-1)
{
+ sp<ThreadBase> baseThread = thread.promote();
+ if (baseThread != 0) {
+ PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
+ mName = playbackThread->getTrackName_l();
+ }
+ LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ if (mName < 0) {
+ LOGE("no more track names available");
+ }
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
- mMute = false;
- mSharedBuffer = sharedBuffer;
mStreamType = streamType;
+ // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
+ // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
+ mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
}
-AudioFlinger::MixerThread::Track::~Track()
+AudioFlinger::PlaybackThread::Track::~Track()
{
- wp<Track> weak(this); // never create a strong ref from the dtor
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- mState = TERMINATED;
+ LOGV("PlaybackThread::Track destructor");
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ mState = TERMINATED;
+ }
}
-void AudioFlinger::MixerThread::Track::destroy()
+void AudioFlinger::PlaybackThread::Track::destroy()
{
- // NOTE: destroyTrack_l() can remove a strong reference to this Track
+ // NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
- // desctructor is called. As the destructor needs to lock AudioFlinger::mLock,
- // we must acquire a strong reference on this Track before locking AudioFlinger::mLock
+ // desctructor is called. As the destructor needs to lock mLock,
+ // we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
- // On the other hand, as long as Track::destroy() is only called by
- // TrackHandle destructor, the TrackHandle still holds a strong ref on
+ // On the other hand, as long as Track::destroy() is only called by
+ // TrackHandle destructor, the TrackHandle still holds a strong ref on
// this Track with its member mTrack.
sp<Track> keep(this);
- { // scope for AudioFlinger::mLock
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- mMixerThread->destroyTrack_l(this);
+ { // scope for mLock
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->destroyTrack_l(this);
+ }
}
}
-void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
mName - AudioMixer::TRACK0,
@@ -1777,7 +2168,7 @@ void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
mCblk->user);
}
-status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
@@ -1814,76 +2205,90 @@ status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Bu
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
+ LOGV("getNextBuffer() no more data");
return NOT_ENOUGH_DATA;
}
-bool AudioFlinger::MixerThread::Track::isReady() const {
+bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
mCblk->forceReady) {
mFillingUpStatus = FS_FILLED;
mCblk->forceReady = 0;
- LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
return true;
}
return false;
}
-status_t AudioFlinger::MixerThread::Track::start()
+status_t AudioFlinger::PlaybackThread::Track::start()
{
- LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- mMixerThread->addTrack_l(this);
+ LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->addTrack_l(this);
+ }
return NO_ERROR;
}
-void AudioFlinger::MixerThread::Track::stop()
+void AudioFlinger::PlaybackThread::Track::stop()
{
- LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- if (mState > STOPPED) {
- mState = STOPPED;
- // If the track is not active (PAUSED and buffers full), flush buffers
- if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
- reset();
+ LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState > STOPPED) {
+ mState = STOPPED;
+ // If the track is not active (PAUSED and buffers full), flush buffers
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ }
+ LOGV("(> STOPPED) => STOPPED (%d)", mName);
}
- LOGV("(> STOPPED) => STOPPED (%d)", mName);
}
}
-void AudioFlinger::MixerThread::Track::pause()
+void AudioFlinger::PlaybackThread::Track::pause()
{
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- if (mState == ACTIVE || mState == RESUMING) {
- mState = PAUSING;
- LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState == ACTIVE || mState == RESUMING) {
+ mState = PAUSING;
+ LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+ }
}
}
-void AudioFlinger::MixerThread::Track::flush()
+void AudioFlinger::PlaybackThread::Track::flush()
{
LOGV("flush(%d)", mName);
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
- return;
- }
- // No point remaining in PAUSED state after a flush => go to
- // STOPPED state
- mState = STOPPED;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+ return;
+ }
+ // No point remaining in PAUSED state after a flush => go to
+ // STOPPED state
+ mState = STOPPED;
- mCblk->lock.lock();
- // NOTE: reset() will reset cblk->user and cblk->server with
- // the risk that at the same time, the AudioMixer is trying to read
- // data. In this case, getNextBuffer() would return a NULL pointer
- // as audio buffer => the AudioMixer code MUST always test that pointer
- // returned by getNextBuffer() is not NULL!
- reset();
- mCblk->lock.unlock();
+ mCblk->lock.lock();
+ // NOTE: reset() will reset cblk->user and cblk->server with
+ // the risk that at the same time, the AudioMixer is trying to read
+ // data. In this case, getNextBuffer() would return a NULL pointer
+ // as audio buffer => the AudioMixer code MUST always test that pointer
+ // returned by getNextBuffer() is not NULL!
+ reset();
+ mCblk->lock.unlock();
+ }
}
-void AudioFlinger::MixerThread::Track::reset()
+void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
@@ -1893,17 +2298,17 @@ void AudioFlinger::MixerThread::Track::reset()
// written to buffer
mCblk->flowControlFlag = 1;
mCblk->forceReady = 0;
- mFillingUpStatus = FS_FILLING;
+ mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
-void AudioFlinger::MixerThread::Track::mute(bool muted)
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
{
mMute = muted;
}
-void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
+void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
@@ -1912,28 +2317,33 @@ void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::RecordTrack::RecordTrack(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
- int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags)
- : TrackBase(mixerThread, client, sampleRate, format,
+ : TrackBase(thread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
- mOverflow(false), mInputSource(inputSource)
+ mOverflow(false)
{
+ LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AudioSystem::PCM_16_BIT) {
+ mCblk->frameSize = channelCount * sizeof(int16_t);
+ } else if (format == AudioSystem::PCM_8_BIT) {
+ mCblk->frameSize = channelCount * sizeof(int8_t);
+ } else {
+ mCblk->frameSize = sizeof(int8_t);
+ }
}
-AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- mMixerThread->deleteTrackName_l(mName);
}
-status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
@@ -1972,180 +2382,231 @@ getNextBuffer_exit:
return NOT_ENOUGH_DATA;
}
-status_t AudioFlinger::MixerThread::RecordTrack::start()
+status_t AudioFlinger::RecordThread::RecordTrack::start()
{
- return mMixerThread->mAudioFlinger->startRecord(this);
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ return recordThread->start(this);
+ }
+ return NO_INIT;
}
-void AudioFlinger::MixerThread::RecordTrack::stop()
+void AudioFlinger::RecordThread::RecordTrack::stop()
{
- mMixerThread->mAudioFlinger->stopRecord(this);
- TrackBase::reset();
- // Force overerrun condition to avoid false overrun callback until first data is
- // read from buffer
- mCblk->flowControlFlag = 1;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ recordThread->stop(this);
+ TrackBase::reset();
+ // Force overerrun condition to avoid false overrun callback until first data is
+ // read from buffer
+ mCblk->flowControlFlag = 1;
+ }
}
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::OutputTrack::OutputTrack(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+ const wp<ThreadBase>& thread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
- : Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
- mOutputMixerThread(mixerThread)
+ : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL),
+ mActive(false)
{
-
+
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
mCblk->out = 1;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
- mCblk->bufferTimeoutMs = 10;
-
- LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",
- mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
-
+ mWaitTimeMs = (playbackThread->frameCount() * 2 * 1000) / playbackThread->sampleRate();
+
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p mWaitTimeMs %d",
+ mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd, mWaitTimeMs);
+
}
-AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
stop();
}
-status_t AudioFlinger::MixerThread::OutputTrack::start()
+status_t AudioFlinger::PlaybackThread::OutputTrack::start()
{
status_t status = Track::start();
-
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ mActive = true;
mRetryCount = 127;
return status;
}
-void AudioFlinger::MixerThread::OutputTrack::stop()
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
+ mActive = false;
}
-void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channels = mCblk->channels;
-
+ bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
-
- if (mCblk->user == 0) {
- mOutputMixerThread->mAudioFlinger->mLock.lock();
- bool isMusicActive = mOutputMixerThread->isMusicActive_l();
- mOutputMixerThread->mAudioFlinger->mLock.unlock();
- if (isMusicActive) {
- mCblk->forceReady = 1;
- LOGV("OutputTrack::start() force ready");
- } else if (mCblk->frameCount > frames){
- if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
- uint32_t startFrames = (mCblk->frameCount - frames);
- LOGV("OutputTrack::start() write %d frames", startFrames);
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[startFrames * channels];
- pInBuffer->frameCount = startFrames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
- } else {
- LOGW ("OutputTrack::write() no more buffers");
+
+ uint32_t waitTimeLeftMs = mWaitTimeMs;
+
+ if (!mActive) {
+ start();
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ MixerThread *mixerThread = (MixerThread *)thread.get();
+ if (mCblk->frameCount > frames){
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+ uint32_t startFrames = (mCblk->frameCount - frames);
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[startFrames * channels];
+ pInBuffer->frameCount = startFrames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ LOGW ("OutputTrack::write() %p no more buffers in queue", this);
+ }
}
- }
+ }
}
- while (1) {
+ while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
-
+
if (pInBuffer->frameCount == 0) {
break;
}
-
+
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
- if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ nsecs_t startTime = systemTime();
+ if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ LOGV ("OutputTrack::write() %p no more output buffers", this);
+ outputBufferFull = true;
break;
}
+ uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+// LOGV("OutputTrack::write() waitTimeMs %d waitTimeLeftMs %d", waitTimeMs, waitTimeLeftMs)
+ if (waitTimeLeftMs >= waitTimeMs) {
+ waitTimeLeftMs -= waitTimeMs;
+ } else {
+ waitTimeLeftMs = 0;
+ }
}
-
+
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channels;
mOutBuffer.frameCount -= outFrames;
- mOutBuffer.i16 += outFrames * channels;
-
+ mOutBuffer.i16 += outFrames * channels;
+
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
+ LOGV("OutputTrack::write() %p released overflow buffer %d", this, mBufferQueue.size());
} else {
break;
}
}
}
-
+
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
- if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
+ LOGV("OutputTrack::write() %p adding overflow buffer %d", this, mBufferQueue.size());
} else {
- LOGW("OutputTrack::write() no more buffers");
+ LOGW("OutputTrack::write() %p no more overflow buffers", this);
}
}
-
+
// Calling write() with a 0 length buffer, means that no more data will be written:
- // If no more buffers are pending, fill output track buffer to make sure it is started
+ // If no more buffers are pending, fill output track buffer to make sure it is started
// by output mixer.
- if (frames == 0 && mBufferQueue.size() == 0 && mCblk->user < mCblk->frameCount) {
- frames = mCblk->frameCount - mCblk->user;
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[frames * channels];
- pInBuffer->frameCount = frames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
+ if (frames == 0 && mBufferQueue.size() == 0) {
+ if (mCblk->user < mCblk->frameCount) {
+ frames = mCblk->frameCount - mCblk->user;
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[frames * channels];
+ pInBuffer->frameCount = frames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ stop();
+ }
}
+ return outputBufferFull;
}
-status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
int active;
- int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = buffer->frameCount;
- LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
buffer->frameCount = 0;
-
+
uint32_t framesAvail = cblk->framesAvailable();
+
if (framesAvail == 0) {
- return AudioTrack::NO_MORE_BUFFERS;
+ Mutex::Autolock _l(cblk->lock);
+ goto start_loop_here;
+ while (framesAvail == 0) {
+ active = mActive;
+ if (UNLIKELY(!active)) {
+ LOGV("Not active and NO_MORE_BUFFERS");
+ return AudioTrack::NO_MORE_BUFFERS;
+ }
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ if (result != NO_ERROR) {
+ return AudioTrack::NO_MORE_BUFFERS;
+ }
+ // read the server count again
+ start_loop_here:
+ framesAvail = cblk->framesAvailable_l();
+ }
}
+// if (framesAvail < framesReq) {
+// return AudioTrack::NO_MORE_BUFFERS;
+// }
+
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
@@ -2163,11 +2624,11 @@ status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvide
}
-void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
+void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
Buffer *pBuffer;
-
+
for (size_t i = 0; i < size; i++) {
pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
@@ -2199,7 +2660,7 @@ const sp<MemoryDealer>& AudioFlinger::Client::heap() const
// ----------------------------------------------------------------------------
-AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
@@ -2251,7 +2712,7 @@ status_t AudioFlinger::TrackHandle::onTransact(
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int inputSource,
+ void *input,
uint32_t sampleRate,
int format,
int channelCount,
@@ -2259,14 +2720,13 @@ sp<IAudioRecord> AudioFlinger::openRecord(
uint32_t flags,
status_t *status)
{
- sp<MixerThread::RecordTrack> recordTrack;
+ sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
- AudioStreamIn* input = 0;
- int inFrameCount;
- size_t inputBufferSize;
status_t lStatus;
+ RecordThread *thread;
+ size_t inFrameCount;
// check calling permissions
if (!recordingAllowed()) {
@@ -2274,30 +2734,15 @@ sp<IAudioRecord> AudioFlinger::openRecord(
goto Exit;
}
- if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
- LOGE("invalid stream type");
- lStatus = BAD_VALUE;
- goto Exit;
- }
-
- if (mAudioRecordThread == 0) {
- LOGE("Audio record thread not started");
- lStatus = NO_INIT;
- goto Exit;
- }
-
-
- // Check that audio input stream accepts requested audio parameters
- inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
- if (inputBufferSize == 0) {
- lStatus = BAD_VALUE;
- LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
- goto Exit;
- }
-
// add client to list
{ // scope for mLock
Mutex::Autolock _l(mLock);
+ thread = checkRecordThread_l(input);
+ if (thread == NULL) {
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
@@ -2306,12 +2751,8 @@ sp<IAudioRecord> AudioFlinger::openRecord(
mClients.add(pid, client);
}
- // frameCount must be a multiple of input buffer size
- inFrameCount = inputBufferSize/channelCount/sizeof(short);
- frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
-
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
- recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
+ recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
format, channelCount, frameCount, flags);
}
if (recordTrack->getCblk() == NULL) {
@@ -2331,22 +2772,9 @@ Exit:
return recordHandle;
}
-status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
- if (mAudioRecordThread != 0) {
- return mAudioRecordThread->start(recordTrack);
- }
- return NO_INIT;
-}
-
-void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
- if (mAudioRecordThread != 0) {
- mAudioRecordThread->stop(recordTrack);
- }
-}
-
// ----------------------------------------------------------------------------
-AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
@@ -2378,86 +2806,165 @@ status_t AudioFlinger::RecordHandle::onTransact(
// ----------------------------------------------------------------------------
-AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware,
- const sp<AudioFlinger>& audioFlinger) :
- mAudioHardware(audioHardware),
- mAudioFlinger(audioFlinger),
- mActive(false)
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels) :
+ ThreadBase(audioFlinger),
+ mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
{
+ mReqChannelCount = AudioSystem::popCount(channels);
+ mReqSampleRate = sampleRate;
+ readInputParameters();
+ sendConfigEvent(AudioSystem::INPUT_OPENED);
}
-AudioFlinger::AudioRecordThread::~AudioRecordThread()
+
+AudioFlinger::RecordThread::~RecordThread()
{
+ mAudioFlinger->mAudioHardware->closeInputStream(mInput);
+ delete[] mRsmpInBuffer;
+ if (mResampler != 0) {
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
+ }
}
-bool AudioFlinger::AudioRecordThread::threadLoop()
+void AudioFlinger::RecordThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Record Thread %p", this);
+
+ run(buffer, PRIORITY_URGENT_AUDIO);
+}
+bool AudioFlinger::RecordThread::threadLoop()
{
- LOGV("AudioRecordThread: start record loop");
AudioBufferProvider::Buffer buffer;
- int inBufferSize = 0;
- int inFrameCount = 0;
- AudioStreamIn* input = 0;
+ sp<RecordTrack> activeTrack;
- mActive = 0;
-
// start recording
while (!exitPending()) {
- if (!mActive) {
- mLock.lock();
- if (!mActive && !exitPending()) {
- LOGV("AudioRecordThread: loop stopping");
- if (input) {
- delete input;
- input = 0;
+
+ processConfigEvents();
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ checkForNewParameters_l();
+ if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+ if (!mStandby) {
+ mInput->standby();
+ mStandby = true;
}
- mRecordTrack.clear();
- mStopped.signal();
+ if (exitPending()) break;
+
+ LOGV("RecordThread: loop stopping");
+ // go to sleep
mWaitWorkCV.wait(mLock);
-
- LOGV("AudioRecordThread: loop starting");
- if (mRecordTrack != 0) {
- input = mAudioHardware->openInputStream(
- mRecordTrack->inputSource(),
- mRecordTrack->format(),
- mRecordTrack->channelCount(),
- mRecordTrack->sampleRate(),
- &mStartStatus,
- (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
- if (input != 0) {
- inBufferSize = input->bufferSize();
- inFrameCount = inBufferSize/input->frameSize();
+ LOGV("RecordThread: loop starting");
+ continue;
+ }
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState == TrackBase::PAUSING) {
+ mActiveTrack.clear();
+ mStartStopCond.broadcast();
+ } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+ mRsmpInIndex = mFrameCount;
+ if (mReqChannelCount != mActiveTrack->channelCount()) {
+ mActiveTrack.clear();
+ } else {
+ mActiveTrack->mState == TrackBase::ACTIVE;
}
- } else {
- mStartStatus = NO_INIT;
+ mStartStopCond.broadcast();
}
- if (mStartStatus !=NO_ERROR) {
- LOGW("record start failed, status %d", mStartStatus);
- mActive = false;
- mRecordTrack.clear();
- }
- mWaitWorkCV.signal();
+ mStandby = false;
}
- mLock.unlock();
- } else if (mRecordTrack != 0) {
-
- buffer.frameCount = inFrameCount;
- if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR &&
- (int)buffer.frameCount == inFrameCount)) {
- LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
- ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
- if (bytesRead < 0) {
- LOGE("Error reading audio input");
- sleep(1);
+ }
+
+ if (mActiveTrack != 0) {
+ buffer.frameCount = mFrameCount;
+ if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ size_t framesOut = buffer.frameCount;
+ if (mResampler == 0) {
+ // no resampling
+ while (framesOut) {
+ size_t framesIn = mFrameCount - mRsmpInIndex;
+ if (framesIn) {
+ int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+ int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
+ if (framesIn > framesOut)
+ framesIn = framesOut;
+ mRsmpInIndex += framesIn;
+ framesOut -= framesIn;
+ if (mChannelCount == mReqChannelCount ||
+ mFormat != AudioSystem::PCM_16_BIT) {
+ memcpy(dst, src, framesIn * mFrameSize);
+ } else {
+ int16_t *src16 = (int16_t *)src;
+ int16_t *dst16 = (int16_t *)dst;
+ if (mChannelCount == 1) {
+ while (framesIn--) {
+ *dst16++ = *src16;
+ *dst16++ = *src16++;
+ }
+ } else {
+ while (framesIn--) {
+ *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
+ src16 += 2;
+ }
+ }
+ }
+ }
+ if (framesOut && mFrameCount == mRsmpInIndex) {
+ ssize_t bytesRead;
+ if (framesOut == mFrameCount &&
+ (mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
+ bytesRead = mInput->read(buffer.raw, mInputBytes);
+ framesOut = 0;
+ } else {
+ bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ mRsmpInIndex = 0;
+ }
+ if (bytesRead < 0) {
+ LOGE("Error reading audio input");
+ sleep(1);
+ mRsmpInIndex = mFrameCount;
+ framesOut = 0;
+ buffer.frameCount = 0;
+ }
+ }
+ }
+ } else {
+ // resampling
+
+ memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+ // alter output frame count as if we were expecting stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ framesOut >>= 1;
+ }
+ mResampler->resample(mRsmpOutBuffer, framesOut, this);
+ // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
+ // are 32 bit aligned which should be always true.
+ if (mChannelCount == 2 && mReqChannelCount == 1) {
+ AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ // the resampler always outputs stereo samples: do post stereo to mono conversion
+ int16_t *src = (int16_t *)mRsmpOutBuffer;
+ int16_t *dst = buffer.i16;
+ while (framesOut--) {
+ *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
+ src += 2;
+ }
+ } else {
+ AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+
}
- mRecordTrack->releaseBuffer(&buffer);
- mRecordTrack->overflow();
+ mActiveTrack->releaseBuffer(&buffer);
+ mActiveTrack->overflow();
}
-
// client isn't retrieving buffers fast enough
else {
- if (!mRecordTrack->setOverflow())
- LOGW("AudioRecordThread: buffer overflow");
+ if (!mActiveTrack->setOverflow())
+ LOGW("RecordThread: buffer overflow");
// Release the processor for a while before asking for a new buffer.
// This will give the application more chance to read from the buffer and
// clear the overflow.
@@ -2466,65 +2973,64 @@ bool AudioFlinger::AudioRecordThread::threadLoop()
}
}
-
- if (input) {
- delete input;
+ if (!mStandby) {
+ mInput->standby();
}
- mRecordTrack.clear();
-
+ mActiveTrack.clear();
+
+ sendConfigEvent(AudioSystem::INPUT_CLOSED);
+ processConfigEvents();
+
+ LOGV("RecordThread %p exiting", this);
return false;
}
-status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
{
- LOGV("AudioRecordThread::start");
+ LOGV("RecordThread::start");
AutoMutex lock(&mLock);
- mActive = true;
- // If starting the active track, just reset mActive in case a stop
- // was pending and exit
- if (recordTrack == mRecordTrack.get()) return NO_ERROR;
- if (mRecordTrack != 0) return -EBUSY;
+ if (mActiveTrack != 0) {
+ if (recordTrack != mActiveTrack.get()) return -EBUSY;
- mRecordTrack = recordTrack;
+ if (mActiveTrack->mState == TrackBase::PAUSING) mActiveTrack->mState = TrackBase::RESUMING;
+ return NO_ERROR;
+ }
+
+ mActiveTrack = recordTrack;
+ mActiveTrack->mState = TrackBase::RESUMING;
// signal thread to start
LOGV("Signal record thread");
mWaitWorkCV.signal();
- mWaitWorkCV.wait(mLock);
- LOGV("Record started, status %d", mStartStatus);
- return mStartStatus;
-}
-
-void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
- LOGV("AudioRecordThread::stop");
- AutoMutex lock(&mLock);
- if (mActive && (recordTrack == mRecordTrack.get())) {
- mActive = false;
- mStopped.wait(mLock);
+ mStartStopCond.wait(mLock);
+ if (mActiveTrack != 0) {
+ LOGV("Record started OK");
+ return NO_ERROR;
+ } else {
+ LOGV("Record failed to start");
+ return BAD_VALUE;
}
}
-void AudioFlinger::AudioRecordThread::exit()
-{
- LOGV("AudioRecordThread::exit");
- {
- AutoMutex lock(&mLock);
- requestExit();
- mWaitWorkCV.signal();
+void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+ LOGV("RecordThread::stop");
+ AutoMutex lock(&mLock);
+ if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
+ mActiveTrack->mState = TrackBase::PAUSING;
+ mStartStopCond.wait(mLock);
}
- requestExitAndWait();
}
-status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
+status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
pid_t pid = 0;
- if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
- snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
+ if (mActiveTrack != 0 && mActiveTrack->mClient != 0) {
+ snprintf(buffer, SIZE, "Record client pid: %d\n", mActiveTrack->mClient->pid());
result.append(buffer);
} else {
result.append("No record client\n");
@@ -2533,6 +3039,463 @@ status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& a
return NO_ERROR;
}
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ size_t framesReq = buffer->frameCount;
+ size_t framesReady = mFrameCount - mRsmpInIndex;
+ int channelCount;
+
+ if (framesReady == 0) {
+ ssize_t bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ if (bytesRead < 0) {
+ LOGE("RecordThread::getNextBuffer() Error reading audio input");
+ sleep(1);
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+ mRsmpInIndex = 0;
+ framesReady = mFrameCount;
+ }
+
+ if (framesReq > framesReady) {
+ framesReq = framesReady;
+ }
+
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ mRsmpInIndex += buffer->frameCount;
+ buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+ bool reconfig = false;
+
+ if (mNewParameters != "") {
+ status_t status = NO_ERROR;
+ AudioParameter param = AudioParameter(mNewParameters);
+ int value;
+ int reqFormat = mFormat;
+ int reqSamplingRate = mReqSampleRate;
+ int reqChannelCount = mReqChannelCount;
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reqSamplingRate = value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ reqFormat = value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ reqChannelCount = AudioSystem::popCount(value);
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (mActiveTrack != 0) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mInput->setParameters(mNewParameters);
+ if (status == INVALID_OPERATION) {
+ mInput->standby();
+ status = mInput->setParameters(mNewParameters);
+ }
+ if (reconfig) {
+ if (status == BAD_VALUE &&
+ reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
+ ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+ status = NO_ERROR;
+ }
+ if (status == NO_ERROR) {
+ readInputParameters();
+ sendConfigEvent(AudioSystem::INPUT_CONFIG_CHANGED);
+ }
+ }
+ }
+ mNewParameters = "";
+ mParamStatus = status;
+ mParamCond.signal();
+ }
+ return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+ return mInput->getParameters(keys);
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = 0;
+
+ switch (event) {
+ case AudioSystem::INPUT_OPENED:
+ case AudioSystem::INPUT_CONFIG_CHANGED:
+ desc.channels = mChannelCount;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = 0;
+ param2 = &desc;
+ break;
+
+ case AudioSystem::INPUT_CLOSED:
+ default:
+ break;
+ }
+ mAudioFlinger->audioConfigChanged(event, this, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+ if (mRsmpInBuffer) delete mRsmpInBuffer;
+ if (mRsmpOutBuffer) delete mRsmpOutBuffer;
+ if (mResampler) delete mResampler;
+ mResampler = 0;
+
+ mSampleRate = mInput->sampleRate();
+ mChannelCount = AudioSystem::popCount(mInput->channels());
+ mFormat = mInput->format();
+ mFrameSize = mInput->frameSize();
+ mInputBytes = mInput->bufferSize();
+ mFrameCount = mInputBytes / mFrameSize;
+ mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+ if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+ {
+ int channelCount;
+ // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+ // stereo to mono post process as the resampler always outputs stereo.
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ mResampler->setSampleRate(mSampleRate);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+ mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+ // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ mFrameCount >>= 1;
+ }
+
+ }
+ mRsmpInIndex = mFrameCount;
+}
+
+// ----------------------------------------------------------------------------
+
+void *AudioFlinger::openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ uint32_t flags)
+{
+ status_t status;
+ PlaybackThread *thread = NULL;
+ mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+ uint32_t format = pFormat ? *pFormat : 0;
+ uint32_t channels = pChannels ? *pChannels : 0;
+ uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+
+ LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
+ pDevices ? *pDevices : 0,
+ samplingRate,
+ format,
+ channels,
+ flags);
+
+ if (pDevices == NULL || *pDevices == 0) {
+ return NULL;
+ }
+ Mutex::Autolock _l(mLock);
+
+ AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status);
+ LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+ output,
+ samplingRate,
+ format,
+ channels,
+ status);
+
+ mHardwareStatus = AUDIO_HW_IDLE;
+ if (output != 0) {
+ if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format != AudioSystem::PCM_16_BIT) ||
+ (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+ thread = new DirectOutputThread(this, output);
+ LOGV("openOutput() created direct output %p", thread);
+ } else {
+ thread = new MixerThread(this, output);
+ LOGV("openOutput() created mixer output %p", thread);
+ }
+ mPlaybackThreads.add(thread);
+
+ if (pSamplingRate) *pSamplingRate = samplingRate;
+ if (pFormat) *pFormat = format;
+ if (pChannels) *pChannels = channels;
+ if (pLatencyMs) *pLatencyMs = thread->latency();
+ }
+
+ return thread;
+}
+
+void *AudioFlinger::openDuplicateOutput(void *output1, void *output2)
+{
+ Mutex::Autolock _l(mLock);
+
+ if (checkMixerThread_l(output1) == NULL ||
+ checkMixerThread_l(output2) == NULL) {
+ LOGW("openDuplicateOutput() wrong output mixer type %p or %p", output1, output2);
+ return NULL;
+ }
+
+ DuplicatingThread *thread = new DuplicatingThread(this, (MixerThread *)output1);
+ thread->addOutputTrack( (MixerThread *)output2);
+ mPlaybackThreads.add(thread);
+ return thread;
+}
+
+status_t AudioFlinger::closeOutput(void *output)
+{
+ PlaybackThread *thread;
+ {
+ Mutex::Autolock _l(mLock);
+ thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("closeOutput() %p", thread);
+
+ if (thread->type() == PlaybackThread::MIXER) {
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads[i]->type() == PlaybackThread::DUPLICATING) {
+ DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads[i].get();
+ dupThread->removeOutputTrack((MixerThread *)thread);
+ }
+ }
+ }
+ mPlaybackThreads.remove(thread);
+ }
+ thread->exit();
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::suspendOutput(void *output)
+{
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("suspendOutput() %p", output);
+ thread->suspend();
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::restoreOutput(void *output)
+{
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("restoreOutput() %p", output);
+
+ thread->restore();
+
+ return NO_ERROR;
+}
+
+void *AudioFlinger::openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics)
+{
+ status_t status;
+ RecordThread *thread = NULL;
+ uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+ uint32_t format = pFormat ? *pFormat : 0;
+ uint32_t channels = pChannels ? *pChannels : 0;
+ uint32_t reqSamplingRate = samplingRate;
+ uint32_t reqFormat = format;
+ uint32_t reqChannels = channels;
+
+ if (pDevices == NULL || *pDevices == 0) {
+ return NULL;
+ }
+ Mutex::Autolock _l(mLock);
+
+ AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
+ input,
+ samplingRate,
+ format,
+ channels,
+ acoustics,
+ status);
+
+ // 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. The AudioFlinger can resample the input and do mono to stereo
+ // or stereo to mono conversions on 16 bit PCM inputs.
+ if (input == 0 && status == BAD_VALUE &&
+ reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+ (samplingRate <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+ LOGV("openInput() reopening with proposed sampling rate and channels");
+ input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ }
+
+ if (input != 0) {
+ // Start record thread
+ thread = new RecordThread(this, input, reqSamplingRate, reqChannels);
+ mRecordThreads.add(thread);
+
+ if (pSamplingRate) *pSamplingRate = reqSamplingRate;
+ if (pFormat) *pFormat = format;
+ if (pChannels) *pChannels = reqChannels;
+
+ input->standby();
+ }
+
+ return thread;
+}
+
+status_t AudioFlinger::closeInput(void *input)
+{
+ RecordThread *thread;
+ {
+ Mutex::Autolock _l(mLock);
+ thread = checkRecordThread_l(input);
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("closeInput() %p", thread);
+ mRecordThreads.remove(thread);
+ }
+ thread->exit();
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::setStreamOutput(uint32_t stream, void *output)
+{
+ Mutex::Autolock _l(mLock);
+ MixerThread *dstThread = checkMixerThread_l(output);
+ if (dstThread == NULL) {
+ LOGW("setStreamOutput() bad output thread %p", output);
+ return BAD_VALUE;
+ }
+
+ LOGV("setStreamOutput() stream %d to output %p", stream, dstThread);
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ PlaybackThread *thread = mPlaybackThreads[i].get();
+ if (thread != dstThread &&
+ thread->type() != PlaybackThread::DIRECT) {
+ MixerThread *srcThread = (MixerThread *)thread;
+ SortedVector < sp<MixerThread::Track> > tracks;
+ SortedVector < wp<MixerThread::Track> > activeTracks;
+ srcThread->getTracks(tracks, activeTracks, stream);
+ if (tracks.size()) {
+ dstThread->putTracks(tracks, activeTracks);
+ dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
+ }
+ }
+ }
+
+ return NO_ERROR;
+}
+
+// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(void *output) const
+{
+ PlaybackThread *thread = NULL;
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads[i] == output) {
+ thread = (PlaybackThread *)output;
+ break;
+ }
+ }
+
+ return thread;
+}
+
+// checkMixerThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(void *output) const
+{
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread != NULL) {
+ if (thread->type() == PlaybackThread::DIRECT) {
+ thread = NULL;
+ }
+ }
+ return (MixerThread *)thread;
+}
+
+// checkRecordThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(void *input) const
+{
+ RecordThread *thread = NULL;
+
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ if (mRecordThreads[i] == input) {
+ thread = (RecordThread *)input;
+ break;
+ }
+ }
+
+ return thread;
+}
+
+// ----------------------------------------------------------------------------
+
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -2540,6 +3503,7 @@ status_t AudioFlinger::onTransact(
}
// ----------------------------------------------------------------------------
+
void AudioFlinger::instantiate() {
defaultServiceManager()->addService(
String16("media.audio_flinger"), new AudioFlinger());
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index 3531a58..7d78749 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -31,7 +31,6 @@
#include <utils/Errors.h>
#include <utils/threads.h>
#include <binder/MemoryDealer.h>
-#include <utils/KeyedVector.h>
#include <utils/SortedVector.h>
#include <utils/Vector.h>
@@ -44,6 +43,7 @@ namespace android {
class audio_track_cblk_t;
class AudioMixer;
class AudioBuffer;
+class AudioResampler;
// ----------------------------------------------------------------------------
@@ -56,7 +56,7 @@ class AudioBuffer;
static const nsecs_t kStandbyTimeInNsecs = seconds(3);
-class AudioFlinger : public BnAudioFlinger, public IBinder::DeathRecipient
+class AudioFlinger : public BnAudioFlinger, public IBinder::DeathRecipient
{
public:
static void instantiate();
@@ -73,13 +73,14 @@ public:
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
+ void *output,
status_t *status);
- virtual uint32_t sampleRate(int output) const;
- virtual int channelCount(int output) const;
- virtual int format(int output) const;
- virtual size_t frameCount(int output) const;
- virtual uint32_t latency(int output) const;
+ virtual uint32_t sampleRate(void *output) const;
+ virtual int channelCount(void *output) const;
+ virtual int format(void *output) const;
+ virtual size_t frameCount(void *output) const;
+ virtual uint32_t latency(void *output) const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
@@ -87,33 +88,51 @@ public:
virtual float masterVolume() const;
virtual bool masterMute() const;
- virtual status_t setStreamVolume(int stream, float value);
+ virtual status_t setStreamVolume(int stream, float value, void *output);
virtual status_t setStreamMute(int stream, bool muted);
- virtual float streamVolume(int stream) const;
+ virtual float streamVolume(int stream, void *output) const;
virtual bool streamMute(int stream) const;
- virtual status_t setRouting(int mode, uint32_t routes, uint32_t mask);
- virtual uint32_t getRouting(int mode) const;
-
virtual status_t setMode(int mode);
- virtual int getMode() const;
virtual status_t setMicMute(bool state);
virtual bool getMicMute() const;
virtual bool isMusicActive() const;
- virtual bool isA2dpEnabled() const;
-
- virtual status_t setParameter(const char* key, const char* value);
+ virtual status_t setParameters(void *ioHandle, const String8& keyValuePairs);
+ virtual String8 getParameters(void *ioHandle, const String8& keys);
virtual void registerClient(const sp<IAudioFlingerClient>& client);
-
+
virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-
- virtual void wakeUp() { mWaitWorkCV.broadcast(); }
-
+
+ virtual void *openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ uint32_t flags);
+
+ virtual void *openDuplicateOutput(void *output1, void *output2);
+
+ virtual status_t closeOutput(void *output);
+
+ virtual status_t suspendOutput(void *output);
+
+ virtual status_t restoreOutput(void *output);
+
+ virtual void *openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics);
+
+ virtual status_t closeInput(void *input);
+
+ virtual status_t setStreamOutput(uint32_t stream, void *output);
+
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
@@ -139,7 +158,7 @@ public:
// record interface
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int inputSource,
+ void *input,
uint32_t sampleRate,
int format,
int channelCount,
@@ -153,30 +172,12 @@ public:
Parcel* reply,
uint32_t flags);
+ void audioConfigChanged(int event, void *param1, void *param2);
+
private:
AudioFlinger();
virtual ~AudioFlinger();
-
- void setOutput(int outputType);
- void doSetOutput(int outputType);
-
-#ifdef WITH_A2DP
- void setA2dpEnabled_l(bool enable);
- void checkA2dpEnabledChange_l();
-#endif
- static bool streamForcedToSpeaker(int streamType);
-
- // Management of forced route to speaker for certain track types.
- enum force_speaker_command {
- ACTIVE_TRACK_ADDED = 0,
- ACTIVE_TRACK_REMOVED,
- CHECK_ROUTE_RESTORE_TIME,
- FORCE_ROUTE_RESTORE
- };
- void handleForcedSpeakerRoute(int command);
-#ifdef WITH_A2DP
- void handleRouteDisablesA2dp_l(int routes);
-#endif
+
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
@@ -201,14 +202,17 @@ private:
class TrackHandle;
class RecordHandle;
- class AudioRecordThread;
-
-
- // --- MixerThread ---
- class MixerThread : public Thread {
+ class RecordThread;
+ class PlaybackThread;
+ class MixerThread;
+ class DirectOutputThread;
+ class Track;
+ class RecordTrack;
+
+ class ThreadBase : public Thread {
public:
-
- // --- Track ---
+ ThreadBase (const sp<AudioFlinger>& audioFlinger);
+ virtual ~ThreadBase();
// base for record and playback
class TrackBase : public AudioBufferProvider, public RefBase {
@@ -230,7 +234,7 @@ private:
// The upper 16 bits are used for track-specific flags.
};
- TrackBase(const sp<MixerThread>& mixerThread,
+ TrackBase(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
@@ -245,9 +249,8 @@ private:
sp<IMemory> getCblk() const;
protected:
- friend class MixerThread;
+ friend class ThreadBase;
friend class RecordHandle;
- friend class AudioRecordThread;
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
@@ -269,10 +272,6 @@ private:
void* getBuffer(uint32_t offset, uint32_t frames) const;
- int name() const {
- return mName;
- }
-
bool isStopped() const {
return mState == STOPPED;
}
@@ -284,14 +283,13 @@ private:
bool step();
void reset();
- sp<MixerThread> mMixerThread;
+ wp<ThreadBase> mThread;
sp<Client> mClient;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
- int mName;
// we don't really need a lock for these
int mState;
int mClientTid;
@@ -299,10 +297,67 @@ private:
uint32_t mFlags;
};
+ class ConfigEvent {
+ public:
+ ConfigEvent() : mEvent(0), mParam(0) {}
+
+ int mEvent;
+ int mParam;
+ };
+
+ uint32_t sampleRate() const;
+ int channelCount() const;
+ int format() const;
+ size_t frameCount() const;
+ void wakeUp() { mWaitWorkCV.broadcast(); }
+ void exit();
+ virtual bool checkForNewParameters_l() = 0;
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys) = 0;
+ virtual void audioConfigChanged(int event, int param = 0) = 0;
+ void sendConfigEvent(int event, int param = 0);
+ void processConfigEvents();
+
+ protected:
+
+ friend class Track;
+ friend class TrackBase;
+ friend class PlaybackThread;
+ friend class MixerThread;
+ friend class DirectOutputThread;
+ friend class DuplicatingThread;
+ friend class RecordThread;
+ friend class RecordTrack;
+
+ mutable Mutex mLock;
+ Condition mWaitWorkCV;
+ sp<AudioFlinger> mAudioFlinger;
+ uint32_t mSampleRate;
+ size_t mFrameCount;
+ int mChannelCount;
+ int mFormat;
+ uint32_t mFrameSize;
+ Condition mParamCond;
+ String8 mNewParameters;
+ status_t mParamStatus;
+ Vector<ConfigEvent *> mConfigEvents;
+ bool mStandby;
+ };
+
+ // --- PlaybackThread ---
+ class PlaybackThread : public ThreadBase {
+ public:
+
+ enum type {
+ MIXER,
+ DIRECT,
+ DUPLICATING
+ };
+
// playback track
class Track : public TrackBase {
public:
- Track( const sp<MixerThread>& mixerThread,
+ Track( const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
@@ -321,6 +376,9 @@ private:
void destroy();
void mute(bool);
void setVolume(float left, float right);
+ int name() const {
+ return mName;
+ }
int type() const {
return mStreamType;
@@ -328,7 +386,7 @@ private:
protected:
- friend class MixerThread;
+ friend class ThreadBase;
friend class AudioFlinger;
friend class AudioFlinger::TrackHandle;
@@ -336,21 +394,14 @@ private:
Track& operator = (const Track&);
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-
- bool isMuted() const {
- return (mMute || mMixerThread->mStreamTypes[mStreamType].mute);
- }
-
+ bool isMuted() { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
-
bool isPaused() const {
return mState == PAUSED;
}
-
bool isReady() const;
-
void setPaused() { mState = PAUSED; }
void reset();
@@ -364,54 +415,20 @@ private:
sp<IMemory> mSharedBuffer;
bool mResetDone;
int mStreamType;
+ int mName;
}; // end of Track
- // record track
- class RecordTrack : public TrackBase {
- public:
- RecordTrack(const sp<MixerThread>& mixerThread,
- const sp<Client>& client,
- int inputSource,
- uint32_t sampleRate,
- int format,
- int channelCount,
- int frameCount,
- uint32_t flags);
- ~RecordTrack();
-
- virtual status_t start();
- virtual void stop();
-
- bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
- bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
-
- int inputSource() const { return mInputSource; }
-
- private:
- friend class AudioFlinger;
- friend class AudioFlinger::RecordHandle;
- friend class AudioFlinger::AudioRecordThread;
- friend class MixerThread;
-
- RecordTrack(const Track&);
- RecordTrack& operator = (const Track&);
-
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-
- bool mOverflow;
- int mInputSource;
- };
// playback track
class OutputTrack : public Track {
public:
-
+
class Buffer: public AudioBufferProvider::Buffer {
public:
int16_t *mBuffer;
};
-
- OutputTrack( const sp<MixerThread>& mixerThread,
+
+ OutputTrack( const wp<ThreadBase>& thread,
uint32_t sampleRate,
int format,
int channelCount,
@@ -420,35 +437,35 @@ private:
virtual status_t start();
virtual void stop();
- void write(int16_t* data, uint32_t frames);
+ bool write(int16_t* data, uint32_t frames);
bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
+ bool isActive() { return mActive; }
+ wp<ThreadBase>& thread() { return mThread; }
private:
- status_t obtainBuffer(AudioBufferProvider::Buffer* buffer);
+ status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
void clearBufferQueue();
-
- sp<MixerThread> mOutputMixerThread;
+
+ // Maximum number of pending buffers allocated by OutputTrack::write()
+ static const uint8_t kMaxOverFlowBuffers = 3;
+
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
- uint32_t mFramesWritten;
-
- }; // end of OutputTrack
+ uint32_t mWaitTimeMs;
+ bool mActive;
- MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType);
- virtual ~MixerThread();
+ }; // end of OutputTrack
+
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+ virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
// Thread virtuals
- virtual bool threadLoop();
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual uint32_t sampleRate() const;
- virtual int channelCount() const;
- virtual int format() const;
- virtual size_t frameCount() const;
virtual uint32_t latency() const;
virtual status_t setMasterVolume(float value);
@@ -463,9 +480,8 @@ private:
virtual float streamVolume(int stream) const;
virtual bool streamMute(int stream) const;
- bool isMusicActive_l() const;
-
-
+ bool isMusicActive() const;
+
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
@@ -475,13 +491,15 @@ private:
int frameCount,
const sp<IMemory>& sharedBuffer,
status_t *status);
-
- void getTracks_l(SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks);
- void putTracks_l(SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks);
- void setOuputTrack(OutputTrack *track) { mOutputTrack = track; }
-
+
+ AudioStreamOut* getOutput() { return mOutput; }
+
+ virtual int type() const { return mType; }
+ void suspend() { mSuspended = true; }
+ void restore() { mSuspended = false; }
+ virtual String8 getParameters(const String8& keys);
+ virtual void audioConfigChanged(int event, int param = 0);
+
struct stream_type_t {
stream_type_t()
: volume(1.0f),
@@ -494,54 +512,109 @@ private:
private:
-
friend class AudioFlinger;
friend class Track;
friend class TrackBase;
- friend class RecordTrack;
-
- MixerThread(const Client&);
- MixerThread& operator = (const MixerThread&);
-
+ friend class MixerThread;
+ friend class DirectOutputThread;
+ friend class DuplicatingThread;
+
+ PlaybackThread(const Client&);
+ PlaybackThread& operator = (const PlaybackThread&);
+
status_t addTrack_l(const sp<Track>& track);
void destroyTrack_l(const sp<Track>& track);
- int getTrackName_l();
- void deleteTrackName_l(int name);
- void addActiveTrack_l(const wp<Track>& t);
- void removeActiveTrack_l(const wp<Track>& t);
- size_t getOutputFrameCount();
+ virtual int getTrackName_l() = 0;
+ virtual void deleteTrackName_l(int name) = 0;
+ void readOutputParameters();
- status_t dumpInternals(int fd, const Vector<String16>& args);
+ virtual status_t dumpInternals(int fd, const Vector<String16>& args);
status_t dumpTracks(int fd, const Vector<String16>& args);
-
- sp<AudioFlinger> mAudioFlinger;
+
SortedVector< wp<Track> > mActiveTracks;
SortedVector< sp<Track> > mTracks;
- stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
- AudioMixer* mAudioMixer;
+ // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
+ stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES + 1];
AudioStreamOut* mOutput;
- int mOutputType;
- uint32_t mSampleRate;
- size_t mFrameCount;
- int mChannelCount;
- int mFormat;
- int16_t* mMixBuffer;
float mMasterVolume;
bool mMasterMute;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
- bool mStandby;
bool mInWrite;
- sp <OutputTrack> mOutputTrack;
+ int16_t* mMixBuffer;
+ bool mSuspended;
+ int mType;
+ int mBytesWritten;
+ int mMinBytesToWrite;
+ };
+
+ class MixerThread : public PlaybackThread {
+ public:
+ MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+ virtual ~MixerThread();
+
+ // Thread virtuals
+ virtual bool threadLoop();
+
+ void getTracks(SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks,
+ int streamType);
+ void putTracks(SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks);
+ virtual int getTrackName_l();
+ virtual void deleteTrackName_l(int name);
+ virtual bool checkForNewParameters_l();
+ virtual status_t dumpInternals(int fd, const Vector<String16>& args);
+
+ protected:
+ size_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove);
+
+ AudioMixer* mAudioMixer;
+ };
+
+ class DirectOutputThread : public PlaybackThread {
+ public:
+
+ DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+ ~DirectOutputThread();
+
+ // Thread virtuals
+ virtual bool threadLoop();
+
+ virtual int getTrackName_l();
+ virtual void deleteTrackName_l(int name);
+ virtual bool checkForNewParameters_l();
+
+ private:
+ float mLeftVolume;
+ float mRightVolume;
};
-
+ class DuplicatingThread : public MixerThread {
+ public:
+ DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread);
+ ~DuplicatingThread();
+
+ // Thread virtuals
+ virtual bool threadLoop();
+ void addOutputTrack(MixerThread* thread);
+ void removeOutputTrack(MixerThread* thread);
+
+ private:
+ SortedVector < sp<OutputTrack> > mOutputTracks;
+ };
+
+ PlaybackThread *checkPlaybackThread_l(void *output) const;
+ MixerThread *checkMixerThread_l(void *output) const;
+ RecordThread *checkRecordThread_l(void *input) const;
+ float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
+
friend class AudioBuffer;
class TrackHandle : public android::BnAudioTrack {
public:
- TrackHandle(const sp<MixerThread::Track>& track);
+ TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
virtual status_t start();
virtual void stop();
@@ -553,20 +626,90 @@ private:
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<MixerThread::Track> mTrack;
+ sp<PlaybackThread::Track> mTrack;
};
friend class Client;
- friend class MixerThread::Track;
+ friend class PlaybackThread::Track;
void removeClient(pid_t pid);
+ // record thread
+ class RecordThread : public ThreadBase, public AudioBufferProvider
+ {
+ public:
+
+ // record track
+ class RecordTrack : public TrackBase {
+ public:
+ RecordTrack(const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags);
+ ~RecordTrack();
+
+ virtual status_t start();
+ virtual void stop();
+
+ bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
+ bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
+
+ private:
+ friend class AudioFlinger;
+
+ RecordTrack(const RecordTrack&);
+ RecordTrack& operator = (const RecordTrack&);
+
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+
+ bool mOverflow;
+ };
+
+
+ RecordThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamIn *input,
+ uint32_t sampleRate,
+ uint32_t channels);
+ ~RecordThread();
+
+ virtual bool threadLoop();
+ virtual status_t readyToRun() { return NO_ERROR; }
+ virtual void onFirstRef();
+
+ status_t start(RecordTrack* recordTrack);
+ void stop(RecordTrack* recordTrack);
+ status_t dump(int fd, const Vector<String16>& args);
+ AudioStreamIn* getInput() { return mInput; }
+
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual bool checkForNewParameters_l();
+ virtual String8 getParameters(const String8& keys);
+ virtual void audioConfigChanged(int event, int param = 0);
+ void readInputParameters();
+
+ private:
+ RecordThread();
+ AudioStreamIn *mInput;
+ sp<RecordTrack> mActiveTrack;
+ Condition mStartStopCond;
+ AudioResampler *mResampler;
+ int32_t *mRsmpOutBuffer;
+ int16_t *mRsmpInBuffer;
+ size_t mRsmpInIndex;
+ size_t mInputBytes;
+ int mReqChannelCount;
+ uint32_t mReqSampleRate;
+ };
class RecordHandle : public android::BnAudioRecord {
public:
- RecordHandle(const sp<MixerThread::RecordTrack>& recordTrack);
+ RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
virtual status_t start();
virtual void stop();
@@ -574,66 +717,30 @@ private:
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<MixerThread::RecordTrack> mRecordTrack;
+ sp<RecordThread::RecordTrack> mRecordTrack;
};
- // record thread
- class AudioRecordThread : public Thread
- {
- public:
- AudioRecordThread(AudioHardwareInterface* audioHardware, const sp<AudioFlinger>& audioFlinger);
- virtual ~AudioRecordThread();
- virtual bool threadLoop();
- virtual status_t readyToRun() { return NO_ERROR; }
- virtual void onFirstRef() {}
+ friend class RecordThread;
+ friend class PlaybackThread;
- status_t start(MixerThread::RecordTrack* recordTrack);
- void stop(MixerThread::RecordTrack* recordTrack);
- void exit();
- status_t dump(int fd, const Vector<String16>& args);
-
- private:
- AudioRecordThread();
- AudioHardwareInterface *mAudioHardware;
- sp<AudioFlinger> mAudioFlinger;
- sp<MixerThread::RecordTrack> mRecordTrack;
- Mutex mLock;
- Condition mWaitWorkCV;
- Condition mStopped;
- volatile bool mActive;
- status_t mStartStatus;
- };
-
- friend class AudioRecordThread;
- friend class MixerThread;
- status_t startRecord(MixerThread::RecordTrack* recordTrack);
- void stopRecord(MixerThread::RecordTrack* recordTrack);
-
- mutable Mutex mHardwareLock;
mutable Mutex mLock;
- mutable Condition mWaitWorkCV;
DefaultKeyedVector< pid_t, wp<Client> > mClients;
- sp<MixerThread> mA2dpMixerThread;
- sp<MixerThread> mHardwareMixerThread;
+ mutable Mutex mHardwareLock;
AudioHardwareInterface* mAudioHardware;
- AudioHardwareInterface* mA2dpAudioInterface;
- sp<AudioRecordThread> mAudioRecordThread;
- bool mA2dpEnabled;
- bool mNotifyA2dpChange;
mutable int mHardwareStatus;
- SortedVector< wp<IBinder> > mNotificationClients;
- int mForcedSpeakerCount;
- int mA2dpDisableCount;
-
- // true if A2DP should resume when mA2dpDisableCount returns to zero
- bool mA2dpSuppressed;
- uint32_t mSavedRoute;
- uint32_t mForcedRoute;
- nsecs_t mRouteRestoreTime;
- bool mMusicMuteSaved;
+
+
+ SortedVector< sp<PlaybackThread> > mPlaybackThreads;
+ PlaybackThread::stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
+ float mMasterVolume;
+ bool mMasterMute;
+
+ SortedVector< sp<RecordThread> > mRecordThreads;
+
+ SortedVector< sp<IBinder> > mNotificationClients;
};
// ----------------------------------------------------------------------------
diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp
index 1e159b8..57874f3 100644
--- a/libs/audioflinger/AudioHardwareGeneric.cpp
+++ b/libs/audioflinger/AudioHardwareGeneric.cpp
@@ -49,8 +49,8 @@ AudioHardwareGeneric::AudioHardwareGeneric()
AudioHardwareGeneric::~AudioHardwareGeneric()
{
if (mFd >= 0) ::close(mFd);
- delete mOutput;
- delete mInput;
+ closeOutputStream((AudioStreamOut *)mOutput);
+ closeInputStream((AudioStreamIn *)mInput);
}
status_t AudioHardwareGeneric::initCheck()
@@ -63,7 +63,7 @@ status_t AudioHardwareGeneric::initCheck()
}
AudioStreamOut* AudioHardwareGeneric::openOutputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status)
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
{
AutoMutex lock(mLock);
@@ -77,7 +77,7 @@ AudioStreamOut* AudioHardwareGeneric::openOutputStream(
// create new output stream
AudioStreamOutGeneric* out = new AudioStreamOutGeneric();
- status_t lStatus = out->set(this, mFd, format, channelCount, sampleRate);
+ status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate);
if (status) {
*status = lStatus;
}
@@ -89,17 +89,19 @@ AudioStreamOut* AudioHardwareGeneric::openOutputStream(
return mOutput;
}
-void AudioHardwareGeneric::closeOutputStream(AudioStreamOutGeneric* out) {
- if (out == mOutput) mOutput = 0;
+void AudioHardwareGeneric::closeOutputStream(AudioStreamOut* out) {
+ if (mOutput && out == mOutput) {
+ delete mOutput;
+ mOutput = 0;
+ }
}
AudioStreamIn* AudioHardwareGeneric::openInputStream(
- int inputSource, int format, int channelCount, uint32_t sampleRate,
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
// check for valid input source
- if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
- (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+ if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
return 0;
}
@@ -115,7 +117,7 @@ AudioStreamIn* AudioHardwareGeneric::openInputStream(
// create new output stream
AudioStreamInGeneric* in = new AudioStreamInGeneric();
- status_t lStatus = in->set(this, mFd, format, channelCount, sampleRate, acoustics);
+ status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
if (status) {
*status = lStatus;
}
@@ -127,8 +129,11 @@ AudioStreamIn* AudioHardwareGeneric::openInputStream(
return mInput;
}
-void AudioHardwareGeneric::closeInputStream(AudioStreamInGeneric* in) {
- if (in == mInput) mInput = 0;
+void AudioHardwareGeneric::closeInputStream(AudioStreamIn* in) {
+ if (mInput && in == mInput) {
+ delete mInput;
+ mInput = 0;
+ }
}
status_t AudioHardwareGeneric::setVoiceVolume(float v)
@@ -185,30 +190,42 @@ status_t AudioHardwareGeneric::dump(int fd, const Vector<String16>& args)
status_t AudioStreamOutGeneric::set(
AudioHardwareGeneric *hw,
int fd,
- int format,
- int channels,
- uint32_t rate)
+ uint32_t devices,
+ int *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pRate)
{
+ int lFormat = pFormat ? *pFormat : 0;
+ uint32_t lChannels = pChannels ? *pChannels : 0;
+ uint32_t lRate = pRate ? *pRate : 0;
+
// fix up defaults
- if (format == 0) format = AudioSystem::PCM_16_BIT;
- if (channels == 0) channels = channelCount();
- if (rate == 0) rate = sampleRate();
+ if (lFormat == 0) lFormat = format();
+ if (lChannels == 0) lChannels = channels();
+ if (lRate == 0) lRate = sampleRate();
// check values
- if ((format != AudioSystem::PCM_16_BIT) ||
- (channels != channelCount()) ||
- (rate != sampleRate()))
+ if ((lFormat != format()) ||
+ (lChannels != channels()) ||
+ (lRate != sampleRate())) {
+ if (pFormat) *pFormat = format();
+ if (pChannels) *pChannels = channels();
+ if (pRate) *pRate = sampleRate();
return BAD_VALUE;
+ }
+
+ if (pFormat) *pFormat = lFormat;
+ if (pChannels) *pChannels = lChannels;
+ if (pRate) *pRate = lRate;
mAudioHardware = hw;
mFd = fd;
+ mDevice = devices;
return NO_ERROR;
}
AudioStreamOutGeneric::~AudioStreamOutGeneric()
{
- if (mAudioHardware)
- mAudioHardware->closeOutputStream(this);
}
ssize_t AudioStreamOutGeneric::write(const void* buffer, size_t bytes)
@@ -234,10 +251,12 @@ status_t AudioStreamOutGeneric::dump(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
result.append(buffer);
- snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+ snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
result.append(buffer);
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
+ snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
+ result.append(buffer);
snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
result.append(buffer);
snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
@@ -246,29 +265,68 @@ status_t AudioStreamOutGeneric::dump(int fd, const Vector<String16>& args)
return NO_ERROR;
}
+status_t AudioStreamOutGeneric::setParameters(const String8& keyValuePairs)
+{
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 key = String8(AudioParameter::keyRouting);
+ status_t status = NO_ERROR;
+ int device;
+ LOGV("setParameters() %s", keyValuePairs.string());
+
+ if (param.getInt(key, device) == NO_ERROR) {
+ mDevice = device;
+ param.remove(key);
+ }
+
+ if (param.size()) {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
+String8 AudioStreamOutGeneric::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ String8 value;
+ String8 key = String8(AudioParameter::keyRouting);
+
+ if (param.get(key, value) == NO_ERROR) {
+ param.addInt(key, (int)mDevice);
+ }
+
+ LOGV("getParameters() %s", param.toString().string());
+ return param.toString();
+}
+
// ----------------------------------------------------------------------------
// record functions
status_t AudioStreamInGeneric::set(
AudioHardwareGeneric *hw,
int fd,
- int format,
- int channels,
- uint32_t rate,
+ uint32_t devices,
+ int *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pRate,
AudioSystem::audio_in_acoustics acoustics)
{
// FIXME: remove logging
- LOGD("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, format, channels, rate);
+ if (pFormat == 0 || pChannels == 0 || pRate == 0) return BAD_VALUE;
+ LOGD("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, *pFormat, *pChannels, *pRate);
// check values
- if ((format != AudioSystem::PCM_16_BIT) ||
- (channels != channelCount()) ||
- (rate != sampleRate())) {
+ if ((*pFormat != format()) ||
+ (*pChannels != channels()) ||
+ (*pRate != sampleRate())) {
LOGE("Error opening input channel");
+ *pFormat = format();
+ *pChannels = channels();
+ *pRate = sampleRate();
return BAD_VALUE;
}
mAudioHardware = hw;
mFd = fd;
+ mDevice = devices;
return NO_ERROR;
}
@@ -276,14 +334,12 @@ AudioStreamInGeneric::~AudioStreamInGeneric()
{
// FIXME: remove logging
LOGD("AudioStreamInGeneric destructor");
- if (mAudioHardware)
- mAudioHardware->closeInputStream(this);
}
ssize_t AudioStreamInGeneric::read(void* buffer, ssize_t bytes)
{
// FIXME: remove logging
- LOGD("AudioStreamInGeneric::read(%p, %d) from fd %d", buffer, bytes, mFd);
+ LOGD("AudioStreamInGeneric::read(%p, %d) from fd %d", buffer, (int)bytes, mFd);
AutoMutex lock(mLock);
if (mFd < 0) {
LOGE("Attempt to read from unopened device");
@@ -303,10 +359,12 @@ status_t AudioStreamInGeneric::dump(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
result.append(buffer);
- snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+ snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
result.append(buffer);
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
+ snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
+ result.append(buffer);
snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
result.append(buffer);
snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
@@ -315,6 +373,39 @@ status_t AudioStreamInGeneric::dump(int fd, const Vector<String16>& args)
return NO_ERROR;
}
+status_t AudioStreamInGeneric::setParameters(const String8& keyValuePairs)
+{
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 key = String8(AudioParameter::keyRouting);
+ status_t status = NO_ERROR;
+ int device;
+ LOGV("setParameters() %s", keyValuePairs.string());
+
+ if (param.getInt(key, device) == NO_ERROR) {
+ mDevice = device;
+ param.remove(key);
+ }
+
+ if (param.size()) {
+ status = BAD_VALUE;
+ }
+ return status;
+}
+
+String8 AudioStreamInGeneric::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ String8 value;
+ String8 key = String8(AudioParameter::keyRouting);
+
+ if (param.get(key, value) == NO_ERROR) {
+ param.addInt(key, (int)mDevice);
+ }
+
+ LOGV("getParameters() %s", param.toString().string());
+ return param.toString();
+}
+
// ----------------------------------------------------------------------------
}; // namespace android
diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h
index c89df87..42da413 100644
--- a/libs/audioflinger/AudioHardwareGeneric.h
+++ b/libs/audioflinger/AudioHardwareGeneric.h
@@ -39,24 +39,28 @@ public:
virtual status_t set(
AudioHardwareGeneric *hw,
int mFd,
- int format,
- int channelCount,
- uint32_t sampleRate);
+ uint32_t devices,
+ int *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pRate);
virtual uint32_t sampleRate() const { return 44100; }
virtual size_t bufferSize() const { return 4096; }
- virtual int channelCount() const { return 2; }
+ virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
virtual int format() const { return AudioSystem::PCM_16_BIT; }
virtual uint32_t latency() const { return 20; }
- virtual status_t setVolume(float volume) { return INVALID_OPERATION; }
+ virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; }
virtual ssize_t write(const void* buffer, size_t bytes);
virtual status_t standby();
virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
private:
AudioHardwareGeneric *mAudioHardware;
Mutex mLock;
int mFd;
+ uint32_t mDevice;
};
class AudioStreamInGeneric : public AudioStreamIn {
@@ -67,24 +71,28 @@ public:
virtual status_t set(
AudioHardwareGeneric *hw,
int mFd,
- int format,
- int channelCount,
- uint32_t sampleRate,
+ uint32_t devices,
+ int *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pRate,
AudioSystem::audio_in_acoustics acoustics);
- uint32_t sampleRate() const { return 8000; }
+ virtual uint32_t sampleRate() const { return 8000; }
virtual size_t bufferSize() const { return 320; }
- virtual int channelCount() const { return 1; }
+ virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; }
virtual int format() const { return AudioSystem::PCM_16_BIT; }
virtual status_t setGain(float gain) { return INVALID_OPERATION; }
virtual ssize_t read(void* buffer, ssize_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
virtual status_t standby() { return NO_ERROR; }
+ virtual status_t setParameters(const String8& keyValuePairs);
+ virtual String8 getParameters(const String8& keys);
private:
AudioHardwareGeneric *mAudioHardware;
Mutex mLock;
int mFd;
+ uint32_t mDevice;
};
@@ -101,28 +109,27 @@ public:
virtual status_t setMicMute(bool state);
virtual status_t getMicMute(bool* state);
- virtual status_t setParameter(const char* key, const char* value)
- { return NO_ERROR; }
-
// create I/O streams
virtual AudioStreamOut* openOutputStream(
- int format=0,
- int channelCount=0,
- uint32_t sampleRate=0,
+ uint32_t devices,
+ int *format=0,
+ uint32_t *channels=0,
+ uint32_t *sampleRate=0,
status_t *status=0);
+ virtual void closeOutputStream(AudioStreamOut* out);
virtual AudioStreamIn* openInputStream(
- int inputSource,
- int format,
- int channelCount,
- uint32_t sampleRate,
+ uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
status_t *status,
AudioSystem::audio_in_acoustics acoustics);
+ virtual void closeInputStream(AudioStreamIn* in);
void closeOutputStream(AudioStreamOutGeneric* out);
void closeInputStream(AudioStreamInGeneric* in);
protected:
- virtual status_t doRouting() { return NO_ERROR; }
virtual status_t dump(int fd, const Vector<String16>& args);
private:
diff --git a/libs/audioflinger/AudioHardwareInterface.cpp b/libs/audioflinger/AudioHardwareInterface.cpp
index cc1bd8f..37be329 100644
--- a/libs/audioflinger/AudioHardwareInterface.cpp
+++ b/libs/audioflinger/AudioHardwareInterface.cpp
@@ -18,6 +18,7 @@
#include <cutils/properties.h>
#include <string.h>
#include <unistd.h>
+//#define LOG_NDEBUG 0
#define LOG_TAG "AudioHardwareInterface"
#include <utils/Log.h>
@@ -25,15 +26,17 @@
#include "AudioHardwareStub.h"
#include "AudioHardwareGeneric.h"
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
-//#define DUMP_FLINGER_OUT // if defined allows recording samples in a file
-#ifdef DUMP_FLINGER_OUT
+#ifdef ENABLE_AUDIO_DUMP
#include "AudioDumpInterface.h"
#endif
// change to 1 to log routing calls
-#define LOG_ROUTING_CALLS 0
+#define LOG_ROUTING_CALLS 1
namespace android {
@@ -48,14 +51,6 @@ static const char* routingModeStrings[] =
"IN_CALL"
};
-static const char* routeStrings[] =
-{
- "EARPIECE ",
- "SPEAKER ",
- "BLUETOOTH ",
- "HEADSET ",
- "BLUETOOTH_A2DP "
-};
static const char* routeNone = "NONE";
static const char* displayMode(int mode)
@@ -64,22 +59,6 @@ static const char* displayMode(int mode)
return routingModeStrings[0];
return routingModeStrings[mode+3];
}
-
-static const char* displayRoutes(uint32_t routes)
-{
- static char routeStr[80];
- if (routes == 0)
- return routeNone;
- routeStr[0] = 0;
- int bitMask = 1;
- for (int i = 0; i < 4; ++i, bitMask <<= 1) {
- if (routes & bitMask) {
- strcat(routeStr, routeStrings[i]);
- }
- }
- routeStr[strlen(routeStr)-1] = 0;
- return routeStr;
-}
#endif
// ----------------------------------------------------------------------------
@@ -112,13 +91,17 @@ AudioHardwareInterface* AudioHardwareInterface::create()
hw = new AudioHardwareStub();
}
-#ifdef DUMP_FLINGER_OUT
+#ifdef WITH_A2DP
+ hw = new A2dpAudioInterface(hw);
+#endif
+
+#ifdef ENABLE_AUDIO_DUMP
// This code adds a record of buffers in a file to write calls made by AudioFlinger.
// It replaces the current AudioHardwareInterface object by an intermediate one which
// will record buffers in a file (after sending them to hardware) for testing purpose.
- // This feature is enabled by defining symbol DUMP_FLINGER_OUT.
- // The output file is FLINGER_DUMP_NAME. Pause are not recorded in the file.
-
+ // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP.
+ // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file.
+ LOGV("opening PCM dump interface");
hw = new AudioDumpInterface(hw); // replace interface
#endif
return hw;
@@ -132,48 +115,9 @@ AudioStreamIn::~AudioStreamIn() {}
AudioHardwareBase::AudioHardwareBase()
{
- // force a routing update on initialization
- memset(&mRoutes, 0, sizeof(mRoutes));
mMode = 0;
}
-// generics for audio routing - the real work is done in doRouting
-status_t AudioHardwareBase::setRouting(int mode, uint32_t routes)
-{
-#if LOG_ROUTING_CALLS
- LOGD("setRouting: mode=%s, routes=[%s]", displayMode(mode), displayRoutes(routes));
-#endif
- if (mode == AudioSystem::MODE_CURRENT)
- mode = mMode;
- if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
- return BAD_VALUE;
- uint32_t old = mRoutes[mode];
- mRoutes[mode] = routes;
- if ((mode != mMode) || (old == routes))
- return NO_ERROR;
-#if LOG_ROUTING_CALLS
- const char* oldRouteStr = strdup(displayRoutes(old));
- LOGD("doRouting: mode=%s, old route=[%s], new route=[%s]",
- displayMode(mode), oldRouteStr, displayRoutes(routes));
- delete oldRouteStr;
-#endif
- return doRouting();
-}
-
-status_t AudioHardwareBase::getRouting(int mode, uint32_t* routes)
-{
- if (mode == AudioSystem::MODE_CURRENT)
- mode = mMode;
- if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
- return BAD_VALUE;
- *routes = mRoutes[mode];
-#if LOG_ROUTING_CALLS
- LOGD("getRouting: mode=%s, routes=[%s]",
- displayMode(mode), displayRoutes(*routes));
-#endif
- return NO_ERROR;
-}
-
status_t AudioHardwareBase::setMode(int mode)
{
#if LOG_ROUTING_CALLS
@@ -182,29 +126,24 @@ status_t AudioHardwareBase::setMode(int mode)
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
return BAD_VALUE;
if (mMode == mode)
- return NO_ERROR;
-#if LOG_ROUTING_CALLS
- LOGD("doRouting: old mode=%s, new mode=%s route=[%s]",
- displayMode(mMode), displayMode(mode), displayRoutes(mRoutes[mode]));
-#endif
+ return ALREADY_EXISTS;
mMode = mode;
- return doRouting();
+ return NO_ERROR;
}
-status_t AudioHardwareBase::getMode(int* mode)
+// default implementation
+status_t AudioHardwareBase::setParameters(const String8& keyValuePairs)
{
- // Implement: set audio routing
- *mode = mMode;
return NO_ERROR;
}
-status_t AudioHardwareBase::setParameter(const char* key, const char* value)
+// default implementation
+String8 AudioHardwareBase::getParameters(const String8& keys)
{
- // default implementation is to ignore
- return NO_ERROR;
+ String8 result = String8("");
+ return result;
}
-
// default implementation
size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
{
@@ -233,10 +172,6 @@ status_t AudioHardwareBase::dumpState(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
result.append(buffer);
- for (int i = 0, n = AudioSystem::NUM_MODES; i < n; ++i) {
- snprintf(buffer, SIZE, "\tmRoutes[%d]: %d\n", i, mRoutes[i]);
- result.append(buffer);
- }
::write(fd, result.string(), result.size());
dump(fd, args); // Dump the state of the concrete child.
return NO_ERROR;
diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp
index 0ab4c60..1a03059 100644
--- a/libs/audioflinger/AudioHardwareStub.cpp
+++ b/libs/audioflinger/AudioHardwareStub.cpp
@@ -43,10 +43,10 @@ status_t AudioHardwareStub::initCheck()
}
AudioStreamOut* AudioHardwareStub::openOutputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status)
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
{
AudioStreamOutStub* out = new AudioStreamOutStub();
- status_t lStatus = out->set(format, channelCount, sampleRate);
+ status_t lStatus = out->set(format, channels, sampleRate);
if (status) {
*status = lStatus;
}
@@ -56,18 +56,22 @@ AudioStreamOut* AudioHardwareStub::openOutputStream(
return 0;
}
+void AudioHardwareStub::closeOutputStream(AudioStreamOut* out)
+{
+ delete out;
+}
+
AudioStreamIn* AudioHardwareStub::openInputStream(
- int inputSource, int format, int channelCount, uint32_t sampleRate,
+ uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
// check for valid input source
- if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
- (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+ if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
return 0;
}
AudioStreamInStub* in = new AudioStreamInStub();
- status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
+ status_t lStatus = in->set(format, channels, sampleRate, acoustics);
if (status) {
*status = lStatus;
}
@@ -77,6 +81,11 @@ AudioStreamIn* AudioHardwareStub::openInputStream(
return 0;
}
+void AudioHardwareStub::closeInputStream(AudioStreamIn* in)
+{
+ delete in;
+}
+
status_t AudioHardwareStub::setVoiceVolume(float volume)
{
return NO_ERROR;
@@ -107,24 +116,19 @@ status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
// ----------------------------------------------------------------------------
-status_t AudioStreamOutStub::set(int format, int channels, uint32_t rate)
+status_t AudioStreamOutStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate)
{
- // fix up defaults
- if (format == 0) format = AudioSystem::PCM_16_BIT;
- if (channels == 0) channels = channelCount();
- if (rate == 0) rate = sampleRate();
+ if (pFormat) *pFormat = format();
+ if (pChannels) *pChannels = channels();
+ if (pRate) *pRate = sampleRate();
- if ((format == AudioSystem::PCM_16_BIT) &&
- (channels == channelCount()) &&
- (rate == sampleRate()))
- return NO_ERROR;
- return BAD_VALUE;
+ return NO_ERROR;
}
ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
{
// fake timing for audio output
- usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+ usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
return bytes;
}
@@ -141,7 +145,7 @@ status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
- snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+ snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
::write(fd, result.string(), result.size());
@@ -150,20 +154,16 @@ status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
// ----------------------------------------------------------------------------
-status_t AudioStreamInStub::set(int format, int channels, uint32_t rate,
+status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate,
AudioSystem::audio_in_acoustics acoustics)
{
- if ((format == AudioSystem::PCM_16_BIT) &&
- (channels == channelCount()) &&
- (rate == sampleRate()))
- return NO_ERROR;
- return BAD_VALUE;
+ return NO_ERROR;
}
ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
{
// fake timing for audio input
- usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+ usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
memset(buffer, 0, bytes);
return bytes;
}
@@ -179,7 +179,7 @@ status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
result.append(buffer);
- snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+ snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
result.append(buffer);
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h
index bf63cc5..8f43259 100644
--- a/libs/audioflinger/AudioHardwareStub.h
+++ b/libs/audioflinger/AudioHardwareStub.h
@@ -29,29 +29,33 @@ namespace android {
class AudioStreamOutStub : public AudioStreamOut {
public:
- virtual status_t set(int format, int channelCount, uint32_t sampleRate);
+ virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate);
virtual uint32_t sampleRate() const { return 44100; }
virtual size_t bufferSize() const { return 4096; }
- virtual int channelCount() const { return 2; }
+ virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
virtual int format() const { return AudioSystem::PCM_16_BIT; }
virtual uint32_t latency() const { return 0; }
- virtual status_t setVolume(float volume) { return NO_ERROR; }
+ virtual status_t setVolume(float left, float right) { return NO_ERROR; }
virtual ssize_t write(const void* buffer, size_t bytes);
virtual status_t standby();
virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;}
+ virtual String8 getParameters(const String8& keys) {String8 result = String8(""); return result;}
};
class AudioStreamInStub : public AudioStreamIn {
public:
- virtual status_t set(int format, int channelCount, uint32_t sampleRate, AudioSystem::audio_in_acoustics acoustics);
+ virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics);
virtual uint32_t sampleRate() const { return 8000; }
virtual size_t bufferSize() const { return 320; }
- virtual int channelCount() const { return 1; }
+ virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; }
virtual int format() const { return AudioSystem::PCM_16_BIT; }
virtual status_t setGain(float gain) { return NO_ERROR; }
virtual ssize_t read(void* buffer, ssize_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
virtual status_t standby() { return NO_ERROR; }
+ virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;}
+ virtual String8 getParameters(const String8& keys) {String8 result = String8(""); return result;}
};
class AudioHardwareStub : public AudioHardwareBase
@@ -67,26 +71,25 @@ public:
virtual status_t setMicMute(bool state) { mMicMute = state; return NO_ERROR; }
virtual status_t getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; }
- virtual status_t setParameter(const char* key, const char* value)
- { return NO_ERROR; }
-
// create I/O streams
virtual AudioStreamOut* openOutputStream(
- int format=0,
- int channelCount=0,
- uint32_t sampleRate=0,
+ uint32_t devices,
+ int *format=0,
+ uint32_t *channels=0,
+ uint32_t *sampleRate=0,
status_t *status=0);
+ virtual void closeOutputStream(AudioStreamOut* out);
virtual AudioStreamIn* openInputStream(
- int inputSource,
- int format,
- int channelCount,
- uint32_t sampleRate,
+ uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
status_t *status,
- AudioSystem::audio_in_acoustics acoustics);
+ AudioSystem::audio_in_acoustics acoustics);
+ virtual void closeInputStream(AudioStreamIn* in);
protected:
- virtual status_t doRouting() { return NO_ERROR; }
virtual status_t dump(int fd, const Vector<String16>& args);
bool mMicMute;
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp
index b02efcc..19a442a 100644
--- a/libs/audioflinger/AudioMixer.cpp
+++ b/libs/audioflinger/AudioMixer.cpp
@@ -610,7 +610,6 @@ void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
t->in = in;
}
-inline
void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
{
for (size_t i=0 ; i<c ; i++) {
diff --git a/libs/audioflinger/AudioMixer.h b/libs/audioflinger/AudioMixer.h
index 72ca28a..15766cd 100644
--- a/libs/audioflinger/AudioMixer.h
+++ b/libs/audioflinger/AudioMixer.h
@@ -85,6 +85,8 @@ public:
uint32_t trackNames() const { return mTrackNames; }
+ static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
+
private:
enum {
@@ -176,7 +178,6 @@ private:
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp);
static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
- static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
static void process__validate(state_t* state, void* output);
static void process__nop(state_t* state, void* output);
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.cpp b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
new file mode 100644
index 0000000..cf9ab88
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
@@ -0,0 +1,764 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#define LOG_TAG "AudioPolicyManagerGeneric"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include "AudioPolicyManagerGeneric.h"
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerGeneric::setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address)
+{
+
+ LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+ // connect/disconnect only 1 device at a time
+ if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
+
+ if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+ LOGE("setDeviceConnectionState() invalid address: %s", device_address);
+ return BAD_VALUE;
+ }
+
+ // handle output devices
+ if (AudioSystem::isOutputDevice(device)) {
+ switch (state)
+ {
+ // handle output device connection
+ case AudioSystem::DEVICE_STATE_AVAILABLE:
+ if (mAvailableOutputDevices & device) {
+ LOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ LOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ mAvailableOutputDevices |= device;
+ break;
+ // handle output device disconnection
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE:
+ if (!(mAvailableOutputDevices & device)) {
+ LOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ LOGV("setDeviceConnectionState() disconnecting device %x", device);
+ // remove device from available output devices
+ mAvailableOutputDevices &= ~device;
+ break;
+
+ default:
+ LOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+ }
+ // handle input devices
+ if (AudioSystem::isInputDevice(device)) {
+ switch (state)
+ {
+ // handle input device connection
+ case AudioSystem::DEVICE_STATE_AVAILABLE:
+ if (mAvailableInputDevices & device) {
+ LOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices |= device;
+ break;
+
+ // handle input device disconnection
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE:
+ if (!(mAvailableInputDevices & device)) {
+ LOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices &= ~device;
+ break;
+
+ default:
+ LOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+ }
+
+ LOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerGeneric::getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address)
+{
+ AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
+ String8 address = String8(device_address);
+ if (AudioSystem::isOutputDevice(device)) {
+ if (device & mAvailableOutputDevices) {
+ state = AudioSystem::DEVICE_STATE_AVAILABLE;
+ }
+ } else if (AudioSystem::isInputDevice(device)) {
+ if (device & mAvailableInputDevices) {
+ state = AudioSystem::DEVICE_STATE_AVAILABLE;
+ }
+ }
+
+ return state;
+}
+
+void AudioPolicyManagerGeneric::setPhoneState(int state)
+{
+ LOGV("setPhoneState() state %d", state);
+ uint32_t newDevice = 0;
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
+ LOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ LOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+
+ // if leaving or entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (state == AudioSystem::MODE_IN_CALL ||
+ oldState == AudioSystem::MODE_IN_CALL) {
+ bool starting = (state == AudioSystem::MODE_IN_CALL) ? true : false;
+ LOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, starting);
+ }
+ }
+}
+
+void AudioPolicyManagerGeneric::setRingerMode(uint32_t mode, uint32_t mask)
+{
+ LOGV("setRingerMode() mode %x, mask %x", mode, mask);
+
+ mRingerMode = mode;
+}
+
+void AudioPolicyManagerGeneric::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+ LOGV("setForceUse) usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+ mForceUse[usage] = config;
+}
+
+AudioSystem::forced_config AudioPolicyManagerGeneric::getForceUse(AudioSystem::force_use usage)
+{
+ return mForceUse[usage];
+}
+
+void AudioPolicyManagerGeneric::setSystemProperty(const char* property, const char* value)
+{
+ LOGV("setSystemProperty() property %s, value %s", property, value);
+ if (strcmp(property, "ro.camera.sound.forced") == 0) {
+ if (atoi(value)) {
+ LOGV("ENFORCED_AUDIBLE cannot be muted");
+ mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
+ } else {
+ LOGV("ENFORCED_AUDIBLE can be muted");
+ mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
+ }
+ }
+}
+
+audio_io_handle_t AudioPolicyManagerGeneric::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::output_flags flags)
+{
+ LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelcount %d, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannelcount, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ LOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannels = (mTestChannelcount == 1) ? AudioSystem::CHANNEL_OUT_MONO : AudioSystem::CHANNEL_OUT_STEREO;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ mOutputs.add(mTestOutputs[mCurOutput], outputDesc);
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+ if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format != 0 && !AudioSystem::isLinearPCM(format)) ||
+ (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+ return NULL;
+ }
+
+ return mHardwareOutput;
+}
+
+status_t AudioPolicyManagerGeneric::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ LOGV("startOutput() output %p, stream %d", output, stream);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("startOutput() unknow output %p", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // handle special case for sonification while in call
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ handleIncallSonification(stream, true);
+ }
+
+ // incremenent usage count for this stream on the requested output:
+ outputDesc->changeRefCount(stream, 1);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ LOGV("stopOutput() output %p, stream %d", output, stream);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("stopOutput() unknow output %p", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // handle special case for sonification while in call
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ handleIncallSonification(stream, false);
+ }
+
+ if (outputDesc->isUsedByStream(stream)) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ return NO_ERROR;
+ } else {
+ LOGW("stopOutput() refcount is already 0 for output %p", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManagerGeneric::releaseOutput(audio_io_handle_t output)
+{
+ LOGV("releaseOutput() %p", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("releaseOutput() releasing unknown output %p", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->refCount() == 0) {
+ mpClientInterface->closeOutput(output);
+ delete mOutputs.valueAt(index);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ }
+#endif //AUDIO_POLICY_TEST
+}
+
+audio_io_handle_t AudioPolicyManagerGeneric::getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::audio_in_acoustics acoustics)
+{
+ audio_io_handle_t input = 0;
+ uint32_t device;
+
+ LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
+
+ AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
+ inputDesc->mDevice = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannels = channels;
+ inputDesc->mAcoustics = acoustics;
+ inputDesc->mRefCount = 0;
+ input = mpClientInterface->openInput(&inputDesc->mDevice,
+ &inputDesc->mSamplingRate,
+ &inputDesc->mFormat,
+ &inputDesc->mChannels,
+ inputDesc->mAcoustics);
+
+ // only accept input with the exact requested set of parameters
+ if ((samplingRate != inputDesc->mSamplingRate) ||
+ (format != inputDesc->mFormat) ||
+ (channels != inputDesc->mChannels)) {
+ LOGV("getOutput() failed opening input: samplingRate %d, format %d, channels %d",
+ samplingRate, format, channels);
+ mpClientInterface->closeInput(input);
+ delete inputDesc;
+ return NULL;
+ }
+ mInputs.add(input, inputDesc);
+ return input;
+}
+
+status_t AudioPolicyManagerGeneric::startInput(audio_io_handle_t input)
+{
+ LOGV("startInput() input %p", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("startInput() unknow input %p", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+#ifdef AUDIO_POLICY_TEST
+ if (mTestInput == 0)
+#endif //AUDIO_POLICY_TEST
+ {
+ // refuse 2 active AudioRecord clients at the same time
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ if (mInputs.valueAt(i)->mRefCount > 0) {
+ LOGW("startInput() input %p, other input %p already started", input, mInputs.keyAt(i));
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ inputDesc->mRefCount = 1;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::stopInput(audio_io_handle_t input)
+{
+ LOGV("stopInput() input %p", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("stopInput() unknow input %p", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+ if (inputDesc->mRefCount == 0) {
+ LOGW("stopInput() input %p already stopped", input);
+ return INVALID_OPERATION;
+ } else {
+ inputDesc->mRefCount = 0;
+ return NO_ERROR;
+ }
+}
+
+void AudioPolicyManagerGeneric::releaseInput(audio_io_handle_t input)
+{
+ LOGV("releaseInput() %p", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("releaseInput() releasing unknown input %p", input);
+ return;
+ }
+ mpClientInterface->closeInput(input);
+ delete mInputs.valueAt(index);
+ mInputs.removeItem(input);
+}
+
+
+
+void AudioPolicyManagerGeneric::initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax)
+{
+ LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ mStreams[stream].mIndexMin = indexMin;
+ mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManagerGeneric::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+
+ LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
+ mStreams[stream].mIndexCur = index;
+
+ // do not change actual stream volume if the stream is muted
+ if (mStreams[stream].mMuteCount != 0) {
+ return NO_ERROR;
+ }
+
+ // Do not changed in call volume if bluetooth is connected and vice versa
+ if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+ (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
+ LOGV("setStreamVolumeIndex() cannot set stream %d volume with force use = %d for comm",
+ stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
+ return INVALID_OPERATION;
+ }
+
+ // compute and apply stream volume on all outputs according to connected device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ uint32_t device = outputDesc->device();
+
+ float volume = computeVolume((int)stream, index, device);
+
+ LOGV("setStreamVolume() for output %p stream %d, volume %f", mOutputs.keyAt(i), stream, volume);
+ mpClientInterface->setStreamVolume(stream, volume, mOutputs.keyAt(i));
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+ if (index == 0) {
+ return BAD_VALUE;
+ }
+ LOGV("getStreamVolumeIndex() stream %d", stream);
+ *index = mStreams[stream].mIndexCur;
+ return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerGeneric
+// ----------------------------------------------------------------------------
+
+// --- class factory
+
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+{
+ return new AudioPolicyManagerGeneric(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+ delete interface;
+}
+
+AudioPolicyManagerGeneric::AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0)
+{
+ mpClientInterface = clientInterface;
+
+ for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+ mForceUse[i] = AudioSystem::FORCE_NONE;
+ }
+
+ // devices available by default are speaker, ear piece and microphone
+ mAvailableOutputDevices = AudioSystem::DEVICE_OUT_SPEAKER;
+ mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+
+ // open hardware output
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+ mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ if (mHardwareOutput == 0) {
+ LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+ } else {
+ mOutputs.add(mHardwareOutput, outputDesc);
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AudioSystem::PCM_16_BIT;
+ mTestChannelcount = 2;
+ mTestLatencyMs = 0;
+ mCurOutput = 0;
+ mDirectOutput = false;
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ mTestOutputs[i] = 0;
+ }
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManagerGeneric::~AudioPolicyManagerGeneric()
+{
+#ifdef AUDIO_POLICY_TEST
+ exit();
+#endif //AUDIO_POLICY_TEST
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ mpClientInterface->closeOutput(mOutputs.keyAt(i));
+ delete mOutputs.valueAt(i);
+ }
+ mOutputs.clear();
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ delete mInputs.valueAt(i);
+ }
+ mInputs.clear();
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManagerGeneric::threadLoop()
+{
+ LOGV("entering threadLoop()");
+ while (!exitPending())
+ {
+ Mutex::Autolock _l(mLock);
+ mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+ String8 command;
+ command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+ if (command != "") {
+ LOGV("Test command %s received", command.string());
+ AudioParameter param = AudioParameter(command);
+ int valueInt;
+ String8 value;
+ if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_output"));
+ mCurOutput = valueInt;
+ }
+ if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_direct"));
+ if (value == "false") {
+ mDirectOutput = false;
+ } else if (value == "true") {
+ mDirectOutput = true;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_input"));
+ mTestInput = valueInt;
+ }
+
+ if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_format"));
+ if (value == "PCM 16 bits") {
+ mTestFormat = AudioSystem::PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ mTestFormat = AudioSystem::PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ mTestFormat = AudioSystem::MP3;
+ }
+ }
+ if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_channels"));
+ if (value == "Channels Stereo") {
+ mTestChannelcount = 2;
+ } else if (value == "Channels Mono") {
+ mTestChannelcount = 1;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_sampleRate"));
+ if (valueInt >= 0 && valueInt <= 96000) {
+ mTestSamplingRate = valueInt;
+ }
+ }
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManagerGeneric::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManagerGeneric::testOutputIndex(audio_io_handle_t output)
+{
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ if (output == mTestOutputs[i]) return i;
+ }
+ return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+AudioPolicyManagerGeneric::routing_strategy AudioPolicyManagerGeneric::getStrategy(AudioSystem::stream_type stream)
+{
+ // stream to strategy mapping
+ switch (stream) {
+ case AudioSystem::VOICE_CALL:
+ case AudioSystem::BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AudioSystem::RING:
+ case AudioSystem::NOTIFICATION:
+ case AudioSystem::ALARM:
+ case AudioSystem::ENFORCED_AUDIBLE:
+ return STRATEGY_SONIFICATION;
+ case AudioSystem::DTMF:
+ return STRATEGY_DTMF;
+ default:
+ LOGE("unknown stream type");
+ case AudioSystem::SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AudioSystem::TTS:
+ case AudioSystem::MUSIC:
+ return STRATEGY_MEDIA;
+ }
+}
+
+
+float AudioPolicyManagerGeneric::computeVolume(int stream, int index, uint32_t device)
+{
+ float volume = 1.0;
+
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ // Force max volume if stream cannot be muted
+ if (!streamDesc.mCanBeMuted) index = streamDesc.mIndexMax;
+
+ int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
+ volume = AudioSystem::linearToLog(volInt);
+
+ return volume;
+}
+
+void AudioPolicyManagerGeneric::setStreamMute(int stream, bool on, audio_io_handle_t output)
+{
+ LOGV("setStreamMute() stream %d, mute %d, output %p", stream, on, output);
+
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ if (on) {
+ if (streamDesc.mMuteCount++ == 0) {
+ if (streamDesc.mCanBeMuted) {
+ mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, 0, output);
+ }
+ }
+ } else {
+ if (streamDesc.mMuteCount == 0) {
+ LOGW("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--streamDesc.mMuteCount == 0) {
+ uint32_t device = mOutputs.valueFor(output)->mDevice;
+ float volume = computeVolume(stream, streamDesc.mIndexCur, device);
+ mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output);
+ }
+ }
+}
+
+void AudioPolicyManagerGeneric::handleIncallSonification(int stream, bool starting)
+{
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
+ LOGV("handleIncallSonification() stream %d starting %d device %x", stream, starting, outputDesc->mDevice);
+ if (outputDesc->isUsedByStream((AudioSystem::stream_type)stream)) {
+ if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
+ LOGV("handleIncallSonification() low visibility");
+ setStreamMute(stream, starting, mHardwareOutput);
+ } else {
+ if (starting) {
+ mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerGeneric::AudioOutputDescriptor::AudioOutputDescriptor()
+ : mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
+ mFlags((AudioSystem::output_flags)0), mDevice(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+ mRefCount[i] = 0;
+ }
+}
+
+uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::device()
+{
+ return mDevice;
+}
+
+void AudioPolicyManagerGeneric::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
+{
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::refCount()
+{
+ uint32_t refcount = 0;
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ refcount += mRefCount[i];
+ }
+ return refcount;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManagerGeneric::AudioInputDescriptor::AudioInputDescriptor()
+ : mSamplingRate(0), mFormat(0), mChannels(0),
+ mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+{
+}
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.h b/libs/audioflinger/AudioPolicyManagerGeneric.h
new file mode 100644
index 0000000..ddcb306
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerGeneric.h
@@ -0,0 +1,189 @@
+/*
+ * Copyright (C) 2009 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 <stdint.h>
+#include <sys/types.h>
+
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <utils/threads.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+#define NUM_TEST_OUTPUTS 5
+
+class AudioPolicyManagerGeneric: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+ , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+ AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManagerGeneric();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address);
+ virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address);
+ virtual void setPhoneState(int state);
+ virtual void setRingerMode(uint32_t mode, uint32_t mask);
+ virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+ virtual void setSystemProperty(const char* property, const char* value);
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::output_flags flags);
+ virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual void releaseOutput(audio_io_handle_t output);
+ virtual audio_io_handle_t getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::audio_in_acoustics acoustics);
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input);
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input);
+ virtual void releaseInput(audio_io_handle_t input);
+ virtual void initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
+ virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
+
+private:
+
+ enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_DTMF,
+ NUM_STRATEGIES
+ };
+
+ // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+ // and keep track of the usage of this output by each audio stream type.
+ class AudioOutputDescriptor
+ {
+ public:
+ AudioOutputDescriptor();
+
+
+ uint32_t device();
+ void changeRefCount(AudioSystem::stream_type, int delta);
+ bool isUsedByStream(AudioSystem::stream_type stream) { return mRefCount[stream] > 0 ? true : false; }
+ uint32_t refCount();
+
+ uint32_t mSamplingRate; //
+ uint32_t mFormat; //
+ uint32_t mChannels; // output configuration
+ uint32_t mLatency; //
+ AudioSystem::output_flags mFlags; //
+ uint32_t mDevice; // current device this output is routed to
+ uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
+ };
+
+ // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+ // and keep track of the usage of this input.
+ class AudioInputDescriptor
+ {
+ public:
+ AudioInputDescriptor();
+
+ uint32_t mSamplingRate; //
+ uint32_t mFormat; // input configuration
+ uint32_t mChannels; //
+ AudioSystem::audio_in_acoustics mAcoustics; //
+ uint32_t mDevice; // current device this input is routed to
+ uint32_t mRefCount; // number of AudioRecord clients using this output
+ };
+
+ // stream descriptor used for volume control
+ class StreamDescriptor
+ {
+ public:
+ StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mIndexCur(1), mMuteCount(0), mCanBeMuted(true) {}
+
+ int mIndexMin; // min volume index
+ int mIndexMax; // max volume index
+ int mIndexCur; // current volume index
+ int mMuteCount; // mute request counter
+ bool mCanBeMuted; // true is the stream can be muted
+ };
+
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(AudioSystem::stream_type stream);
+ // return the output handle of an output routed to the specified device, 0 if no output
+ // is routed to the device
+ float computeVolume(int stream, int index, uint32_t device);
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(int stream, bool on, audio_io_handle_t output);
+ // handle special cases for sonification strategy while in call: mute streams or replace by
+ // a special tone in the device used for communication
+ void handleIncallSonification(int stream, bool starting);
+
+
+#ifdef AUDIO_POLICY_TEST
+ virtual bool threadLoop();
+ void exit();
+ int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+
+ AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
+ audio_io_handle_t mHardwareOutput; // hardware output handler
+
+ KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; // list ot output descritors
+ KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
+ uint32_t mAvailableOutputDevices; // bit field of all available output devices
+ uint32_t mAvailableInputDevices; // bit field of all available input devices
+ int mPhoneState; // current phone state
+ uint32_t mRingerMode; // current ringer mode
+ AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration
+
+ StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control
+
+#ifdef AUDIO_POLICY_TEST
+ Mutex mLock;
+ Condition mWaitWorkCV;
+
+ int mCurOutput;
+ bool mDirectOutput;
+ audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+ int mTestInput;
+ uint32_t mTestDevice;
+ uint32_t mTestSamplingRate;
+ uint32_t mTestFormat;
+ uint32_t mTestChannelcount;
+ uint32_t mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+};
+
+};
diff --git a/libs/audioflinger/AudioPolicyService.cpp b/libs/audioflinger/AudioPolicyService.cpp
new file mode 100644
index 0000000..4810a44
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyService.cpp
@@ -0,0 +1,677 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#define LOG_TAG "AudioPolicyService"
+//#define LOG_NDEBUG 0
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <cutils/properties.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+#include "AudioPolicyService.h"
+#include <cutils/properties.h>
+#include <dlfcn.h>
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+namespace android {
+
+const char *AudioPolicyService::sAudioPolicyLibrary = "/system/lib/libaudiopolicy.so";
+const char *AudioPolicyService::sAudioPolicyGenericLibrary = "/system/lib/libaudiopolicygeneric.so";
+
+static bool checkPermission() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+ if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+ bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+ if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioPolicyService::AudioPolicyService()
+ : BnAudioPolicyService() , mpPolicyManager(NULL), mpPolicyManagerLibHandle(NULL)
+{
+ const char *audioPolicyLibrary;
+ char value[PROPERTY_VALUE_MAX];
+
+#if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST)
+ audioPolicyLibrary = sAudioPolicyGenericLibrary;
+ LOGV("build for GENERIC_AUDIO - using generic audio policy");
+#else
+ // if running in emulation - use the emulator driver
+ if (property_get("ro.kernel.qemu", value, 0)) {
+ LOGV("Running in emulation - using generic audio policy");
+ audioPolicyLibrary = sAudioPolicyGenericLibrary;
+ }
+ else {
+ LOGV("Using hardware specific audio policy");
+ audioPolicyLibrary = sAudioPolicyLibrary;
+ }
+#endif
+
+
+ mpPolicyManagerLibHandle = dlopen(audioPolicyLibrary, RTLD_NOW | RTLD_LOCAL);
+ if (mpPolicyManagerLibHandle == NULL) {
+ LOGW("Could not load libaudio policy library");
+ return;
+ }
+
+ AudioPolicyInterface *(*createManager)(AudioPolicyClientInterface *) =
+ reinterpret_cast<AudioPolicyInterface* (*)(AudioPolicyClientInterface *)>(dlsym(mpPolicyManagerLibHandle, "createAudioPolicyManager"));
+
+ if (createManager == NULL ) {
+ LOGW("Could not get createAudioPolicyManager method");
+ return;
+ }
+
+ // start tone playback thread
+ mTonePlaybacThread = new AudioCommandThread();
+ // start audio commands thread
+ mAudioCommandThread = new AudioCommandThread();
+
+ mpPolicyManager = (*createManager)(this);
+
+ // load properties
+ property_get("ro.camera.sound.forced", value, "0");
+ mpPolicyManager->setSystemProperty("ro.camera.sound.forced", value);
+}
+
+AudioPolicyService::~AudioPolicyService()
+{
+ mTonePlaybacThread->exit();
+ mTonePlaybacThread.clear();
+ mAudioCommandThread->exit();
+ mAudioCommandThread.clear();
+
+ if (mpPolicyManager) {
+ void(*destroyManager)(AudioPolicyInterface *) =
+ reinterpret_cast<void(*)(AudioPolicyInterface *)>(dlsym(mpPolicyManagerLibHandle, "destroyAudioPolicyManager"));
+
+ if (destroyManager == NULL ) {
+ LOGW("Could not get destroyAudioPolicyManager method");
+ return;
+ }
+ (*destroyManager)(mpPolicyManager);
+ }
+}
+
+
+status_t AudioPolicyService::setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (!AudioSystem::isOutputDevice(device) && !AudioSystem::isInputDevice(device)) {
+ return BAD_VALUE;
+ }
+ if (state != AudioSystem::DEVICE_STATE_AVAILABLE && state != AudioSystem::DEVICE_STATE_UNAVAILABLE) {
+ return BAD_VALUE;
+ }
+
+ LOGV("setDeviceConnectionState() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->setDeviceConnectionState(device, state, device_address);
+}
+
+AudioSystem::device_connection_state AudioPolicyService::getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address)
+{
+ if (mpPolicyManager == NULL) {
+ return AudioSystem::DEVICE_STATE_UNAVAILABLE;
+ }
+ if (!checkPermission()) {
+ return AudioSystem::DEVICE_STATE_UNAVAILABLE;
+ }
+ return mpPolicyManager->getDeviceConnectionState(device, device_address);
+}
+
+status_t AudioPolicyService::setPhoneState(int state)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
+ return BAD_VALUE;
+ }
+
+ LOGV("setPhoneState() tid %d", gettid());
+
+ // TODO: check if it is more appropriate to do it in platform specific policy manager
+ AudioSystem::setMode(state);
+
+ Mutex::Autolock _l(mLock);
+ mpPolicyManager->setPhoneState(state);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::setRingerMode(uint32_t mode, uint32_t mask)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+
+ mpPolicyManager->setRingerMode(mode, mask);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) {
+ return BAD_VALUE;
+ }
+ if (config < 0 || config >= AudioSystem::NUM_FORCE_CONFIG) {
+ return BAD_VALUE;
+ }
+ LOGV("setForceUse() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ mpPolicyManager->setForceUse(usage, config);
+ return NO_ERROR;
+}
+
+AudioSystem::forced_config AudioPolicyService::getForceUse(AudioSystem::force_use usage)
+{
+ if (mpPolicyManager == NULL) {
+ return AudioSystem::FORCE_NONE;
+ }
+ if (!checkPermission()) {
+ return AudioSystem::FORCE_NONE;
+ }
+ if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) {
+ return AudioSystem::FORCE_NONE;
+ }
+ return mpPolicyManager->getForceUse(usage);
+}
+
+audio_io_handle_t AudioPolicyService::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::output_flags flags)
+{
+ if (mpPolicyManager == NULL) {
+ return NULL;
+ }
+ LOGV("getOutput() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->getOutput(stream, samplingRate, format, channels, flags);
+}
+
+status_t AudioPolicyService::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ LOGV("startOutput() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->startOutput(output, stream);
+}
+
+status_t AudioPolicyService::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ LOGV("stopOutput() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->stopOutput(output, stream);
+}
+
+void AudioPolicyService::releaseOutput(audio_io_handle_t output)
+{
+ if (mpPolicyManager == NULL) {
+ return;
+ }
+ LOGV("releaseOutput() tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ mpPolicyManager->releaseOutput(output);
+}
+
+audio_io_handle_t AudioPolicyService::getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::audio_in_acoustics acoustics)
+{
+ if (mpPolicyManager == NULL) {
+ return NULL;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->getInput(inputSource, samplingRate, format, channels, acoustics);
+}
+
+status_t AudioPolicyService::startInput(audio_io_handle_t input)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->startInput(input);
+}
+
+status_t AudioPolicyService::stopInput(audio_io_handle_t input)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpPolicyManager->stopInput(input);
+}
+
+void AudioPolicyService::releaseInput(audio_io_handle_t input)
+{
+ if (mpPolicyManager == NULL) {
+ return;
+ }
+ Mutex::Autolock _l(mLock);
+ mpPolicyManager->releaseInput(input);
+}
+
+status_t AudioPolicyService::initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+ mpPolicyManager->initStreamVolume(stream, indexMin, indexMax);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+
+ return mpPolicyManager->setStreamVolumeIndex(stream, index);
+}
+
+status_t AudioPolicyService::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+ if (mpPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ if (!checkPermission()) {
+ return PERMISSION_DENIED;
+ }
+ if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+ return mpPolicyManager->getStreamVolumeIndex(stream, index);
+}
+
+void AudioPolicyService::binderDied(const wp<IBinder>& who) {
+ LOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
+}
+
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+{
+ if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ dumpPermissionDenial(fd, args);
+ } else {
+
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+ snprintf(buffer, SIZE, "Permission Denial: "
+ "can't dump AudioPolicyService from pid=%d, uid=%d\n",
+ IPCThreadState::self()->getCallingPid(),
+ IPCThreadState::self()->getCallingUid());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioPolicyService::onTransact(code, data, reply, flags);
+}
+
+
+// ----------------------------------------------------------------------------
+void AudioPolicyService::instantiate() {
+ defaultServiceManager()->addService(
+ String16("media.audio_policy"), new AudioPolicyService());
+}
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyClientInterface implementation
+// ----------------------------------------------------------------------------
+
+
+audio_io_handle_t AudioPolicyService::openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ AudioSystem::output_flags flags)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ LOGW("openOutput() could not get AudioFlinger");
+ return NULL;
+ }
+
+ return af->openOutput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, pLatencyMs, flags);
+}
+
+audio_io_handle_t AudioPolicyService::openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ LOGW("openDuplicateOutput() could not get AudioFlinger");
+ return NULL;
+ }
+ return af->openDuplicateOutput(output1, output2);
+}
+
+status_t AudioPolicyService::closeOutput(audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
+
+ return af->closeOutput(output);
+}
+
+
+status_t AudioPolicyService::suspendOutput(audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ LOGW("suspendOutput() could not get AudioFlinger");
+ return PERMISSION_DENIED;
+ }
+
+ return af->suspendOutput(output);
+}
+
+status_t AudioPolicyService::restoreOutput(audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ LOGW("restoreOutput() could not get AudioFlinger");
+ return PERMISSION_DENIED;
+ }
+
+ return af->restoreOutput(output);
+}
+
+audio_io_handle_t AudioPolicyService::openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ LOGW("openInput() could not get AudioFlinger");
+ return NULL;
+ }
+
+ return af->openInput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, acoustics);
+}
+
+status_t AudioPolicyService::closeInput(audio_io_handle_t input)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
+
+ return af->closeInput(input);
+}
+
+status_t AudioPolicyService::setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output)
+{
+ return mAudioCommandThread->volumeCommand((int)stream, volume, (void *)output);
+}
+
+status_t AudioPolicyService::setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
+
+ return af->setStreamOutput(stream, output);
+}
+
+
+void AudioPolicyService::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
+ mAudioCommandThread->parametersCommand((void *)ioHandle, keyValuePairs);
+}
+
+String8 AudioPolicyService::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
+ String8 result = AudioSystem::getParameters(ioHandle, keys);
+ return result;
+}
+
+status_t AudioPolicyService::startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream)
+{
+ mTonePlaybacThread->startToneCommand(tone, stream);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::stopTone()
+{
+ mTonePlaybacThread->stopToneCommand();
+ return NO_ERROR;
+}
+
+
+// ----------- AudioPolicyService::AudioCommandThread implementation ----------
+
+AudioPolicyService::AudioCommandThread::AudioCommandThread()
+ : Thread(false)
+{
+ mpToneGenerator = NULL;
+}
+
+
+AudioPolicyService::AudioCommandThread::~AudioCommandThread()
+{
+ mAudioCommands.clear();
+ if (mpToneGenerator != NULL) delete mpToneGenerator;
+}
+
+void AudioPolicyService::AudioCommandThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "AudioCommandThread");
+
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+}
+
+bool AudioPolicyService::AudioCommandThread::threadLoop()
+{
+ mLock.lock();
+ while (!exitPending())
+ {
+ while(!mAudioCommands.isEmpty()) {
+ AudioCommand *command = mAudioCommands[0];
+ mAudioCommands.removeAt(0);
+ switch (command->mCommand) {
+ case START_TONE: {
+ mLock.unlock();
+ ToneData *data = (ToneData *)command->mParam;
+ LOGV("AudioCommandThread() processing start tone %d on stream %d",
+ data->mType, data->mStream);
+ if (mpToneGenerator != NULL)
+ delete mpToneGenerator;
+ mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
+ mpToneGenerator->startTone(data->mType);
+ delete data;
+ mLock.lock();
+ }break;
+ case STOP_TONE: {
+ mLock.unlock();
+ LOGV("AudioCommandThread() processing stop tone");
+ if (mpToneGenerator != NULL) {
+ mpToneGenerator->stopTone();
+ delete mpToneGenerator;
+ mpToneGenerator = NULL;
+ }
+ mLock.lock();
+ }break;
+ case SET_VOLUME: {
+ VolumeData *data = (VolumeData *)command->mParam;
+ LOGV("AudioCommandThread() processing set volume stream %d, volume %f, output %p", data->mStream, data->mVolume, data->mIO);
+ mCommandStatus = AudioSystem::setStreamVolume(data->mStream, data->mVolume, data->mIO);
+ mCommandCond.signal();
+ mWaitWorkCV.wait(mLock);
+ delete data;
+ }break;
+ case SET_PARAMETERS: {
+ ParametersData *data = (ParametersData *)command->mParam;
+ LOGV("AudioCommandThread() processing set parameters string %s, io %p", data->mKeyValuePairs.string(), data->mIO);
+ mCommandStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+ mCommandCond.signal();
+ mWaitWorkCV.wait(mLock);
+ delete data;
+ }break;
+ default:
+ LOGW("AudioCommandThread() unknown command %d", command->mCommand);
+ }
+ delete command;
+ }
+ LOGV("AudioCommandThread() going to sleep");
+ mWaitWorkCV.wait(mLock);
+ LOGV("AudioCommandThread() waking up");
+ }
+ mLock.unlock();
+ return false;
+}
+
+void AudioPolicyService::AudioCommandThread::startToneCommand(int type, int stream)
+{
+ Mutex::Autolock _l(mLock);
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = START_TONE;
+ ToneData *data = new ToneData();
+ data->mType = type;
+ data->mStream = stream;
+ command->mParam = (void *)data;
+ mAudioCommands.add(command);
+ LOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream);
+ mWaitWorkCV.signal();
+}
+
+void AudioPolicyService::AudioCommandThread::stopToneCommand()
+{
+ Mutex::Autolock _l(mLock);
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = STOP_TONE;
+ command->mParam = NULL;
+ mAudioCommands.add(command);
+ LOGV("AudioCommandThread() adding tone stop");
+ mWaitWorkCV.signal();
+}
+
+status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream, float volume, void *output)
+{
+ Mutex::Autolock _l(mLock);
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = SET_VOLUME;
+ VolumeData *data = new VolumeData();
+ data->mStream = stream;
+ data->mVolume = volume;
+ data->mIO = output;
+ command->mParam = data;
+ mAudioCommands.add(command);
+ LOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %p", stream, volume, output);
+ mWaitWorkCV.signal();
+ mCommandCond.wait(mLock);
+ status_t status = mCommandStatus;
+ mWaitWorkCV.signal();
+ return status;
+}
+
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(void *ioHandle, const String8& keyValuePairs)
+{
+ Mutex::Autolock _l(mLock);
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = SET_PARAMETERS;
+ ParametersData *data = new ParametersData();
+ data->mIO = ioHandle;
+ data->mKeyValuePairs = keyValuePairs;
+ command->mParam = data;
+ mAudioCommands.add(command);
+ LOGV("AudioCommandThread() adding set parameter string %s, io %p", keyValuePairs.string(), ioHandle);
+ mWaitWorkCV.signal();
+ mCommandCond.wait(mLock);
+ status_t status = mCommandStatus;
+ mWaitWorkCV.signal();
+ return status;
+}
+
+void AudioPolicyService::AudioCommandThread::exit()
+{
+ LOGV("AudioCommandThread::exit");
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyService.h b/libs/audioflinger/AudioPolicyService.h
new file mode 100644
index 0000000..47173dd
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyService.h
@@ -0,0 +1,201 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYSERVICE_H
+#define ANDROID_AUDIOPOLICYSERVICE_H
+
+#include <media/IAudioPolicyService.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <media/ToneGenerator.h>
+
+namespace android {
+
+class String8;
+
+// ----------------------------------------------------------------------------
+
+class AudioPolicyService: public BnAudioPolicyService, public AudioPolicyClientInterface, public IBinder::DeathRecipient
+{
+
+public:
+ static void instantiate();
+
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+ //
+ // BnAudioPolicyService (see AudioPolicyInterface for method descriptions)
+ //
+
+ virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address);
+ virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address);
+ virtual status_t setPhoneState(int state);
+ virtual status_t setRingerMode(uint32_t mode, uint32_t mask);
+ virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT);
+ virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual void releaseOutput(audio_io_handle_t output);
+ virtual audio_io_handle_t getInput(int inputSource,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0);
+ virtual status_t startInput(audio_io_handle_t input);
+ virtual status_t stopInput(audio_io_handle_t input);
+ virtual void releaseInput(audio_io_handle_t input);
+ virtual status_t initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
+ virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
+
+ virtual status_t onTransact(
+ uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags);
+
+ // IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder>& who);
+
+ //
+ // AudioPolicyClientInterface
+ //
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ AudioSystem::output_flags flags);
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2);
+ virtual status_t closeOutput(audio_io_handle_t output);
+ virtual status_t suspendOutput(audio_io_handle_t output);
+ virtual status_t restoreOutput(audio_io_handle_t output);
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics);
+ virtual status_t closeInput(audio_io_handle_t input);
+ virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output);
+ virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output);
+ virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+ virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
+ virtual status_t stopTone();
+
+private:
+ AudioPolicyService();
+ virtual ~AudioPolicyService();
+
+ static const char *sAudioPolicyLibrary;
+ static const char *sAudioPolicyGenericLibrary;
+ // Thread used for tone playback and to send audio config commands to audio flinger
+ // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone()
+ // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause
+ // calls to AudioPolicyService and an attempt to lock mLock.
+ // For audio config commands, it is necessary because audio flinger requires that the calling process (user)
+ // has permission to modify audio settings.
+ class AudioCommandThread : public Thread {
+ public:
+
+ // commands for tone AudioCommand
+ enum {
+ START_TONE,
+ STOP_TONE,
+ SET_VOLUME,
+ SET_PARAMETERS
+ };
+
+ AudioCommandThread ();
+ virtual ~AudioCommandThread();
+
+ // Thread virtuals
+ virtual void onFirstRef();
+ virtual bool threadLoop();
+
+ void exit();
+ void startToneCommand(int type = 0, int stream = 0);
+ void stopToneCommand();
+ status_t volumeCommand(int stream, float volume, void *output);
+ status_t parametersCommand(void *ioHandle, const String8& keyValuePairs);
+
+ private:
+ // descriptor for requested tone playback event
+ class AudioCommand {
+ public:
+ int mCommand; // START_TONE, STOP_TONE ...
+ void *mParam;
+ };
+
+ class ToneData {
+ public:
+ int mType; // tone type (START_TONE only)
+ int mStream; // stream type (START_TONE only)
+ };
+
+ class VolumeData {
+ public:
+ int mStream;
+ float mVolume;
+ void *mIO;
+ };
+ class ParametersData {
+ public:
+ void *mIO;
+ String8 mKeyValuePairs;
+ };
+
+
+ Mutex mLock;
+ Condition mWaitWorkCV;
+ Vector<AudioCommand *> mAudioCommands; // list of pending tone events
+ Condition mCommandCond;
+ status_t mCommandStatus;
+ ToneGenerator *mpToneGenerator; // the tone generator
+ };
+
+ // Internal dump utilities.
+ status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
+
+
+ Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing device
+ // connection stated our routing
+ AudioPolicyInterface* mpPolicyManager; // the platform specific policy manager
+ sp <AudioCommandThread> mAudioCommandThread; // audio commands thread
+ sp <AudioCommandThread> mTonePlaybacThread; // tone playback thread
+ void *mpPolicyManagerLibHandle;
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYSERVICE_H
+
+
+
+
+
+
+
+
generated by cgit v1.1 at 2024-10-27 08:37:44 +0000