Revert^2 "Reapply "AudioFlinger: Control volume using Port ID""
This reverts commit d3e99d20da87b3a8f0bae64e16f916c0fde85098.
Reason for revert: Internal project fixed with ag/28965993
Test: Treehugger
Bug: 361842578
Change-Id: I76427c29383fa1676de2c3635803a4f16ad9d552
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 2abf682..e5ec5d8 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -146,6 +146,7 @@
"audioflinger-aidl-cpp",
"av-types-aidl-cpp",
"com.android.media.audio-aconfig-cc",
+ "com.android.media.audioserver-aconfig-cc",
"effect-aidl-cpp",
"libactivitymanager_aidl",
"libaudioclient",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 20cd40c..b2c9e32 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -187,6 +187,7 @@
BINDER_METHOD_ENTRY(masterMute) \
BINDER_METHOD_ENTRY(setStreamVolume) \
BINDER_METHOD_ENTRY(setStreamMute) \
+BINDER_METHOD_ENTRY(setPortsVolume) \
BINDER_METHOD_ENTRY(setMode) \
BINDER_METHOD_ENTRY(setMicMute) \
BINDER_METHOD_ENTRY(getMicMute) \
@@ -617,6 +618,7 @@
std::vector<audio_io_handle_t> secondaryOutputs;
bool isSpatialized;
bool isBitPerfect;
+ float volume;
ret = AudioSystem::getOutputForAttr(&localAttr, &io,
actualSessionId,
&streamType, adjAttributionSource,
@@ -624,7 +626,8 @@
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
deviceId, &portId, &secondaryOutputs, &isSpatialized,
- &isBitPerfect);
+ &isBitPerfect,
+ &volume);
if (ret != NO_ERROR) {
config->sample_rate = fullConfig.sample_rate;
config->channel_mask = fullConfig.channel_mask;
@@ -1061,6 +1064,7 @@
std::vector<audio_io_handle_t> secondaryOutputs;
bool isSpatialized = false;
bool isBitPerfect = false;
+ float volume;
audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
std::vector<int> effectIds;
@@ -1121,7 +1125,7 @@
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
adjAttributionSource, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs,
- &isSpatialized, &isBitPerfect);
+ &isSpatialized, &isBitPerfect, &volume);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
@@ -1178,7 +1182,7 @@
if (effectThread == nullptr) {
effectChain = getOrphanEffectChain_l(sessionId);
}
- ALOGV("createTrack() sessionId: %d", sessionId);
+ ALOGV("createTrack() sessionId: %d volume: %f", sessionId, volume);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
@@ -1193,7 +1197,7 @@
input.sharedBuffer, sessionId, &output.flags,
callingPid, adjAttributionSource, input.clientInfo.clientTid,
&lStatus, portId, input.audioTrackCallback, isSpatialized,
- isBitPerfect, &output.afTrackFlags);
+ isBitPerfect, &output.afTrackFlags, volume);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
@@ -1644,6 +1648,33 @@
return NO_ERROR;
}
+status_t AudioFlinger::setPortsVolume(
+ const std::vector<audio_port_handle_t>& ports, float volume, audio_io_handle_t output)
+{
+ for (const auto& port : ports) {
+ if (port == AUDIO_PORT_HANDLE_NONE) {
+ return BAD_VALUE;
+ }
+ }
+ if (isnan(volume) || volume > 1.0f || volume < 0.0f) {
+ return BAD_VALUE;
+ }
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ return BAD_VALUE;
+ }
+ audio_utils::lock_guard lock(mutex());
+ IAfPlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread != nullptr) {
+ return thread->setPortsVolume(ports, volume);
+ }
+ const sp<IAfMmapThread> mmapThread = checkMmapThread_l(output);
+ if (mmapThread != nullptr && mmapThread->isOutput()) {
+ IAfMmapPlaybackThread *mmapPlaybackThread = mmapThread->asIAfMmapPlaybackThread().get();
+ return mmapPlaybackThread->setPortsVolume(ports, volume);
+ }
+ return BAD_VALUE;
+}
+
status_t AudioFlinger::setRequestedLatencyMode(
audio_io_handle_t output, audio_latency_mode_t mode) {
if (output == AUDIO_IO_HANDLE_NONE) {
@@ -3824,8 +3855,7 @@
// checkPlaybackThread_l() must be called with AudioFlinger::mutex() held
-sp<VolumeInterface> AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const
-{
+sp<VolumeInterface> AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const {
sp<VolumeInterface> volumeInterface = mPlaybackThreads.valueFor(output).get();
if (volumeInterface == nullptr) {
IAfMmapThread* const mmapThread = mMmapThreads.valueFor(output).get();
@@ -4020,7 +4050,8 @@
outputFlags,
0ns /* timeout */,
frameCountToBeReady,
- track->getSpeed());
+ track->getSpeed(),
+ track->getPortVolume());
status = patchTrack->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchTrack init failed: %d", status);
@@ -5119,6 +5150,7 @@
case TransactionCode::GET_AUDIO_MIX_PORT:
case TransactionCode::SET_TRACKS_INTERNAL_MUTE:
case TransactionCode::RESET_REFERENCES_FOR_TEST:
+ case TransactionCode::SET_PORTS_VOLUME:
ALOGW("%s: transaction %d received from PID %d",
__func__, static_cast<int>(code), IPCThreadState::self()->getCallingPid());
// return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index adec4aa..902df0a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -96,6 +96,9 @@
status_t setStreamMute(audio_stream_type_t stream, bool muted) final
EXCLUDES_AudioFlinger_Mutex;
+ status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume,
+ audio_io_handle_t output) final EXCLUDES_AudioFlinger_Mutex;
+
status_t setMode(audio_mode_t mode) final EXCLUDES_AudioFlinger_Mutex;
status_t setMicMute(bool state) final EXCLUDES_AudioFlinger_Mutex;
@@ -551,6 +554,7 @@
IAfPlaybackThread* checkMixerThread_l(audio_io_handle_t output) const REQUIRES(mutex());
sp<VolumeInterface> getVolumeInterface_l(audio_io_handle_t output) const REQUIRES(mutex());
+
std::vector<sp<VolumeInterface>> getAllVolumeInterfaces_l() const REQUIRES(mutex());
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index 4d26aa0..8596acb 100644
--- a/services/audioflinger/IAfThread.h
+++ b/services/audioflinger/IAfThread.h
@@ -26,6 +26,7 @@
#include <datapath/AudioStreamIn.h>
#include <datapath/AudioStreamOut.h>
#include <datapath/VolumeInterface.h>
+#include <datapath/VolumePortInterface.h>
#include <fastpath/FastMixerDumpState.h>
#include <media/DeviceDescriptorBase.h>
#include <media/MmapStreamInterface.h>
@@ -479,7 +480,8 @@
const sp<media::IAudioTrackCallback>& callback,
bool isSpatialized,
bool isBitPerfect,
- audio_output_flags_t* afTrackFlags)
+ audio_output_flags_t* afTrackFlags,
+ float volume)
REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
virtual status_t addTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex()) = 0;
@@ -555,6 +557,9 @@
virtual void setTracksInternalMute(std::map<audio_port_handle_t, bool>* tracksInternalMute)
EXCLUDES_ThreadBase_Mutex = 0;
+
+ virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+ EXCLUDES_ThreadBase_Mutex = 0;
};
class IAfDirectOutputThread : public virtual IAfPlaybackThread {
@@ -694,6 +699,9 @@
AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady);
virtual AudioStreamOut* clearOutput() EXCLUDES_ThreadBase_Mutex = 0;
+
+ virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+ EXCLUDES_ThreadBase_Mutex = 0;
};
class IAfMmapCaptureThread : public virtual IAfMmapThread {
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index a9c87ad..ee834d6 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -21,6 +21,7 @@
#include <audio_utils/mutex.h>
#include <audiomanager/IAudioManager.h>
#include <binder/IMemory.h>
+#include <datapath/VolumePortInterface.h>
#include <fastpath/FastMixerDumpState.h>
#include <media/AudioSystem.h>
#include <media/VolumeShaper.h>
@@ -254,7 +255,7 @@
};
// Common interface for Playback tracks.
-class IAfTrack : public virtual IAfTrackBase {
+class IAfTrack : public virtual IAfTrackBase, public virtual VolumePortInterface {
public:
// FillingStatus is used for suppressing volume ramp at begin of playing
enum FillingStatus { FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE };
@@ -289,7 +290,8 @@
size_t frameCountToBeReady = SIZE_MAX,
float speed = 1.0f,
bool isSpatialized = false,
- bool isBitPerfect = false);
+ bool isBitPerfect = false,
+ float volume = 0.0f);
virtual void pause() = 0;
virtual void flush() = 0;
@@ -452,7 +454,7 @@
virtual ExtendedTimestamp getClientProxyTimestamp() const = 0;
};
-class IAfMmapTrack : public virtual IAfTrackBase {
+class IAfMmapTrack : public virtual IAfTrackBase, public virtual VolumePortInterface {
public:
static sp<IAfMmapTrack> create(IAfThreadBase* thread,
const audio_attributes_t& attr,
@@ -463,7 +465,8 @@
bool isOut,
const android::content::AttributionSourceState& attributionSource,
pid_t creatorPid,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+ float volume = 0.0f);
// protected by MMapThread::mLock
virtual void setSilenced_l(bool silenced) = 0;
@@ -583,7 +586,8 @@
* as soon as possible to have
* the lowest possible latency
* even if it might glitch. */
- float speed = 1.0f);
+ float speed = 1.0f,
+ float volume = 1.0f);
};
class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index 85ce142..8758bd0 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -35,7 +35,8 @@
bool isOut,
const android::content::AttributionSourceState& attributionSource,
pid_t creatorPid,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+ float volume = 0.0f);
~MmapTrack() override;
status_t initCheck() const final;
@@ -65,6 +66,13 @@
void processMuteEvent_l(const sp<IAudioManager>& audioManager,
mute_state_t muteState)
/* REQUIRES(MmapPlaybackThread::mLock) */ final;
+
+ // VolumePortInterface implementation
+ void setPortVolume(float volume) override {
+ mVolume = volume;
+ }
+ float getPortVolume() const override { return mVolume; }
+
private:
DISALLOW_COPY_AND_ASSIGN(MmapTrack);
@@ -87,6 +95,8 @@
/* GUARDED_BY(MmapPlaybackThread::mLock) */;
mute_state_t mMuteState
/* GUARDED_BY(MmapPlaybackThread::mLock) */;
+
+ float mVolume = 0.0f;
}; // end of Track
} // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index f57470f..06f7887 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -647,7 +647,8 @@
outputFlags,
{} /*timeout*/,
frameCountToBeReady,
- 1.0f);
+ 1.0f /*speed*/,
+ 1.0f /*volume*/);
status = mPlayback.checkTrack(tempPatchTrack.get());
if (status != NO_ERROR) {
return status;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 2cc6236..3edeee3 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -96,7 +96,8 @@
size_t frameCountToBeReady = SIZE_MAX,
float speed = 1.0f,
bool isSpatialized = false,
- bool isBitPerfect = false);
+ bool isBitPerfect = false,
+ float volume = 0.0f);
~Track() override;
status_t initCheck() const final;
void appendDumpHeader(String8& result) const final;
@@ -222,6 +223,13 @@
bool getInternalMute() const final { return mInternalMute; }
void setInternalMute(bool muted) final { mInternalMute = muted; }
+
+ // VolumePortInterface implementation
+ void setPortVolume(float volume) override {
+ mVolume = volume;
+ }
+ float getPortVolume() const override { return mVolume; }
+
protected:
DISALLOW_COPY_AND_ASSIGN(Track);
@@ -362,6 +370,8 @@
for (auto& tp : mTeePatches) { f(tp.patchTrack); }
};
+ void populateUsageAndContentTypeFromStreamType();
+
size_t mPresentationCompleteFrames = 0; // (Used for Mixed tracks)
// The number of frames written to the
// audio HAL when this track is considered fully rendered.
@@ -403,8 +413,8 @@
// access these two variables only when holding player thread lock.
std::unique_ptr<os::PersistableBundle> mMuteEventExtras;
mute_state_t mMuteState;
-
bool mInternalMute = false;
+ float mVolume = 0.0f;
}; // end of Track
@@ -501,7 +511,8 @@
* as soon as possible to have
* the lowest possible latency
* even if it might glitch. */
- float speed = 1.0f);
+ float speed = 1.0f,
+ float volume = 1.0f);
~PatchTrack() override;
size_t framesReady() const final;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7c7d812..6359846 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -49,6 +49,7 @@
#include <binder/IServiceManager.h>
#include <binder/PersistableBundle.h>
#include <com_android_media_audio.h>
+#include <com_android_media_audioserver.h>
#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <fastpath/AutoPark.h>
@@ -122,6 +123,7 @@
}
using com::android::media::permission::ValidatedAttributionSourceState;
+namespace audioserver_flags = com::android::media::audioserver;
namespace android {
@@ -2217,17 +2219,18 @@
(int64_t)(mIsMsdDevice ? AUDIO_DEVICE_OUT_BUS // turn on by default for MSD
: AUDIO_DEVICE_NONE));
}
-
- for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
- const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
- mStreamTypes[stream].volume = 0.0f;
- mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+ if (!audioserver_flags::portid_volume_management()) {
+ for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
+ const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
+ mStreamTypes[stream].volume = 0.0f;
+ mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+ }
+ // Audio patch and call assistant volume are always max
+ mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
}
- // Audio patch and call assistant volume are always max
- mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
- mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
- mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
- mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
}
PlaybackThread::~PlaybackThread()
@@ -2278,16 +2281,17 @@
void PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
-
- result.appendFormat(" Stream volumes in dB: ");
- for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
- const stream_type_t *st = &mStreamTypes[i];
- if (i > 0) {
- result.appendFormat(", ");
- }
- result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
- if (st->mute) {
- result.append("M");
+ if (!audioserver_flags::portid_volume_management()) {
+ result.appendFormat(" Stream volumes in dB: ");
+ for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
+ const stream_type_t *st = &mStreamTypes[i];
+ if (i > 0) {
+ result.appendFormat(", ");
+ }
+ result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
+ if (st->mute) {
+ result.append("M");
+ }
}
}
result.append("\n");
@@ -2395,7 +2399,8 @@
const sp<media::IAudioTrackCallback>& callback,
bool isSpatialized,
bool isBitPerfect,
- audio_output_flags_t *afTrackFlags)
+ audio_output_flags_t *afTrackFlags,
+ float volume)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
@@ -2724,7 +2729,7 @@
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, creatorPid, attributionSource, trackFlags,
IAfTrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
- speed, isSpatialized, isBitPerfect);
+ speed, isSpatialized, isBitPerfect, volume);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
@@ -2852,6 +2857,22 @@
return mStreamTypes[stream].volume;
}
+status_t PlaybackThread::setPortsVolume(
+ const std::vector<audio_port_handle_t>& portIds, float volume) {
+ audio_utils::lock_guard _l(mutex());
+ for (const auto& portId : portIds) {
+ for (size_t i = 0; i < mTracks.size(); i++) {
+ sp<IAfTrack> track = mTracks[i].get();
+ if (portId == track->portId()) {
+ track->setPortVolume(volume);
+ break;
+ }
+ }
+ }
+ broadcast_l();
+ return NO_ERROR;
+}
+
void PlaybackThread::setVolumeForOutput_l(float left, float right) const
{
mOutput->stream->setVolume(left, right);
@@ -5783,12 +5804,19 @@
}
sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
float volume;
- if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
- volume = 0.f;
+ if (!audioserver_flags::portid_volume_management()) {
+ if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
+ volume = 0.f;
+ } else {
+ volume = masterVolume * mStreamTypes[track->streamType()].volume;
+ }
} else {
- volume = masterVolume * mStreamTypes[track->streamType()].volume;
+ if (track->isPlaybackRestricted()) {
+ volume = 0.f;
+ } else {
+ volume = masterVolume * track->getPortVolume();
+ }
}
-
handleVoipVolume_l(&volume);
// cache the combined master volume and stream type volume for fast mixer; this
@@ -5800,15 +5828,23 @@
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
-
- track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
- /*muteState=*/{masterVolume == 0.f,
- mStreamTypes[track->streamType()].volume == 0.f,
- mStreamTypes[track->streamType()].mute,
- track->isPlaybackRestricted(),
- vlf == 0.f && vrf == 0.f,
- vh == 0.f});
-
+ if (!audioserver_flags::portid_volume_management()) {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{masterVolume == 0.f,
+ mStreamTypes[track->streamType()].volume == 0.f,
+ mStreamTypes[track->streamType()].mute,
+ track->isPlaybackRestricted(),
+ vlf == 0.f && vrf == 0.f,
+ vh == 0.f});
+ } else {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{masterVolume == 0.f,
+ track->getPortVolume() == 0.f,
+ /* muteFromStreamMuted= */ false,
+ track->isPlaybackRestricted(),
+ vlf == 0.f && vrf == 0.f,
+ vh == 0.f});
+ }
vlf *= volume;
vrf *= volume;
@@ -5959,16 +5995,22 @@
uint32_t vl, vr; // in U8.24 integer format
float vlf, vrf, vaf; // in [0.0, 1.0] float format
// read original volumes with volume control
- float v = masterVolume * mStreamTypes[track->streamType()].volume;
// Always fetch volumeshaper volume to ensure state is updated.
const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
const float vh = track->getVolumeHandler()->getVolume(
track->audioTrackServerProxy()->framesReleased()).first;
-
- if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
- v = 0;
+ float v;
+ if (!audioserver_flags::portid_volume_management()) {
+ v = masterVolume * mStreamTypes[track->streamType()].volume;
+ if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
+ v = 0;
+ }
+ } else {
+ v = masterVolume * track->getPortVolume();
+ if (track->isPlaybackRestricted()) {
+ v = 0;
+ }
}
-
handleVoipVolume_l(&v);
if (track->isPausing()) {
@@ -5988,15 +6030,23 @@
ALOGV("Track right volume out of range: %.3g", vrf);
vrf = GAIN_FLOAT_UNITY;
}
-
- track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
- /*muteState=*/{masterVolume == 0.f,
- mStreamTypes[track->streamType()].volume == 0.f,
- mStreamTypes[track->streamType()].mute,
- track->isPlaybackRestricted(),
- vlf == 0.f && vrf == 0.f,
- vh == 0.f});
-
+ if (!audioserver_flags::portid_volume_management()) {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{masterVolume == 0.f,
+ mStreamTypes[track->streamType()].volume == 0.f,
+ mStreamTypes[track->streamType()].mute,
+ track->isPlaybackRestricted(),
+ vlf == 0.f && vrf == 0.f,
+ vh == 0.f});
+ } else {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{masterVolume == 0.f,
+ track->getPortVolume() == 0.f,
+ /* muteFromStreamMuted= */ false,
+ track->isPlaybackRestricted(),
+ vlf == 0.f && vrf == 0.f,
+ vh == 0.f});
+ }
// now apply the master volume and stream type volume and shaper volume
vlf *= v * vh;
vrf *= v * vh;
@@ -6722,34 +6772,64 @@
const bool clientVolumeMute = (left == 0.f && right == 0.f);
- if (mMasterMute || mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
- left = right = 0;
- } else {
- float typeVolume = mStreamTypes[track->streamType()].volume;
- const float v = mMasterVolume * typeVolume * shaperVolume;
+ if (!audioserver_flags::portid_volume_management()) {
+ if (mMasterMute || mStreamTypes[track->streamType()].mute ||
+ track->isPlaybackRestricted()) {
+ left = right = 0;
+ } else {
+ float typeVolume = mStreamTypes[track->streamType()].volume;
+ const float v = mMasterVolume * typeVolume * shaperVolume;
- if (left > GAIN_FLOAT_UNITY) {
- left = GAIN_FLOAT_UNITY;
- }
- if (right > GAIN_FLOAT_UNITY) {
- right = GAIN_FLOAT_UNITY;
- }
- left *= v;
- right *= v;
- if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
+ if (left > GAIN_FLOAT_UNITY) {
+ left = GAIN_FLOAT_UNITY;
+ }
+ if (right > GAIN_FLOAT_UNITY) {
+ right = GAIN_FLOAT_UNITY;
+ }
+ left *= v;
+ right *= v;
+ if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
|| audio_channel_count_from_out_mask(mChannelMask) > 1) {
- left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
- right *= mMasterBalanceRight;
+ left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
+ right *= mMasterBalanceRight;
+ }
}
- }
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{mMasterMute,
+ mStreamTypes[track->streamType()].volume == 0.f,
+ mStreamTypes[track->streamType()].mute,
+ track->isPlaybackRestricted(),
+ clientVolumeMute,
+ shaperVolume == 0.f});
+ } else {
+ if (mMasterMute || track->isPlaybackRestricted()) {
+ left = right = 0;
+ } else {
+ float typeVolume = track->getPortVolume();
+ const float v = mMasterVolume * typeVolume * shaperVolume;
- track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
- /*muteState=*/{mMasterMute,
- mStreamTypes[track->streamType()].volume == 0.f,
- mStreamTypes[track->streamType()].mute,
- track->isPlaybackRestricted(),
- clientVolumeMute,
- shaperVolume == 0.f});
+ if (left > GAIN_FLOAT_UNITY) {
+ left = GAIN_FLOAT_UNITY;
+ }
+ if (right > GAIN_FLOAT_UNITY) {
+ right = GAIN_FLOAT_UNITY;
+ }
+ left *= v;
+ right *= v;
+ if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
+ || audio_channel_count_from_out_mask(mChannelMask) > 1) {
+ left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
+ right *= mMasterBalanceRight;
+ }
+ }
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{mMasterMute,
+ track->getPortVolume() == 0.f,
+ /* muteFromStreamMuted= */ false,
+ track->isPlaybackRestricted(),
+ clientVolumeMute,
+ shaperVolume == 0.f});
+ }
if (lastTrack) {
track->setFinalVolume(left, right);
@@ -7843,7 +7923,9 @@
ALOGE("addOutputTrack() initCheck failed %d", status);
return;
}
- thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
+ if (!audioserver_flags::portid_volume_management()) {
+ thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
+ }
mOutputTracks.add(outputTrack);
ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
@@ -10330,6 +10412,7 @@
const auto localSessionId = mSessionId;
auto localAttr = mAttr;
+ float volume = 0.0f;
if (isOutput()) {
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = mSampleRate;
@@ -10353,7 +10436,8 @@
&portId,
&secondaryOutputs,
&isSpatialized,
- &isBitPerfect);
+ &isBitPerfect,
+ &volume);
mutex().lock();
mAttr = localAttr;
ALOGD_IF(!secondaryOutputs.empty(),
@@ -10422,7 +10506,8 @@
this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
mChannelMask, mSessionId, isOutput(),
client.attributionSource,
- IPCThreadState::self()->getCallingPid(), portId);
+ IPCThreadState::self()->getCallingPid(), portId,
+ volume);
if (!isOutput()) {
track->setSilenced_l(isClientSilenced_l(portId));
}
@@ -11007,18 +11092,18 @@
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mMasterVolume = afThreadCallback->masterVolume_l();
mMasterMute = afThreadCallback->masterMute_l();
-
- for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
- const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
- mStreamTypes[stream].volume = 0.0f;
- mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+ if (!audioserver_flags::portid_volume_management()) {
+ for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
+ const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
+ mStreamTypes[stream].volume = 0.0f;
+ mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+ }
+ // Audio patch and call assistant volume are always max
+ mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
}
- // Audio patch and call assistant volume are always max
- mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
- mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
- mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
- mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
-
if (mAudioHwDev) {
if (mAudioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
@@ -11097,6 +11182,21 @@
}
}
+status_t MmapPlaybackThread::setPortsVolume(
+ const std::vector<audio_port_handle_t>& portIds, float volume) {
+ audio_utils::lock_guard _l(mutex());
+ for (const auto& portId : portIds) {
+ for (const sp<IAfMmapTrack>& track : mActiveTracks) {
+ if (portId == track->portId()) {
+ track->setPortVolume(volume);
+ break;
+ }
+ }
+ }
+ broadcast_l();
+ return NO_ERROR;
+}
+
void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
audio_utils::lock_guard _l(mutex());
@@ -11130,14 +11230,26 @@
void MmapPlaybackThread::processVolume_l()
NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
{
- float volume;
-
- if (mMasterMute || streamMuted_l()) {
- volume = 0;
+ float volume = 0;
+ if (!audioserver_flags::portid_volume_management()) {
+ if (mMasterMute || streamMuted_l()) {
+ volume = 0;
+ } else {
+ volume = mMasterVolume * streamVolume_l();
+ }
} else {
- volume = mMasterVolume * streamVolume_l();
+ if (mMasterMute) {
+ volume = 0;
+ } else {
+ // All mmap tracks are declared with the same audio attributes to the audio policy
+ // manager. Hence, they follow the same routing / volume group. Any change of volume
+ // will be broadcasted to all tracks. Thus, take arbitrarily first track volume.
+ size_t numtracks = mActiveTracks.size();
+ if (numtracks) {
+ volume = mMasterVolume * mActiveTracks[0]->getPortVolume();
+ }
+ }
}
-
if (volume != mHalVolFloat) {
// Convert volumes from float to 8.24
uint32_t vol = (uint32_t)(volume * (1 << 24));
@@ -11170,14 +11282,25 @@
}
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
track->setMetadataHasChanged();
- track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
- /*muteState=*/{mMasterMute,
- streamVolume_l() == 0.f,
- streamMuted_l(),
- // TODO(b/241533526): adjust logic to include mute from AppOps
- false /*muteFromPlaybackRestricted*/,
- false /*muteFromClientVolume*/,
- false /*muteFromVolumeShaper*/});
+ if (!audioserver_flags::portid_volume_management()) {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{mMasterMute,
+ streamVolume_l() == 0.f,
+ streamMuted_l(),
+ // TODO(b/241533526): adjust logic to include mute from AppOps
+ false /*muteFromPlaybackRestricted*/,
+ false /*muteFromClientVolume*/,
+ false /*muteFromVolumeShaper*/});
+ } else {
+ track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+ /*muteState=*/{mMasterMute,
+ track->getPortVolume() == 0.f,
+ /* muteFromStreamMuted= */ false,
+ // TODO(b/241533526): adjust logic to include mute from AppOps
+ false /*muteFromPlaybackRestricted*/,
+ false /*muteFromClientVolume*/,
+ false /*muteFromVolumeShaper*/});
+ }
}
}
}
@@ -11284,9 +11407,13 @@
void MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MmapThread::dumpInternals_l(fd, args);
-
- dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
- mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
+ if (!audioserver_flags::portid_volume_management()) {
+ dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d",
+ mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
+ } else {
+ dprintf(fd, " HAL volume: %f", mHalVolFloat);
+ }
+ dprintf(fd, "\n");
dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 654b841..89e41c8 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -836,6 +836,12 @@
typename SortedVector<sp<T>>::iterator end() {
return mActiveTracks.end();
}
+ typename SortedVector<const sp<T>>::iterator begin() const {
+ return mActiveTracks.begin();
+ }
+ typename SortedVector<const sp<T>>::iterator end() const {
+ return mActiveTracks.end();
+ }
// Due to Binder recursion optimization, clear() and updatePowerState()
// cannot be called from a Binder thread because they may call back into
@@ -1011,6 +1017,9 @@
void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
+ status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+ final EXCLUDES_ThreadBase_Mutex;
+
void setVolumeForOutput_l(float left, float right) const final;
sp<IAfTrack> createTrack_l(
@@ -1035,7 +1044,8 @@
const sp<media::IAudioTrackCallback>& callback,
bool isSpatialized,
bool isBitPerfect,
- audio_output_flags_t* afTrackFlags) final
+ audio_output_flags_t* afTrackFlags,
+ float volume) final
REQUIRES(audio_utils::AudioFlinger_Mutex);
bool isTrackActive(const sp<IAfTrack>& track) const final {
@@ -2385,6 +2395,8 @@
void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
+ status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+ final EXCLUDES_ThreadBase_Mutex;
void setMasterMute_l(bool muted) REQUIRES(mutex()) { mMasterMute = muted; }
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index a0b85f7..1342b7b 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -347,7 +347,7 @@
size_t mBufferSize; // size of mBuffer in bytes
// we don't really need a lock for these
MirroredVariable<track_state> mState;
- const audio_attributes_t mAttr;
+ audio_attributes_t mAttr;
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
const audio_format_t mFormat;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index f5f11cc..51e140d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -715,7 +715,8 @@
size_t frameCountToBeReady,
float speed,
bool isSpatialized,
- bool isBitPerfect) {
+ bool isBitPerfect,
+ float volume) {
return sp<Track>::make(thread,
client,
streamType,
@@ -736,7 +737,8 @@
frameCountToBeReady,
speed,
isSpatialized,
- isBitPerfect);
+ isBitPerfect,
+ volume);
}
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
@@ -761,7 +763,8 @@
size_t frameCountToBeReady,
float speed,
bool isSpatialized,
- bool isBitPerfect)
+ bool isBitPerfect,
+ float volume)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
@@ -797,7 +800,8 @@
mFlags(flags),
mSpeed(speed),
mIsSpatialized(isSpatialized),
- mIsBitPerfect(isBitPerfect)
+ mIsBitPerfect(isBitPerfect),
+ mVolume(volume)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -843,6 +847,14 @@
thread->fastTrackAvailMask_l() &= ~(1 << i);
}
+ populateUsageAndContentTypeFromStreamType();
+
+ // Audio patch and call assistant volume are always max
+ if (mAttr.usage == AUDIO_USAGE_CALL_ASSISTANT
+ || mAttr.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ mVolume = 1.0f;
+ }
+
mServerLatencySupported = checkServerLatencySupported(format, flags);
#ifdef TEE_SINK
mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
@@ -865,6 +877,62 @@
mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
}
+// When attributes are undefined, derive default values from stream type.
+// See AudioAttributes.java, usageForStreamType() and Builder.setInternalLegacyStreamType()
+void Track::populateUsageAndContentTypeFromStreamType() {
+ if (mAttr.usage == AUDIO_USAGE_UNKNOWN) {
+ switch (mStreamType) {
+ case AUDIO_STREAM_VOICE_CALL:
+ mAttr.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ break;
+ case AUDIO_STREAM_SYSTEM:
+ mAttr.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_RING:
+ mAttr.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_MUSIC:
+ mAttr.usage = AUDIO_USAGE_MEDIA;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ break;
+ case AUDIO_STREAM_ALARM:
+ mAttr.usage = AUDIO_USAGE_ALARM;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_NOTIFICATION:
+ mAttr.usage = AUDIO_USAGE_NOTIFICATION;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_DTMF:
+ mAttr.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_ACCESSIBILITY:
+ mAttr.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ break;
+ case AUDIO_STREAM_ASSISTANT:
+ mAttr.usage = AUDIO_USAGE_ASSISTANT;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ break;
+ case AUDIO_STREAM_REROUTING:
+ case AUDIO_STREAM_PATCH:
+ mAttr.usage = AUDIO_USAGE_VIRTUAL_SOURCE;
+ // unknown content type
+ break;
+ case AUDIO_STREAM_CALL_ASSISTANT:
+ mAttr.usage = AUDIO_USAGE_CALL_ASSISTANT;
+ mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ break;
+ default:
+ break;
+ }
+ }
+}
+
Track::~Track()
{
ALOGV("%s(%d)", __func__, mId);
@@ -923,7 +991,7 @@
result.appendFormat("Type Id Active Client Session Port Id S Flags "
" Format Chn mask SRate "
"ST Usg CT "
- " G db L dB R dB VS dB "
+ " G db L dB R dB VS dB PortVol dB "
" Server FrmCnt FrmRdy F Underruns Flushed BitPerfect InternalMute"
"%s\n",
isServerLatencySupported() ? " Latency" : "");
@@ -1009,7 +1077,7 @@
result.appendFormat("%7s %6u %7u %7u %2s 0x%03X "
"%08X %08X %6u "
"%2u %3x %2x "
- "%5.2g %5.2g %5.2g %5.2g%c "
+ "%5.2g %5.2g %5.2g %5.2g%c %11.2g "
"%08X %6zu%c %6zu %c %9u%c %7u %10s %12s",
active ? "yes" : "no",
(mClient == 0) ? getpid() : mClient->pid(),
@@ -1031,6 +1099,7 @@
20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
20.0 * log10(vsVolume.first), // VolumeShaper(s) total volume
vsVolume.second ? 'A' : ' ', // if any VolumeShapers active
+ 20.0 * log10(mVolume),
mCblk->mServer,
bufferSizeInFrames,
@@ -1587,59 +1656,6 @@
.gain = mFinalVolume,
};
- // When attributes are undefined, derive default values from stream type.
- // See AudioAttributes.java, usageForStreamType() and Builder.setInternalLegacyStreamType()
- if (mAttr.usage == AUDIO_USAGE_UNKNOWN) {
- switch (mStreamType) {
- case AUDIO_STREAM_VOICE_CALL:
- metadata.base.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- break;
- case AUDIO_STREAM_SYSTEM:
- metadata.base.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- break;
- case AUDIO_STREAM_RING:
- metadata.base.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- break;
- case AUDIO_STREAM_MUSIC:
- metadata.base.usage = AUDIO_USAGE_MEDIA;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_MUSIC;
- break;
- case AUDIO_STREAM_ALARM:
- metadata.base.usage = AUDIO_USAGE_ALARM;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- break;
- case AUDIO_STREAM_NOTIFICATION:
- metadata.base.usage = AUDIO_USAGE_NOTIFICATION;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- break;
- case AUDIO_STREAM_DTMF:
- metadata.base.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- break;
- case AUDIO_STREAM_ACCESSIBILITY:
- metadata.base.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- break;
- case AUDIO_STREAM_ASSISTANT:
- metadata.base.usage = AUDIO_USAGE_ASSISTANT;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- break;
- case AUDIO_STREAM_REROUTING:
- metadata.base.usage = AUDIO_USAGE_VIRTUAL_SOURCE;
- // unknown content type
- break;
- case AUDIO_STREAM_CALL_ASSISTANT:
- metadata.base.usage = AUDIO_USAGE_CALL_ASSISTANT;
- metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- break;
- default:
- break;
- }
- }
-
metadata.channel_mask = mChannelMask;
strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
*backInserter++ = metadata;
@@ -2191,14 +2207,13 @@
size_t frameCount,
const AttributionSourceState& attributionSource)
: Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
- audio_attributes_t{} /* currently unused for output track */,
+ AUDIO_ATTRIBUTES_INITIALIZER ,
sampleRate, format, channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, nullptr /* sharedBuffer */,
AUDIO_SESSION_NONE, getpid(), attributionSource, AUDIO_OUTPUT_FLAG_NONE,
TYPE_OUTPUT),
mActive(false), mSourceThread(sourceThread)
{
-
if (mCblk != NULL) {
mOutBuffer.frameCount = 0;
playbackThread->addOutputTrack_l(this);
@@ -2464,7 +2479,8 @@
* as soon as possible to have
* the lowest possible latency
* even if it might glitch. */
- float speed)
+ float speed,
+ float volume)
{
return sp<PatchTrack>::make(
playbackThread,
@@ -2478,7 +2494,8 @@
flags,
timeout,
frameCountToBeReady,
- speed);
+ speed,
+ volume);
}
PatchTrack::PatchTrack(IAfPlaybackThread* playbackThread,
@@ -2492,13 +2509,15 @@
audio_output_flags_t flags,
const Timeout& timeout,
size_t frameCountToBeReady,
- float speed)
+ float speed,
+ float volume)
: Track(playbackThread, NULL, streamType,
- audio_attributes_t{} /* currently unused for patch track */,
+ AUDIO_ATTRIBUTES_INITIALIZER,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, nullptr /* sharedBuffer */,
AUDIO_SESSION_NONE, getpid(), audioServerAttributionSource(getpid()), flags,
- TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady, speed),
+ TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady, speed,
+ false /*isSpatialized*/, false /*isBitPerfect*/, volume),
PatchTrackBase(mCblk ? new AudioTrackClientProxy(mCblk, mBuffer, frameCount, mFrameSize,
true /*clientInServer*/) : nullptr,
playbackThread, timeout)
@@ -3482,7 +3501,8 @@
bool isOut,
const android::content::AttributionSourceState& attributionSource,
pid_t creatorPid,
- audio_port_handle_t portId)
+ audio_port_handle_t portId,
+ float volume)
{
return sp<MmapTrack>::make(
thread,
@@ -3494,7 +3514,8 @@
isOut,
attributionSource,
creatorPid,
- portId);
+ portId,
+ volume);
}
MmapTrack::MmapTrack(IAfThreadBase* thread,
@@ -3506,7 +3527,8 @@
bool isOut,
const AttributionSourceState& attributionSource,
pid_t creatorPid,
- audio_port_handle_t portId)
+ audio_port_handle_t portId,
+ float volume)
: TrackBase(thread, NULL, attr, sampleRate, format,
channelMask, (size_t)0 /* frameCount */,
nullptr /* buffer */, (size_t)0 /* bufferSize */,
@@ -3517,10 +3539,15 @@
TYPE_DEFAULT, portId,
std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_MMAP) + std::to_string(portId)),
mPid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.pid))),
- mSilenced(false), mSilencedNotified(false)
+ mSilenced(false), mSilencedNotified(false), mVolume(volume)
{
// Once this item is logged by the server, the client can add properties.
mTrackMetrics.logConstructor(creatorPid, uid(), id());
+ if (isOut && (attr.usage == AUDIO_USAGE_CALL_ASSISTANT
+ || attr.usage == AUDIO_USAGE_VIRTUAL_SOURCE)) {
+ // Audio patch and call assistant volume are always max
+ mVolume = 1.0f;
+ }
}
MmapTrack::~MmapTrack()
@@ -3599,8 +3626,8 @@
void MmapTrack::appendDumpHeader(String8& result) const
{
- result.appendFormat("Client Session Port Id Format Chn mask SRate Flags %s\n",
- isOut() ? "Usg CT": "Source");
+ result.appendFormat("Client Session Port Id Format Chn mask SRate Flags %s %s\n",
+ isOut() ? "Usg CT": "Source", isOut() ? "PortVol dB" : "");
}
void MmapTrack::appendDump(String8& result, bool active __unused) const
@@ -3615,6 +3642,7 @@
mAttr.flags);
if (isOut()) {
result.appendFormat("%3x %2x", mAttr.usage, mAttr.content_type);
+ result.appendFormat("%11.2g", 20.0 * log10(mVolume));
} else {
result.appendFormat("%6x", mAttr.source);
}
diff --git a/services/audioflinger/datapath/VolumePortInterface.h b/services/audioflinger/datapath/VolumePortInterface.h
new file mode 100644
index 0000000..fb1c463
--- /dev/null
+++ b/services/audioflinger/datapath/VolumePortInterface.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (C) 2024 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+
+namespace android {
+
+class VolumePortInterface : public virtual RefBase {
+public:
+ virtual void setPortVolume(float volume) = 0;
+ virtual float getPortVolume() const = 0;
+};
+
+} // namespace android
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index deb7345..8e8fac8 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -147,7 +147,8 @@
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
bool *isSpatialized,
- bool *isBitPerfect) = 0;
+ bool *isBitPerfect,
+ float *volume) = 0;
// indicates to the audio policy manager that the output starts being used by corresponding
// stream.
virtual status_t startOutput(audio_port_handle_t portId) = 0;
@@ -514,6 +515,18 @@
// for each output (destination device) it is attached to.
virtual status_t setStreamVolume(audio_stream_type_t stream, float volume,
audio_io_handle_t output, int delayMs = 0) = 0;
+ /**
+ * Set volume for given AudioTrack port ids for a particular output.
+ * For the same user setting, a given volume group and associated output port id
+ * can have different volumes for each output (destination device) it is attached to.
+ * @param ports to consider
+ * @param volume to apply
+ * @param output to consider
+ * @param delayMs to use
+ * @return NO_ERROR if successful
+ */
+ virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& ports, float volume,
+ audio_io_handle_t output, int delayMs = 0) = 0;
// function enabling to send proprietary informations directly from audio policy manager to
// audio hardware interface.
diff --git a/services/audiopolicy/common/managerdefinitions/Android.bp b/services/audiopolicy/common/managerdefinitions/Android.bp
index 051e975..4dedcd6 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.bp
+++ b/services/audiopolicy/common/managerdefinitions/Android.bp
@@ -39,6 +39,8 @@
"android.media.audiopolicy-aconfig-cc",
"audioclient-types-aidl-cpp",
"audiopolicy-types-aidl-cpp",
+ "com.android.media.audioserver-aconfig-cc",
+ "libaconfig_storage_read_api_cc",
"libaudioclient_aidl_conversion",
"libaudiofoundation",
"libaudiopolicy",
@@ -51,6 +53,7 @@
"libmedia_helper",
"libutils",
"libxml2",
+ "server_configurable_flags",
],
export_shared_lib_headers: [
"libaudiofoundation",
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 914f3fe..203fa80 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -490,6 +490,13 @@
virtual std::string info() const override;
+ /**
+ * Finds all ports matching the given volume source.
+ * @param vs to be considered
+ * @return vector of ports following the given volume source.
+ */
+ std::vector<audio_port_handle_t> getPortsForVolumeSource(const VolumeSource& vs);
+
const sp<IOProfile> mProfile; // I/O profile this output derives from
audio_io_handle_t mIoHandle; // output handle
uint32_t mLatency; //
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 0131ba0..a0f1006 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -27,6 +27,7 @@
#include "HwModule.h"
#include "TypeConverter.h"
#include "policy.h"
+#include <com_android_media_audioserver.h>
#include <media/AudioGain.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicy.h>
@@ -34,6 +35,8 @@
// A device mask for all audio output devices that are considered "remote" when evaluating
// active output devices in isStreamActiveRemotely()
+namespace audioserver_flags = com::android::media::audioserver;
+
namespace android {
static const DeviceTypeSet& getAllOutRemoteDevices() {
@@ -498,17 +501,33 @@
const DeviceTypeSet& deviceTypes, uint32_t delayMs) {
// volume source active and more than one volume source is active, otherwise, no-op or let
// setVolume controlling SW and/or HW Gains
- if (!streamTypes.empty() && isActive(vs) && (getActiveVolumeSources().size() > 1)) {
- for (const auto& devicePort : devices()) {
- if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
+ if (!audioserver_flags::portid_volume_management()) {
+ if (!streamTypes.empty() && isActive(vs) && (getActiveVolumeSources().size() > 1)) {
+ for (const auto& devicePort : devices()) {
+ if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
devicePort->hasGainController(true /*canUseForVolume*/)) {
- float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
- ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
- mIoHandle, vs, muted, getActiveVolumeSources().size());
- for (const auto &stream : streamTypes) {
- mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
+ ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
+ mIoHandle, vs, muted, getActiveVolumeSources().size());
+ for (const auto &stream : streamTypes) {
+ mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ }
+ return;
}
- return;
+ }
+ }
+ } else {
+ if (isActive(vs) && (getActiveVolumeSources().size() > 1)) {
+ for (const auto &devicePort: devices()) {
+ if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
+ devicePort->hasGainController(true /*canUseForVolume*/)) {
+ float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
+ ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
+ mIoHandle, vs, muted, getActiveVolumeSources().size());
+ mClientInterface->setPortsVolume(
+ getPortsForVolumeSource(vs), volumeAmpl, mIoHandle, delayMs);
+ return;
+ }
}
}
}
@@ -528,8 +547,14 @@
VolumeSource callVolSrc = getVoiceSource();
if (callVolSrc != VOLUME_SOURCE_NONE && volumeDb != getCurVolume(callVolSrc)) {
setCurVolume(callVolSrc, volumeDb, true);
- mClientInterface->setStreamVolume(
- AUDIO_STREAM_VOICE_CALL, Volume::DbToAmpl(volumeDb), mIoHandle, delayMs);
+ float volumeAmpl = Volume::DbToAmpl(volumeDb);
+ if (audioserver_flags::portid_volume_management()) {
+ mClientInterface->setPortsVolume(getPortsForVolumeSource(callVolSrc),
+ volumeAmpl, mIoHandle, delayMs);
+ } else {
+ mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL,
+ volumeAmpl, mIoHandle, delayMs);
+ }
}
}
return false;
@@ -539,25 +564,34 @@
}
for (const auto& devicePort : devices()) {
// APM loops on all group, so filter on active group to set the port gain,
- // let the other groups set the stream volume as per legacy
+ // let the other groups set the sw volume as per legacy
// TODO: Pass in the device address and check against it.
if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
devicePort->hasGainController(true) && isActive(vs)) {
ALOGV("%s: device %s has gain controller", __func__, devicePort->toString().c_str());
// @todo: here we might be in trouble if the SwOutput has several active clients with
// different Volume Source (or if we allow several curves within same volume group)
- //
- // @todo: default stream volume to max (0) when using HW Port gain?
- // Allows to set SW Gain on AudioFlinger if:
- // -volume group has explicit stream(s) associated
- // -volume group with no explicit stream(s) is the only active source on this output
- // Allows to mute SW Gain on AudioFlinger only for volume group with explicit stream(s)
- if (!streamTypes.empty() || (getActiveVolumeSources().size() == 1)) {
- const bool canMute = muted && (volumeDb != 0.0f) && !streamTypes.empty();
- float volumeAmpl = canMute ? 0.0f : Volume::DbToAmpl(0);
- for (const auto &stream : streams) {
- mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ if (!audioserver_flags::portid_volume_management()) {
+ // @todo: default stream volume to max (0) when using HW Port gain?
+ // Allows to set SW Gain on AudioFlinger if:
+ // -volume group has explicit stream(s) associated
+ // -volume group with no explicit stream(s) is the only active source on this
+ // output
+ // Allows to mute SW Gain on AudioFlinger only for volume group with explicit
+ // stream(s)
+ if (!streamTypes.empty() || (getActiveVolumeSources().size() == 1)) {
+ const bool canMute = muted && (volumeDb != 0.0f) && !streamTypes.empty();
+ float volumeAmpl = canMute ? 0.0f : Volume::DbToAmpl(0);
+ for (const auto &stream: streams) {
+ mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ }
}
+ } else {
+ float volumeAmpl = (muted && volumeDb != 0.0f) ? 0.0f : Volume::DbToAmpl(0);
+ ALOGV("%s: output: %d, vs: %d, active vs count: %zu", __func__,
+ mIoHandle, vs, getActiveVolumeSources().size());
+ mClientInterface->setPortsVolume(
+ getPortsForVolumeSource(vs), volumeAmpl, mIoHandle, delayMs);
}
AudioGains gains = devicePort->getGains();
int gainMinValueInMb = gains[0]->getMinValueInMb();
@@ -577,20 +611,47 @@
// Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is enabled
float volumeAmpl = Volume::DbToAmpl(getCurVolume(vs));
if (hasStream(streams, AUDIO_STREAM_BLUETOOTH_SCO)) {
- mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volumeAmpl, mIoHandle, delayMs);
VolumeSource callVolSrc = getVoiceSource();
+ if (audioserver_flags::portid_volume_management()) {
+ if (callVolSrc != VOLUME_SOURCE_NONE) {
+ mClientInterface->setPortsVolume(getPortsForVolumeSource(callVolSrc), volumeAmpl,
+ mIoHandle, delayMs);
+ }
+ } else {
+ mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volumeAmpl, mIoHandle,
+ delayMs);
+ }
if (callVolSrc != VOLUME_SOURCE_NONE) {
setCurVolume(callVolSrc, getCurVolume(vs), true);
}
}
- for (const auto &stream : streams) {
- ALOGV("%s output %d for volumeSource %d, volume %f, delay %d stream=%s", __func__,
- mIoHandle, vs, volumeDb, delayMs, toString(stream).c_str());
- mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ if (audioserver_flags::portid_volume_management()) {
+ ALOGV("%s output %d for volumeSource %d, volume %f, delay %d active=%d", __func__,
+ mIoHandle, vs, volumeDb, delayMs, isActive(vs));
+ mClientInterface->setPortsVolume(getPortsForVolumeSource(vs), volumeAmpl, mIoHandle,
+ delayMs);
+ } else {
+ for (const auto &stream : streams) {
+ ALOGV("%s output %d for volumeSource %d, volume %f, delay %d stream=%s", __func__,
+ mIoHandle, vs, volumeDb, delayMs, toString(stream).c_str());
+ mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+ }
}
return true;
}
+std::vector<audio_port_handle_t> SwAudioOutputDescriptor::getPortsForVolumeSource(
+ const VolumeSource& vs)
+{
+ std::vector<audio_port_handle_t> portsForVolumeSource;
+ for (const auto& client : getClientIterable()) {
+ if (client->volumeSource() == vs) {
+ portsForVolumeSource.push_back(client->portId());
+ }
+ }
+ return portsForVolumeSource;
+}
+
status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
index 3dc2229..c9a77a4 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
@@ -38,6 +38,8 @@
shared_libs: [
"libaudiopolicycomponents",
"libaudiopolicyengineconfigurable",
+ "libbase",
+ "libcutils",
"liblog",
"libmedia_helper",
"libparameter",
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
index 6416a47..fd40c04 100644
--- a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -265,6 +265,7 @@
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized;
bool isBitPerfect;
+ float volume;
// TODO b/182392769: use attribution source util
AttributionSourceState attributionSource;
@@ -272,7 +273,7 @@
attributionSource.token = sp<BBinder>::make();
if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, attributionSource,
&config, &flags, selectedDeviceId, portId, {}, &outputType, &isSpatialized,
- &isBitPerfect) != OK) {
+ &isBitPerfect, &volume) != OK) {
return false;
}
if (*output == AUDIO_IO_HANDLE_NONE || *portId == AUDIO_PORT_HANDLE_NONE) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 739e201..7cc6791 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1488,7 +1488,8 @@
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
bool *isSpatialized,
- bool *isBitPerfect)
+ bool *isBitPerfect,
+ float *volume)
{
// The supplied portId must be AUDIO_PORT_HANDLE_NONE
if (*portId != AUDIO_PORT_HANDLE_NONE) {
@@ -1544,6 +1545,8 @@
outputDesc->mPolicyMix);
outputDesc->addClient(clientDesc);
+ *volume = Volume::DbToAmpl(outputDesc->getCurVolume(toVolumeSource(resultAttr)));
+
ALOGV("%s() returns output %d requestedPortId %d selectedDeviceId %d for port ID %d", __func__,
*output, requestedPortId, *selectedDeviceId, *portId);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 98853ce..a67ba78 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -128,7 +128,8 @@
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType,
bool *isSpatialized,
- bool *isBitPerfect) override;
+ bool *isBitPerfect,
+ float *volume) override;
virtual status_t startOutput(audio_port_handle_t portId);
virtual status_t stopOutput(audio_port_handle_t portId);
virtual bool releaseOutput(audio_port_handle_t portId);
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index f70dc52..5008d68 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -188,6 +188,16 @@
delay_ms);
}
+status_t AudioPolicyService::AudioPolicyClient::setPortsVolume(
+ const std::vector<audio_port_handle_t> &ports, float volume, audio_io_handle_t output,
+ int delayMs)
+{
+ if (ports.empty()) {
+ return NO_ERROR;
+ }
+ return mAudioPolicyService->setPortsVolume(ports, volume, output, delayMs);
+}
+
void AudioPolicyService::AudioPolicyClient::setParameters(audio_io_handle_t io_handle,
const String8& keyValuePairs,
int delay_ms)
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index f414862..6194002 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -423,6 +423,7 @@
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized = false;
bool isBitPerfect = false;
+ float volume;
status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
&stream,
attributionSource,
@@ -431,7 +432,8 @@
&secondaryOutputs,
&outputType,
&isSpatialized,
- &isBitPerfect);
+ &isBitPerfect,
+ &volume);
// FIXME: Introduce a way to check for the the telephony device before opening the output
if (result == NO_ERROR) {
@@ -495,6 +497,7 @@
_aidl_return->isBitPerfect = isBitPerfect;
_aidl_return->attr = VALUE_OR_RETURN_BINDER_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributes(attr));
+ _aidl_return->volume = volume;
} else {
_aidl_return->configBase.format = VALUE_OR_RETURN_BINDER_STATUS(
legacy2aidl_audio_format_t_AudioFormatDescription(config.format));
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index cc67481..a8b7954 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -1815,6 +1815,16 @@
data->mIO);
ul.lock();
}break;
+ case SET_PORTS_VOLUME: {
+ VolumePortsData *data = (VolumePortsData *)command->mParam.get();
+ ALOGV("AudioCommandThread() processing set volume Ports %s volume %f, \
+ output %d", data->dumpPorts().c_str(), data->mVolume, data->mIO);
+ ul.unlock();
+ command->mStatus = AudioSystem::setPortsVolume(data->mPorts,
+ data->mVolume,
+ data->mIO);
+ ul.lock();
+ } break;
case SET_PARAMETERS: {
ParametersData *data = (ParametersData *)command->mParam.get();
ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
@@ -2127,6 +2137,23 @@
return sendCommand(command, delayMs);
}
+status_t AudioPolicyService::AudioCommandThread::volumePortsCommand(
+ const std::vector<audio_port_handle_t> &ports, float volume, audio_io_handle_t output,
+ int delayMs)
+{
+ sp<AudioCommand> command = new AudioCommand();
+ command->mCommand = SET_PORTS_VOLUME;
+ sp<VolumePortsData> data = new VolumePortsData();
+ data->mPorts = ports;
+ data->mVolume = volume;
+ data->mIO = output;
+ command->mParam = data;
+ command->mWaitStatus = true;
+ ALOGV("AudioCommandThread() adding set volume ports %s, volume %f, output %d",
+ data->dumpPorts().c_str(), volume, output);
+ return sendCommand(command, delayMs);
+}
+
status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs,
int delayMs)
@@ -2457,6 +2484,31 @@
delayMs = 1;
} break;
+ case SET_PORTS_VOLUME: {
+ VolumePortsData *data = (VolumePortsData *)command->mParam.get();
+ VolumePortsData *data2 = (VolumePortsData *)command2->mParam.get();
+ if (data->mIO != data2->mIO) break;
+ // Can remove command only if port ids list is the same, otherwise, remove from
+ // command 2 all port whose volume will be replaced with command 1 volume.
+ std::vector<audio_port_handle_t> portsOnlyInCommand2{};
+ std::copy_if(data2->mPorts.begin(), data2->mPorts.end(),
+ std::back_inserter(portsOnlyInCommand2), [&](const auto &portId) {
+ return std::find(data->mPorts.begin(), data->mPorts.end(), portId) ==
+ data->mPorts.end();
+ });
+ if (!portsOnlyInCommand2.empty()) {
+ data2->mPorts = portsOnlyInCommand2;
+ break;
+ }
+ ALOGV("Filtering out volume command on output %d for ports %s",
+ data->mIO, data->dumpPorts().c_str());
+ removedCommands.add(command2);
+ command->mTime = command2->mTime;
+ // force delayMs to non 0 so that code below does not request to wait for
+ // command status as the command is now delayed
+ delayMs = 1;
+ } break;
+
case SET_VOICE_VOLUME: {
VoiceVolumeData *data = (VoiceVolumeData *)command->mParam.get();
VoiceVolumeData *data2 = (VoiceVolumeData *)command2->mParam.get();
@@ -2603,6 +2655,12 @@
output, delayMs);
}
+int AudioPolicyService::setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+ audio_io_handle_t output, int delayMs)
+{
+ return (int)mAudioCommandThread->volumePortsCommand(ports, volume, output, delayMs);
+}
+
int AudioPolicyService::setVoiceVolume(float volume, int delayMs)
{
return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 720ba84..0492cd3 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -47,6 +47,7 @@
#include <android/hardware/BnSensorPrivacyListener.h>
#include <android/content/AttributionSourceState.h>
+#include <numeric>
#include <unordered_map>
namespace android {
@@ -354,6 +355,21 @@
float volume,
audio_io_handle_t output,
int delayMs = 0);
+
+ /**
+ * Set a volume on AudioTrack port id(s) for a particular output.
+ * For the same user setting, a volume group (and associated given port of the
+ * client's track) can have different volumes for each output destination device
+ * it is attached to.
+ *
+ * @param ports to consider
+ * @param volume to set
+ * @param output to consider
+ * @param delayMs to use
+ * @return NO_ERROR if successful
+ */
+ virtual status_t setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+ audio_io_handle_t output, int delayMs = 0);
virtual status_t setVoiceVolume(float volume, int delayMs = 0);
void doOnNewAudioModulesAvailable();
@@ -577,6 +593,7 @@
// commands for tone AudioCommand
enum {
SET_VOLUME,
+ SET_PORTS_VOLUME,
SET_PARAMETERS,
SET_VOICE_VOLUME,
STOP_OUTPUT,
@@ -610,6 +627,8 @@
void exit();
status_t volumeCommand(audio_stream_type_t stream, float volume,
audio_io_handle_t output, int delayMs = 0);
+ status_t volumePortsCommand(const std::vector<audio_port_handle_t> &ports,
+ float volume, audio_io_handle_t output, int delayMs = 0);
status_t parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs, int delayMs = 0);
status_t voiceVolumeCommand(float volume, int delayMs = 0);
@@ -684,6 +703,20 @@
audio_io_handle_t mIO;
};
+ class VolumePortsData : public AudioCommandData {
+ public:
+ std::vector<audio_port_handle_t> mPorts;
+ float mVolume;
+ audio_io_handle_t mIO;
+ std::string dumpPorts() {
+ return std::string("volume ") + std::to_string(mVolume) + " on IO " +
+ std::to_string(mIO) + " and ports " +
+ std::accumulate(std::begin(mPorts), std::end(mPorts), std::string{},
+ [] (const std::string& ls, int rs) {
+ return ls + std::to_string(rs) + " "; });
+ }
+ };
+
class ParametersData : public AudioCommandData {
public:
audio_io_handle_t mIO;
@@ -823,6 +856,19 @@
// set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
// for each output (destination device) it is attached to.
virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0);
+ /**
+ * Set a volume on port(s) for a particular output. For the same user setting, a volume
+ * group (and associated given port of the client's track) can have different volumes for
+ * each output (destination device) it is attached to.
+ *
+ * @param ports to consider
+ * @param volume to set
+ * @param output to consider
+ * @param delayMs to use
+ * @return NO_ERROR if successful
+ */
+ status_t setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+ audio_io_handle_t output, int delayMs = 0) override;
// function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0);
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index c15adcb..ea76685 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -57,6 +57,10 @@
float /*volume*/,
audio_io_handle_t /*output*/,
int /*delayMs*/) override { return NO_INIT; }
+
+ status_t setPortsVolume(const std::vector<audio_port_handle_t>& /*ports*/, float /*volume*/,
+ audio_io_handle_t /*output*/, int /*delayMs*/) override { return NO_INIT; }
+
void setParameters(audio_io_handle_t /*ioHandle*/,
const String8& /*keyValuePairs*/,
int /*delayMs*/) override { }
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 07aad0c..eb4240a 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -299,11 +299,12 @@
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized;
bool isBitPerfectInternal;
+ float volume;
AttributionSourceState attributionSource = createAttributionSourceState(uid);
ASSERT_EQ(OK, mManager->getOutputForAttr(
&attr, output, session, &stream, attributionSource, &config, &flags,
selectedDeviceId, portId, {}, &outputType, &isSpatialized,
- isBitPerfect == nullptr ? &isBitPerfectInternal : isBitPerfect));
+ isBitPerfect == nullptr ? &isBitPerfectInternal : isBitPerfect, &volume));
ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
ASSERT_NE(AUDIO_IO_HANDLE_NONE, *output);
}
@@ -2065,6 +2066,7 @@
audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER;
bool mIsSpatialized;
bool mIsBitPerfect;
+ float mVolume;
};
TEST_P(AudioPolicyManagerTestMMapPlaybackRerouting, MmapPlaybackStreamMatchingLoopbackDapMixFails) {
@@ -2083,7 +2085,7 @@
mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
createAttributionSourceState(testUid), &audioConfig,
&outputFlags, &mSelectedDeviceId, &mPortId, {},
- &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+ &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
}
TEST_P(AudioPolicyManagerTestMMapPlaybackRerouting,
@@ -2102,7 +2104,7 @@
mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
createAttributionSourceState(testUid), &audioConfig,
&outputFlags, &mSelectedDeviceId, &mPortId, {},
- &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+ &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
}
TEST_F(AudioPolicyManagerTestMMapPlaybackRerouting,
@@ -2133,7 +2135,7 @@
mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
createAttributionSourceState(testUid), &audioConfig,
&outputFlags, &mSelectedDeviceId, &mPortId, {},
- &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+ &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
ASSERT_EQ(usbDevicePort.id, mSelectedDeviceId);
auto outputDesc = mManager->getOutputs().valueFor(mOutput);
ASSERT_NE(nullptr, outputDesc);
@@ -2149,7 +2151,7 @@
mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
createAttributionSourceState(testUid), &audioConfig,
&outputFlags, &mSelectedDeviceId, &mPortId, {},
- &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+ &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
ASSERT_EQ(usbDevicePort.id, mSelectedDeviceId);
outputDesc = mManager->getOutputs().valueFor(mOutput);
ASSERT_NE(nullptr, outputDesc);
@@ -2178,7 +2180,7 @@
mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
createAttributionSourceState(testUid), &audioConfig,
&outputFlags, &mSelectedDeviceId, &mPortId, {},
- &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+ &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
}
INSTANTIATE_TEST_SUITE_P(
@@ -3632,11 +3634,12 @@
AudioPolicyInterface::output_type_t outputType;
bool isSpatialized;
bool isBitPerfect;
+ float volume;
EXPECT_EQ(expected,
mManager->getOutputForAttr(&sMediaAttr, &mBitPerfectOutput, AUDIO_SESSION_NONE,
&stream, attributionSource, &config, &flags,
&mSelectedDeviceId, &mBitPerfectPortId, {}, &outputType,
- &isSpatialized, &isBitPerfect));
+ &isSpatialized, &isBitPerfect, &volume));
}
class AudioPolicyManagerTestBitPerfect : public AudioPolicyManagerTestBitPerfectBase {