summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--include/media/AudioSystem.h6
-rw-r--r--include/media/EffectApi.h6
-rw-r--r--include/media/mediarecorder.h3
-rw-r--r--media/java/android/media/AudioManager.java7
-rw-r--r--media/java/android/media/AudioRecord.java3
-rw-r--r--media/java/android/media/AudioService.java3
-rw-r--r--media/java/android/media/AudioSystem.java3
-rw-r--r--media/java/android/media/MediaRecorder.java9
-rw-r--r--media/libmedia/AudioSystem.cpp1
-rw-r--r--services/audioflinger/AudioFlinger.cpp3
-rw-r--r--services/audioflinger/AudioHardwareInterface.cpp5
-rw-r--r--services/audioflinger/AudioPolicyManagerBase.cpp52
-rw-r--r--telephony/java/com/android/internal/telephony/CallManager.java15
-rw-r--r--voip/jni/rtp/AudioGroup.cpp10
14 files changed, 87 insertions, 39 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 9fd905f..e881747 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -156,6 +156,7 @@ public:
MODE_NORMAL = 0,
MODE_RINGTONE,
MODE_IN_CALL,
+ MODE_IN_COMMUNICATION,
NUM_MODES // not a valid entry, denotes end-of-list
};
@@ -466,7 +467,7 @@ public:
AudioParameter(const String8& keyValuePairs);
virtual ~AudioParameter();
- // reserved parameter keys for changeing standard parameters with setParameters() function.
+ // reserved parameter keys for changing standard parameters with setParameters() function.
// Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
// configuration changes and act accordingly.
// keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
@@ -474,11 +475,14 @@ public:
// keyFormat: to change audio format, value is an int in AudioSystem::audio_format
// keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
// keyFrameCount: to change audio output frame count, value is an int
+ // keyInputSource: to change audio input source, value is an int in audio_source
+ // (defined in media/mediarecorder.h)
static const char *keyRouting;
static const char *keySamplingRate;
static const char *keyFormat;
static const char *keyChannels;
static const char *keyFrameCount;
+ static const char *keyInputSource;
String8 toString();
diff --git a/include/media/EffectApi.h b/include/media/EffectApi.h
index 16fb43c..b97c22e 100644
--- a/include/media/EffectApi.h
+++ b/include/media/EffectApi.h
@@ -602,9 +602,9 @@ enum audio_device_e {
// Audio mode
enum audio_mode_e {
- AUDIO_MODE_NORMAL, // phone idle
- AUDIO_MODE_RINGTONE, // phone ringing
- AUDIO_MODE_IN_CALL // phone call connected
+ AUDIO_MODE_NORMAL, // device idle
+ AUDIO_MODE_RINGTONE, // device ringing
+ AUDIO_MODE_IN_CALL // audio call connected (VoIP or telephony)
};
// Values for "accessMode" field of buffer_config_t:
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 5ab1640..9a76393 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -44,7 +44,8 @@ enum audio_source {
AUDIO_SOURCE_VOICE_CALL = 4,
AUDIO_SOURCE_CAMCORDER = 5,
AUDIO_SOURCE_VOICE_RECOGNITION = 6,
- AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_RECOGNITION,
+ AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
+ AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_COMMUNICATION,
AUDIO_SOURCE_LIST_END // must be last - used to validate audio source type
};
diff --git a/media/java/android/media/AudioManager.java b/media/java/android/media/AudioManager.java
index bbbba74..4459505 100644
--- a/media/java/android/media/AudioManager.java
+++ b/media/java/android/media/AudioManager.java
@@ -966,9 +966,14 @@ public class AudioManager {
*/
public static final int MODE_RINGTONE = AudioSystem.MODE_RINGTONE;
/**
- * In call audio mode. A call is established.
+ * In call audio mode. A telephony call is established.
*/
public static final int MODE_IN_CALL = AudioSystem.MODE_IN_CALL;
+ /**
+ * @hide
+ * In communication audio mode. An audio/video chat or VoIP call is established.
+ */
+ public static final int MODE_IN_COMMUNICATION = AudioSystem.MODE_IN_COMMUNICATION;
/* Routing bits for setRouting/getRouting API */
/**
diff --git a/media/java/android/media/AudioRecord.java b/media/java/android/media/AudioRecord.java
index c48eaad..0eb39db 100644
--- a/media/java/android/media/AudioRecord.java
+++ b/media/java/android/media/AudioRecord.java
@@ -252,7 +252,8 @@ public class AudioRecord
//--------------
// audio source
if ( (audioSource < MediaRecorder.AudioSource.DEFAULT) ||
- (audioSource > MediaRecorder.getAudioSourceMax()) ) {
+ (audioSource > MediaRecorder.AudioSource.VOICE_COMMUNICATION) ) {
+ //(audioSource > MediaRecorder.getAudioSourceMax()) ) {
throw (new IllegalArgumentException("Invalid audio source."));
} else {
mRecordSource = audioSource;
diff --git a/media/java/android/media/AudioService.java b/media/java/android/media/AudioService.java
index 41d2cc5..06e00c5 100644
--- a/media/java/android/media/AudioService.java
+++ b/media/java/android/media/AudioService.java
@@ -708,7 +708,7 @@ public class AudioService extends IAudioService.Stub {
return;
}
- if (mode < AudioSystem.MODE_CURRENT || mode > AudioSystem.MODE_IN_CALL) {
+ if (mode < AudioSystem.MODE_CURRENT || mode >= AudioSystem.NUM_MODES) {
return;
}
@@ -2259,6 +2259,7 @@ public class AudioService extends IAudioService.Stub {
// add modify the phone app to take advantage of the new API
synchronized(mRingingLock) {
if (mIsRinging || (getMode() == AudioSystem.MODE_IN_CALL) ||
+ (getMode() == AudioSystem.MODE_IN_COMMUNICATION) ||
(getMode() == AudioSystem.MODE_RINGTONE) ) {
return;
}
diff --git a/media/java/android/media/AudioSystem.java b/media/java/android/media/AudioSystem.java
index a4818ff..4ddbb35 100644
--- a/media/java/android/media/AudioSystem.java
+++ b/media/java/android/media/AudioSystem.java
@@ -106,7 +106,8 @@ public class AudioSystem
public static final int MODE_NORMAL = 0;
public static final int MODE_RINGTONE = 1;
public static final int MODE_IN_CALL = 2;
- public static final int NUM_MODES = 3;
+ public static final int MODE_IN_COMMUNICATION = 3;
+ public static final int NUM_MODES = 4;
/* Routing bits for setRouting/getRouting API */
diff --git a/media/java/android/media/MediaRecorder.java b/media/java/android/media/MediaRecorder.java
index 32c02bc..d9f2302 100644
--- a/media/java/android/media/MediaRecorder.java
+++ b/media/java/android/media/MediaRecorder.java
@@ -145,6 +145,15 @@ public class MediaRecorder
/** Microphone audio source tuned for voice recognition if available, behaves like
* {@link #DEFAULT} otherwise. */
public static final int VOICE_RECOGNITION = 6;
+
+ /**
+ * @hide
+ * Microphone audio source tuned for voice communications such as VoIP. It
+ * will for instance take advantage of echo cancellation or automatic gain control
+ * if available. It otherwise behaves like {@link #DEFAULT} if no voice processing
+ * is available.
+ */
+ public static final int VOICE_COMMUNICATION = 7;
}
/**
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9c2a8ba..1a3fcd6 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -835,6 +835,7 @@ const char *AudioParameter::keySamplingRate = "sampling_rate";
const char *AudioParameter::keyFormat = "format";
const char *AudioParameter::keyChannels = "channels";
const char *AudioParameter::keyFrameCount = "frame_count";
+const char *AudioParameter::keyInputSource = "input_source";
AudioParameter::AudioParameter(const String8& keyValuePairs)
{
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 5071555..5935bf9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -5810,7 +5810,8 @@ uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t devic
const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL
AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
- AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL
+ AUDIO_MODE_IN_CALL, // AudioSystem::MODE_IN_CALL
+ AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_COMMUNICATION, same conversion as for MODE_IN_CALL
};
int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
diff --git a/services/audioflinger/AudioHardwareInterface.cpp b/services/audioflinger/AudioHardwareInterface.cpp
index 9a4a7f9..f58e4c0 100644
--- a/services/audioflinger/AudioHardwareInterface.cpp
+++ b/services/audioflinger/AudioHardwareInterface.cpp
@@ -48,14 +48,15 @@ static const char* routingModeStrings[] =
"CURRENT",
"NORMAL",
"RINGTONE",
- "IN_CALL"
+ "IN_CALL",
+ "IN_COMMUNICATION"
};
static const char* routeNone = "NONE";
static const char* displayMode(int mode)
{
- if ((mode < -2) || (mode > 2))
+ if ((mode < AudioSystem::MODE_INVALID) || (mode >= AudioSystem::NUM_MODES))
return routingModeStrings[0];
return routingModeStrings[mode+3];
}
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
index 8d16ab4..0da353a 100644
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ b/services/audioflinger/AudioPolicyManagerBase.cpp
@@ -235,7 +235,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
// if leaving call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ if (isInCall()) {
LOGV("setPhoneState() in call state management: new state is %d", state);
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
handleIncallSonification(stream, false, true);
@@ -248,16 +248,21 @@ void AudioPolicyManagerBase::setPhoneState(int state)
bool force = false;
// are we entering or starting a call
- if ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) {
+ if (!isStateInCall(oldState) && isStateInCall(state)) {
LOGV(" Entering call in setPhoneState()");
// force routing command to audio hardware when starting a call
// even if no device change is needed
force = true;
- } else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) {
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
LOGV(" Exiting call in setPhoneState()");
// force routing command to audio hardware when exiting a call
// even if no device change is needed
force = true;
+ } else if (isStateInCall(state) && (state != oldState)) {
+ LOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
}
// check for device and output changes triggered by new phone state
@@ -272,7 +277,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
// force routing command to audio hardware when ending call
// even if no device change is needed
- if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) {
+ if (isStateInCall(oldState) && newDevice == 0) {
newDevice = hwOutputDesc->device();
}
@@ -280,7 +285,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
// immediately and delay the route change to avoid sending the ring tone
// tail into the earpiece or headset.
int delayMs = 0;
- if (state == AudioSystem::MODE_IN_CALL && oldState == AudioSystem::MODE_RINGTONE) {
+ if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
// delay the device change command by twice the output latency to have some margin
// and be sure that audio buffers not yet affected by the mute are out when
// we actually apply the route change
@@ -293,7 +298,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
// if entering in call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
- if (state == AudioSystem::MODE_IN_CALL) {
+ if (isStateInCall(state)) {
LOGV("setPhoneState() in call state management: new state is %d", state);
// unmute the ringing tone after a sufficient delay if it was muted before
// setting output device above
@@ -564,7 +569,7 @@ status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
setOutputDevice(output, getNewDevice(output));
// handle special case for sonification while in call
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ if (isInCall()) {
handleIncallSonification(stream, true, false);
}
@@ -589,7 +594,7 @@ status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
// handle special case for sonification while in call
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ if (isInCall()) {
handleIncallSonification(stream, false, false);
}
@@ -738,10 +743,8 @@ status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
- // use Voice Recognition mode or not for this input based on input source
- int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
- param.addInt(String8("vr_mode"), vr_enabled);
- LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
+ param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
+ LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
mpClientInterface->setParameters(input, param.toString());
@@ -1494,7 +1497,7 @@ uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fro
// use device for strategy media
// 4: the strategy DTMF is active on the hardware output:
// use device for strategy DTMF
- if (mPhoneState == AudioSystem::MODE_IN_CALL ||
+ if (isInCall() ||
outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
} else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
@@ -1549,7 +1552,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
switch (strategy) {
case STRATEGY_DTMF:
- if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ if (!isInCall()) {
// when off call, DTMF strategy follows the same rules as MEDIA strategy
device = getDeviceForStrategy(STRATEGY_MEDIA, false);
break;
@@ -1562,7 +1565,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
// of priority
switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
case AudioSystem::FORCE_BT_SCO:
- if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
if (device) break;
}
@@ -1580,7 +1583,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
if (device) break;
#ifdef WITH_A2DP
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ if (!isInCall()) {
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
if (device) break;
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
@@ -1594,14 +1597,14 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
break;
case AudioSystem::FORCE_SPEAKER:
- if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
if (device) break;
}
#ifdef WITH_A2DP
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
// A2DP speaker when forcing to speaker output
- if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ if (!isInCall()) {
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
if (device) break;
}
@@ -1618,7 +1621,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
// If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
// handleIncallSonification().
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ if (isInCall()) {
device = getDeviceForStrategy(STRATEGY_PHONE, false);
break;
}
@@ -1739,6 +1742,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
case AUDIO_SOURCE_DEFAULT:
case AUDIO_SOURCE_MIC:
case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
@@ -1960,6 +1964,16 @@ void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting,
}
}
+bool AudioPolicyManagerBase::isInCall()
+{
+ return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManagerBase::isStateInCall(int state) {
+ return ((state == AudioSystem::MODE_IN_CALL) ||
+ (state == AudioSystem::MODE_IN_COMMUNICATION));
+}
+
bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
uint32_t samplingRate,
uint32_t format,
diff --git a/telephony/java/com/android/internal/telephony/CallManager.java b/telephony/java/com/android/internal/telephony/CallManager.java
index 0335ca2..b89058f 100644
--- a/telephony/java/com/android/internal/telephony/CallManager.java
+++ b/telephony/java/com/android/internal/telephony/CallManager.java
@@ -380,12 +380,17 @@ public final class CallManager {
break;
case OFFHOOK:
Phone fgPhone = getFgPhone();
- // Enable IN_CALL mode while foreground call is in DIALING,
- // ALERTING, ACTIVE and DISCONNECTING state and not from sipPhone
+ // While foreground call is in DIALING,
+ // ALERTING, ACTIVE and DISCONNECTING state
if (getActiveFgCallState() != Call.State.IDLE
- && getActiveFgCallState() != Call.State.DISCONNECTED
- && !(fgPhone instanceof SipPhone)) {
- mode = AudioManager.MODE_IN_CALL;
+ && getActiveFgCallState() != Call.State.DISCONNECTED) {
+ if (fgPhone instanceof SipPhone) {
+ // enable IN_COMMUNICATION audio mode for sipPhone
+ mode = AudioManager.MODE_IN_COMMUNICATION;
+ } else {
+ // enable IN_CALL audio mode for telephony
+ mode = AudioManager.MODE_IN_CALL;
+ }
}
break;
}
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 60abf2a..2cbd023 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -483,7 +483,7 @@ public:
ON_HOLD = 0,
MUTED = 1,
NORMAL = 2,
- EC_ENABLED = 3,
+ ECHO_SUPPRESSION = 3,
LAST_MODE = 3,
};
@@ -619,6 +619,10 @@ bool AudioGroup::setMode(int mode)
if (mode < 0 || mode > LAST_MODE) {
return false;
}
+ if (mode == ECHO_SUPPRESSION && AudioSystem::getParameters(
+ 0, String8("ec_supported")) == "ec_supported=yes") {
+ mode = NORMAL;
+ }
if (mMode == mode) {
return true;
}
@@ -775,8 +779,8 @@ bool AudioGroup::DeviceThread::threadLoop()
AudioTrack track;
AudioRecord record;
if (track.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
- AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR ||
- record.set(AUDIO_SOURCE_MIC, sampleRate, AudioSystem::PCM_16_BIT,
+ AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR || record.set(
+ AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AudioSystem::PCM_16_BIT,
AudioSystem::CHANNEL_IN_MONO, input) != NO_ERROR) {
LOGE("cannot initialize audio device");
return false;
generated by cgit v1.1 at 2025-07-03 09:57:01 +0000