audio flinger: pass mixer configuration when opening an audio output
Add the possibility for audio policy manager to indicate a mixer
audio configuration (channel mask, format, sampling rate)
different from the audio HAL stream configuration when opening an
output stream with IAudioflinger::openOutput().
Bug: 188502620
Test: atest audiopolicy_tests
Change-Id: I0b4eaa9cd3e5574b75e3b914c854f97c104ca07e
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 06b12e9..1541948 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -17,6 +17,7 @@
package android.media;
import android.media.AudioConfig;
+import android.media.AudioConfigBase;
import android.media.AudioPort;
/**
@@ -25,7 +26,8 @@
parcelable OpenOutputRequest {
/** Interpreted as audio_module_handle_t. */
int module;
- AudioConfig config;
+ AudioConfig halConfig;
+ AudioConfigBase mixerConfig;
/** Type must be DEVICE. */
AudioPort device;
/** Bitmask, indexed by AudioOutputFlag. */
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index d03c6fa..12473fc 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -648,11 +648,15 @@
sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
+ audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+
media::OpenOutputRequest request{};
media::OpenOutputResponse response{};
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
- request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.mixerConfig =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(mixerConfig));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 76a0ba7..2b688a1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2505,7 +2505,8 @@
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig __unused,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
@@ -2533,16 +2534,16 @@
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
- //config->format = AUDIO_FORMAT_PCM_FLOAT;
- //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
- //config->format = AUDIO_FORMAT_PCM_32_BIT;
- //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
- // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
+ //halConfig->format = AUDIO_FORMAT_PCM_FLOAT;
+ //halConfig->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ //halConfig->format = AUDIO_FORMAT_PCM_32_BIT;
+ //halConfig->format = AUDIO_FORMAT_PCM_8_24_BIT;
+ // ALOGV("openOutput_l() upgrading format to %#08x", halConfig->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
- //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
- //config->channel_mask = audio_channel_mask_from_representation_and_bits(
+ //halConfig->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
+ //halConfig->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
@@ -2553,7 +2554,7 @@
*output,
deviceType,
flags,
- config,
+ halConfig,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
@@ -2573,8 +2574,8 @@
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
- || !isValidPcmSinkFormat(config->format)
- || !isValidPcmSinkChannelMask(config->channel_mask)) {
+ || !isValidPcmSinkFormat(halConfig->format)
+ || !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
@@ -2601,8 +2602,10 @@
{
audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_module_handle_t(request.module));
- audio_config_t config = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioConfig_audio_config_t(request.config));
+ audio_config_t halConfig = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfig_audio_config_t(request.halConfig));
+ audio_config_base_t mixerConfig = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfigBase_audio_config_base_t(request.mixerConfig));
sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
@@ -2615,9 +2618,9 @@
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
- config.sample_rate,
- config.format,
- config.channel_mask,
+ halConfig.sample_rate,
+ halConfig.format,
+ halConfig.channel_mask,
flags);
audio_devices_t deviceType = device->type();
@@ -2629,7 +2632,8 @@
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
+ sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+ &mixerConfig, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -2654,7 +2658,8 @@
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ response->config =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(halConfig));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
response->flags = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index e3402f9..8fcd6e4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -738,7 +738,8 @@
const String8& outputDeviceAddress);
sp<ThreadBase> openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags);
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index b260700..c222de8 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -163,8 +163,13 @@
bool isOffloadOrMmap() const override { return false; }
uint32_t sampleRate() const override { return 0; }
- audio_channel_mask_t channelMask() const override { return AUDIO_CHANNEL_NONE; }
- uint32_t channelCount() const override { return 0; }
+ audio_channel_mask_t inChannelMask(int id __unused) const override {
+ return AUDIO_CHANNEL_NONE;
+ }
+ uint32_t inChannelCount(int id __unused) const override { return 0; }
+ audio_channel_mask_t outChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+ uint32_t outChannelCount() const override { return 0; }
+
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index c12f03b..bd661f9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -881,9 +881,9 @@
// similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
// in which case input channel masks should be used here.
callback = getCallback();
- channelMask = callback->channelMask();
+ channelMask = callback->inChannelMask(mId);
mConfig.inputCfg.channels = channelMask;
- mConfig.outputCfg.channels = channelMask;
+ mConfig.outputCfg.channels = callback->outChannelMask();
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
if (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_MONO) {
@@ -2133,8 +2133,8 @@
if (mInBuffer == NULL) {
return;
}
- const size_t frameSize =
- audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * mEffectCallback->channelCount();
+ const size_t frameSize = audio_bytes_per_sample(EFFECT_BUFFER_FORMAT)
+ * mEffectCallback->inChannelCount(mEffects[0]->id());
memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
mInBuffer->commit();
@@ -2244,6 +2244,9 @@
numSamples * sizeof(int32_t), &halBuffer);
#endif
if (result != OK) return result;
+
+ effect->configure();
+
effect->setInBuffer(halBuffer);
// auxiliary effects output samples to chain input buffer for further processing
// by insert effects
@@ -2311,6 +2314,10 @@
}
}
+ mEffects.insertAt(effect, idx_insert);
+
+ effect->configure();
+
// always read samples from chain input buffer
effect->setInBuffer(mInBuffer);
@@ -2318,14 +2325,13 @@
// output buffer, otherwise to chain input buffer
if (idx_insert == size) {
if (idx_insert != 0) {
- mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
mEffects[idx_insert-1]->configure();
+ mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
}
effect->setOutBuffer(mOutBuffer);
} else {
effect->setOutBuffer(mInBuffer);
}
- mEffects.insertAt(effect, idx_insert);
ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
idx_insert);
@@ -2358,14 +2364,21 @@
if (type != EFFECT_FLAG_TYPE_AUXILIARY) {
if (i == size - 1 && i != 0) {
- mEffects[i - 1]->setOutBuffer(mOutBuffer);
mEffects[i - 1]->configure();
+ mEffects[i - 1]->setOutBuffer(mOutBuffer);
}
}
mEffects.removeAt(i);
+
+ // make sure the input buffer configuration for the new first effect in the chain
+ // is updated if needed (can switch from HAL channel mask to mixer channel mask)
+ if (i == 0 && size > 1) {
+ mEffects[0]->configure();
+ mEffects[0]->setInBuffer(mInBuffer);
+ }
+
ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
this, i);
-
break;
}
}
@@ -2940,7 +2953,43 @@
return t->sampleRate();
}
-audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::inChannelMask(int id) const {
+ sp<ThreadBase> t = thread().promote();
+ if (t == nullptr) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ sp<EffectChain> c = chain().promote();
+ if (c == nullptr) {
+ return AUDIO_CHANNEL_NONE;
+ }
+
+ if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+ || c->isFirstEffect(id)) {
+ return t->mixerChannelMask();
+ } else {
+ return t->channelMask();
+ }
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::inChannelCount(int id) const {
+ sp<ThreadBase> t = thread().promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ sp<EffectChain> c = chain().promote();
+ if (c == nullptr) {
+ return 0;
+ }
+
+ if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+ || c->isFirstEffect(id)) {
+ return audio_channel_count_from_out_mask(t->mixerChannelMask());
+ } else {
+ return t->channelCount();
+ }
+}
+
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::outChannelMask() const {
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
@@ -2948,7 +2997,7 @@
return t->channelMask();
}
-uint32_t AudioFlinger::EffectChain::EffectCallback::channelCount() const {
+uint32_t AudioFlinger::EffectChain::EffectCallback::outChannelCount() const {
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
@@ -3372,7 +3421,8 @@
return proxy->sampleRate();
}
-audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelMask(
+ int id __unused) const {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return AUDIO_CHANNEL_OUT_STEREO;
@@ -3380,7 +3430,23 @@
return proxy->channelMask();
}
-uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelCount() const {
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelCount(int id __unused) const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return 2;
+ }
+ return proxy->channelCount();
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelMask() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return AUDIO_CHANNEL_OUT_STEREO;
+ }
+ return proxy->channelMask();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelCount() const {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return 2;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index b6f9758..1d0d00d 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -34,8 +34,10 @@
virtual bool isOffloadOrDirect() const = 0;
virtual bool isOffloadOrMmap() const = 0;
virtual uint32_t sampleRate() const = 0;
- virtual audio_channel_mask_t channelMask() const = 0;
- virtual uint32_t channelCount() const = 0;
+ virtual audio_channel_mask_t inChannelMask(int id) const = 0;
+ virtual uint32_t inChannelCount(int id) const = 0;
+ virtual audio_channel_mask_t outChannelMask() const = 0;
+ virtual uint32_t outChannelCount() const = 0;
virtual audio_channel_mask_t hapticChannelMask() const = 0;
virtual size_t frameCount() const = 0;
@@ -525,6 +527,8 @@
sp<EffectCallbackInterface> effectCallback() const { return mEffectCallback; }
wp<ThreadBase> thread() const { return mEffectCallback->thread(); }
+ bool isFirstEffect(int id) const { return !mEffects.isEmpty() && id == mEffects[0]->id(); }
+
void dump(int fd, const Vector<String16>& args);
private:
@@ -558,8 +562,10 @@
bool isOffloadOrMmap() const override;
uint32_t sampleRate() const override;
- audio_channel_mask_t channelMask() const override;
- uint32_t channelCount() const override;
+ audio_channel_mask_t inChannelMask(int id) const override;
+ uint32_t inChannelCount(int id) const override;
+ audio_channel_mask_t outChannelMask() const override;
+ uint32_t outChannelCount() const override;
audio_channel_mask_t hapticChannelMask() const override;
size_t frameCount() const override;
uint32_t latency() const override;
@@ -712,8 +718,10 @@
bool isOffloadOrMmap() const override { return false; }
uint32_t sampleRate() const override;
- audio_channel_mask_t channelMask() const override;
- uint32_t channelCount() const override;
+ audio_channel_mask_t inChannelMask(int id) const override;
+ uint32_t inChannelCount(int id) const override;
+ audio_channel_mask_t outChannelMask() const override;
+ uint32_t outChannelCount() const override;
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index a381c7d..93118b8 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -258,6 +258,7 @@
reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
} else {
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
@@ -276,6 +277,7 @@
patch->sinks[0].ext.device.hw_module,
&output,
&config,
+ &mixerConfig,
outputDevice,
outputDeviceAddress,
flags);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 08d2d93..f4aff6d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1904,7 +1904,8 @@
AudioStreamOut* output,
audio_io_handle_t id,
type_t type,
- bool systemReady)
+ bool systemReady,
+ audio_config_base_t *mixerConfig)
: ThreadBase(audioFlinger, id, type, systemReady, true /* isOut */),
mNormalFrameCount(0), mSinkBuffer(NULL),
mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
@@ -1964,6 +1965,10 @@
mOutput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
}
+ if (mixerConfig != nullptr && mixerConfig->channel_mask != AUDIO_CHANNEL_NONE) {
+ mMixerChannelMask = mixerConfig->channel_mask;
+ }
+
readOutputParameters_l();
// TODO: We may also match on address as well as device type for
@@ -2092,6 +2097,8 @@
{
dprintf(fd, " Master volume: %f\n", mMasterVolume);
dprintf(fd, " Master mute: %s\n", mMasterMute ? "on" : "off");
+ dprintf(fd, " Mixer channel Mask: %#x (%s)\n",
+ mMixerChannelMask, channelMaskToString(mMixerChannelMask, true /* output */).c_str());
if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
dprintf(fd, " Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
channelMaskToString(mHapticChannelMask, true /* output */).c_str());
@@ -2838,9 +2845,16 @@
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
mChannelMask);
}
+
+ if (mMixerChannelMask == AUDIO_CHANNEL_NONE) {
+ mMixerChannelMask = mChannelMask;
+ }
+
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mBalance.setChannelMask(mChannelMask);
+ uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mMixerChannelMask);
+
// Get actual HAL format.
status_t result = mOutput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
@@ -2960,7 +2974,7 @@
mMixerBuffer = NULL;
if (mMixerBufferEnabled) {
mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
- mMixerBufferSize = mNormalFrameCount * mChannelCount
+ mMixerBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mMixerBufferFormat);
(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
}
@@ -2968,7 +2982,7 @@
mEffectBuffer = NULL;
if (mEffectBufferEnabled) {
mEffectBufferFormat = EFFECT_BUFFER_FORMAT;
- mEffectBufferSize = mNormalFrameCount * mChannelCount
+ mEffectBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mEffectBufferFormat);
(void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
}
@@ -2977,6 +2991,7 @@
mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
mChannelCount -= mHapticChannelCount;
+ mMixerChannelMask = static_cast<audio_channel_mask_t>(mMixerChannelMask & ~mHapticChannelMask);
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
@@ -3367,7 +3382,8 @@
// Only one effect chain can be present in direct output thread and it uses
// the sink buffer as input
if (mType != DIRECT) {
- size_t numSamples = mNormalFrameCount * (mChannelCount + mHapticChannelCount);
+ size_t numSamples = mNormalFrameCount
+ * (audio_channel_count_from_out_mask(mMixerChannelMask) + mHapticChannelCount);
status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
numSamples * sizeof(effect_buffer_t),
&halInBuffer);
@@ -3776,6 +3792,8 @@
if (mMixerBufferValid) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
+ uint32_t channelCount = mEffectBufferValid ?
+ audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
// mono blend occurs for mixer threads only (not direct or offloaded)
// and is handled here if we're going directly to the sink.
@@ -3793,7 +3811,7 @@
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
- mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+ mNormalFrameCount * (channelCount + mHapticChannelCount));
// If we're going directly to the sink and there are haptic channels,
// we should adjust channels as the sample data is partially interleaved
@@ -4467,8 +4485,8 @@
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, bool systemReady, type_t type)
- : PlaybackThread(audioFlinger, output, id, type, systemReady),
+ audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
+ : PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
// mAudioMixer below
// mFastMixer below
mFastMixerFutex(0),
@@ -5377,7 +5395,7 @@
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK,
- (void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
+ (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
uint32_t reqSampleRate = proxy->getSampleRate();
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index b9dbc7f..6ece59f 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -290,8 +290,11 @@
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
+ virtual audio_channel_mask_t mixerChannelMask() const { return mChannelMask; }
+
audio_format_t format() const { return mHALFormat; }
uint32_t channelCount() const { return mChannelCount; }
+
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
@@ -826,7 +829,8 @@
static const nsecs_t kMaxNextBufferDelayNs = 100000000;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, type_t type, bool systemReady);
+ audio_io_handle_t id, type_t type, bool systemReady,
+ audio_config_base_t *mixerConfig = nullptr);
virtual ~PlaybackThread();
// Thread virtuals
@@ -975,6 +979,10 @@
virtual size_t frameCount() const { return mNormalFrameCount; }
+ audio_channel_mask_t mixerChannelMask() const override {
+ return mMixerChannelMask;
+ }
+
status_t getTimestamp_l(AudioTimestamp& timestamp);
void addPatchTrack(const sp<PatchTrack>& track);
@@ -1103,6 +1111,9 @@
// haptic playback.
audio_channel_mask_t mHapticChannelMask = AUDIO_CHANNEL_NONE;
uint32_t mHapticChannelCount = 0;
+
+ audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
+
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
@@ -1324,7 +1335,8 @@
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
- type_t type = MIXER);
+ type_t type = MIXER,
+ audio_config_base_t *mixerConfig = nullptr);
virtual ~MixerThread();
// Thread virtuals
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 2e49e71..6b11d9a 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -359,7 +359,8 @@
// The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 1f9b535..7c7f02d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -362,7 +362,8 @@
const struct audio_port_config *srcConfig = NULL) const;
virtual void toAudioPort(struct audio_port_v7 *port) const;
- status_t open(const audio_config_t *config,
+ status_t open(const audio_config_t *halConfig,
+ const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
@@ -423,6 +424,7 @@
uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
audio_session_t mDirectClientSession; // session id of the direct output client
bool mPendingReopenToQueryProfiles = false;
+ audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
};
// Audio output driven by an input device directly.
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 6b08f7c..d4a23c7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -491,7 +491,8 @@
return true;
}
-status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
+status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
+ const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
@@ -504,32 +505,42 @@
"with the requested devices, all device types: %s",
__func__, dumpDeviceTypes(devices.types()).c_str());
- audio_config_t lConfig;
- if (config == nullptr) {
- lConfig = AUDIO_CONFIG_INITIALIZER;
- lConfig.sample_rate = mSamplingRate;
- lConfig.channel_mask = mChannelMask;
- lConfig.format = mFormat;
+ audio_config_t lHalConfig;
+ if (halConfig == nullptr) {
+ lHalConfig = AUDIO_CONFIG_INITIALIZER;
+ lHalConfig.sample_rate = mSamplingRate;
+ lHalConfig.channel_mask = mChannelMask;
+ lHalConfig.format = mFormat;
} else {
- lConfig = *config;
+ lHalConfig = *halConfig;
}
// if the selected profile is offloaded and no offload info was specified,
// create a default one
if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
- lConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
+ lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- lConfig.offload_info = AUDIO_INFO_INITIALIZER;
- lConfig.offload_info.sample_rate = lConfig.sample_rate;
- lConfig.offload_info.channel_mask = lConfig.channel_mask;
- lConfig.offload_info.format = lConfig.format;
- lConfig.offload_info.stream_type = stream;
- lConfig.offload_info.duration_us = -1;
- lConfig.offload_info.has_video = true; // conservative
- lConfig.offload_info.is_streaming = true; // likely
- lConfig.offload_info.encapsulation_mode = lConfig.offload_info.encapsulation_mode;
- lConfig.offload_info.content_id = lConfig.offload_info.content_id;
- lConfig.offload_info.sync_id = lConfig.offload_info.sync_id;
+ lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
+ lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
+ lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
+ lHalConfig.offload_info.format = lHalConfig.format;
+ lHalConfig.offload_info.stream_type = stream;
+ lHalConfig.offload_info.duration_us = -1;
+ lHalConfig.offload_info.has_video = true; // conservative
+ lHalConfig.offload_info.is_streaming = true; // likely
+ lHalConfig.offload_info.encapsulation_mode = lHalConfig.offload_info.encapsulation_mode;
+ lHalConfig.offload_info.content_id = lHalConfig.offload_info.content_id;
+ lHalConfig.offload_info.sync_id = lHalConfig.offload_info.sync_id;
+ }
+
+ audio_config_base_t lMixerConfig;
+ if (mixerConfig == nullptr) {
+ lMixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+ lMixerConfig.sample_rate = lHalConfig.sample_rate;
+ lMixerConfig.channel_mask = lHalConfig.channel_mask;
+ lMixerConfig.format = lHalConfig.format;
+ } else {
+ lMixerConfig = *mixerConfig;
}
mFlags = (audio_output_flags_t)(mFlags | flags);
@@ -539,7 +550,8 @@
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
- &lConfig,
+ &lHalConfig,
+ &lMixerConfig,
device,
&mLatency,
mFlags);
@@ -550,9 +562,10 @@
"selected device %s for opening",
__FUNCTION__, *output, devices.toString().c_str(),
device->toString().c_str());
- mSamplingRate = lConfig.sample_rate;
- mChannelMask = lConfig.channel_mask;
- mFormat = lConfig.format;
+ mSamplingRate = lHalConfig.sample_rate;
+ mChannelMask = lHalConfig.channel_mask;
+ mFormat = lHalConfig.format;
+ mMixerChannelMask = lMixerConfig.channel_mask;
mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *output;
mProfile->curOpenCount++;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 83a4a37..d6dd762 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1265,7 +1265,8 @@
// all MSD patches to prioritize this request over any active output on MSD.
releaseMsdOutputPatches(devices);
- status_t status = outputDesc->open(config, devices, stream, flags, output);
+ status_t status =
+ outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
@@ -4990,7 +4991,8 @@
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+ status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
+ DeviceVector(supportedDevice),
AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
@@ -6996,7 +6998,7 @@
}
sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = desc->open(nullptr, devices,
+ status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
@@ -7026,7 +7028,7 @@
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
- status = desc->open(&config, devices,
+ status = desc->open(&config, nullptr /* mixerConfig */, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index cd53073..79252d4 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -40,7 +40,8 @@
status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
@@ -55,14 +56,17 @@
media::OpenOutputResponse response;
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
- request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+ request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*halConfig));
+ request.mixerConfig =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
status_t status = af->openOutput(request, &response);
if (status == OK) {
*output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
- *config = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
+ *halConfig =
+ VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
*latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(response.latencyMs));
}
return status;
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 3b77ed8..c0fbde5 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -659,7 +659,8 @@
// The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags);
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index f7b0565..84b40d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -37,7 +37,8 @@
status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t * /*config*/,
+ audio_config_t * /*halConfig*/,
+ audio_config_base_t * /*mixerConfig*/,
const sp<DeviceDescriptorBase>& /*device*/,
uint32_t * /*latencyMs*/,
audio_output_flags_t /*flags*/) override {
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 1384864..4e0735b 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -30,7 +30,8 @@
}
status_t openOutput(audio_module_handle_t /*module*/,
audio_io_handle_t* /*output*/,
- audio_config_t* /*config*/,
+ audio_config_t* /*halConfig*/,
+ audio_config_base_t* /*mixerConfig*/,
const sp<DeviceDescriptorBase>& /*device*/,
uint32_t* /*latencyMs*/,
audio_output_flags_t /*flags*/) override { return NO_INIT; }