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-rw-r--r--services/audioflinger/AudioFlinger.cpp24
-rw-r--r--services/audioflinger/FastMixer.cpp2
-rw-r--r--services/audioflinger/Threads.cpp16
-rw-r--r--services/audioflinger/Tracks.cpp2
-rw-r--r--services/audiopolicy/AudioPolicyManager.cpp53
-rw-r--r--services/audiopolicy/AudioPolicyService.cpp36
-rw-r--r--services/camera/libcameraservice/api1/CameraClient.cpp15
-rw-r--r--services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp17
8 files changed, 135 insertions, 30 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1f77b2f..1843722 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1941,9 +1941,8 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m
TEE_SINK_NEW, // copy input using a new pipe
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
- NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
- audio_channel_count_from_in_mask(
- inStream->common.get_channels(&inStream->common)));
+ NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
+ audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (!Format_isValid(format)) {
@@ -2700,24 +2699,26 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
// if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
if (teeFd >= 0) {
+ // FIXME use libsndfile
char wavHeader[44];
memcpy(wavHeader,
"RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
sizeof(wavHeader));
NBAIO_Format format = teeSource->format();
unsigned channelCount = Format_channelCount(format);
- ALOG_ASSERT(channelCount <= FCC_2);
uint32_t sampleRate = Format_sampleRate(format);
+ size_t frameSize = Format_frameSize(format);
wavHeader[22] = channelCount; // number of channels
wavHeader[24] = sampleRate; // sample rate
wavHeader[25] = sampleRate >> 8;
- wavHeader[32] = channelCount * 2; // block alignment
+ wavHeader[32] = frameSize; // block alignment
+ wavHeader[33] = frameSize >> 8;
write(teeFd, wavHeader, sizeof(wavHeader));
size_t total = 0;
bool firstRead = true;
+#define TEE_SINK_READ 1024 // frames per I/O operation
+ void *buffer = malloc(TEE_SINK_READ * frameSize);
for (;;) {
-#define TEE_SINK_READ 1024
- short buffer[TEE_SINK_READ * FCC_2];
size_t count = TEE_SINK_READ;
ssize_t actual = teeSource->read(buffer, count,
AudioBufferProvider::kInvalidPTS);
@@ -2730,14 +2731,17 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
break;
}
ALOG_ASSERT(actual <= (ssize_t)count);
- write(teeFd, buffer, actual * channelCount * sizeof(short));
+ write(teeFd, buffer, actual * frameSize);
total += actual;
}
+ free(buffer);
lseek(teeFd, (off_t) 4, SEEK_SET);
- uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
+ uint32_t temp = 44 + total * frameSize - 8;
+ // FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
lseek(teeFd, (off_t) 40, SEEK_SET);
- temp = total * channelCount * sizeof(short);
+ temp = total * frameSize;
+ // FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
close(teeFd);
if (fd >= 0) {
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 9e15293..2678cbf 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -420,7 +420,7 @@ void FastMixer::onWork()
// if non-NULL, then duplicate write() to this non-blocking sink
NBAIO_Sink* teeSink;
if ((teeSink = current->mTeeSink) != NULL) {
- (void) teeSink->write(mMixerBuffer, frameCount);
+ (void) teeSink->write(buffer, frameCount);
}
// FIXME write() is non-blocking and lock-free for a properly implemented NBAIO sink,
// but this code should be modified to handle both non-blocking and blocking sinks
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f721d5c..97b1753 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -662,12 +662,14 @@ void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
binder,
getWakeLockTag(),
String16("media"),
- uid);
+ uid,
+ true /* FIXME force oneway contrary to .aidl */);
} else {
status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
- String16("media"));
+ String16("media"),
+ true /* FIXME force oneway contrary to .aidl */);
}
if (status == NO_ERROR) {
mWakeLockToken = binder;
@@ -687,7 +689,8 @@ void AudioFlinger::ThreadBase::releaseWakeLock_l()
if (mWakeLockToken != 0) {
ALOGV("releaseWakeLock_l() %s", mName);
if (mPowerManager != 0) {
- mPowerManager->releaseWakeLock(mWakeLockToken, 0);
+ mPowerManager->releaseWakeLock(mWakeLockToken, 0,
+ true /* FIXME force oneway contrary to .aidl */);
}
mWakeLockToken.clear();
}
@@ -723,7 +726,8 @@ void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uid
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
status_t status;
- status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array());
+ status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
+ true /* FIXME force oneway contrary to .aidl */);
ALOGV("acquireWakeLock_l() %s status %d", mName, status);
}
}
@@ -3662,6 +3666,10 @@ track_is_ready: ;
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
+ if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
+ mEffectBufferValid = true;
+ }
+
// sink or mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to the sink or mix buffer
// and track effects will accumulate into it
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index c5ab832..6cbab04 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -182,7 +182,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
- NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
+ NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
if (Format_isValid(pipeFormat)) {
Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
size_t numCounterOffers = 0;
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index a805923..6446a6e 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -4634,8 +4634,15 @@ audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
// - one A2DP device + another device: happens with duplicated output. In this case
// retain the device on the A2DP output as the other must not correspond to an active
// selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
if (device & AUDIO_DEVICE_OUT_SPEAKER) {
device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+ device = AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+ device = AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+ device = AUDIO_DEVICE_OUT_SPDIF;
} else {
device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
}
@@ -6032,14 +6039,26 @@ uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
return 0;
}
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t samplingRate = UINT_MAX;
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ }
+
uint32_t samplingRate = 0;
uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
// For mixed output and inputs, use max mixer sampling rates. Do not
// limit sampling rate otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)))) {
+ if (mType != AUDIO_PORT_TYPE_MIX) {
maxRate = UINT_MAX;
}
for (size_t i = 0; i < mSamplingRates.size(); i ++) {
@@ -6056,16 +6075,35 @@ audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
return AUDIO_CHANNEL_NONE;
}
-
audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t channelCount = UINT_MAX;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+ }
+
uint32_t channelCount = 0;
uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
// For mixed output and inputs, use max mixer channel count. Do not
// limit channel count otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)))) {
+ if (mType != AUDIO_PORT_TYPE_MIX) {
maxCount = UINT_MAX;
}
for (size_t i = 0; i < mChannelMasks.size(); i ++) {
@@ -6077,6 +6115,7 @@ audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
}
if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
}
}
return channelMask;
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
index 7f14960..50bb8c7 100644
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -765,7 +765,16 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(sp<AudioCommand>& c
sp<AudioCommand> command2 = mAudioCommands[i];
// commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands
if (command2->mTime <= command->mTime) break;
- if (command2->mCommand != command->mCommand) continue;
+
+ // create audio patch or release audio patch commands are equivalent
+ // with regard to filtering
+ if ((command->mCommand == CREATE_AUDIO_PATCH) ||
+ (command->mCommand == RELEASE_AUDIO_PATCH)) {
+ if ((command2->mCommand != CREATE_AUDIO_PATCH) &&
+ (command2->mCommand != RELEASE_AUDIO_PATCH)) {
+ continue;
+ }
+ } else if (command2->mCommand != command->mCommand) continue;
switch (command->mCommand) {
case SET_PARAMETERS: {
@@ -817,6 +826,31 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(sp<AudioCommand>& c
// command status as the command is now delayed
delayMs = 1;
} break;
+
+ case CREATE_AUDIO_PATCH:
+ case RELEASE_AUDIO_PATCH: {
+ audio_patch_handle_t handle;
+ if (command->mCommand == CREATE_AUDIO_PATCH) {
+ handle = ((CreateAudioPatchData *)command->mParam.get())->mHandle;
+ } else {
+ handle = ((ReleaseAudioPatchData *)command->mParam.get())->mHandle;
+ }
+ audio_patch_handle_t handle2;
+ if (command2->mCommand == CREATE_AUDIO_PATCH) {
+ handle2 = ((CreateAudioPatchData *)command2->mParam.get())->mHandle;
+ } else {
+ handle2 = ((ReleaseAudioPatchData *)command2->mParam.get())->mHandle;
+ }
+ if (handle != handle2) break;
+ ALOGV("Filtering out %s audio patch command for handle %d",
+ (command->mCommand == CREATE_AUDIO_PATCH) ? "create" : "release", handle);
+ 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 START_TONE:
case STOP_TONE:
default:
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index abe1235..33bdaa3 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -938,7 +938,20 @@ void CameraClient::copyFrameAndPostCopiedFrame(
}
previewBuffer = mPreviewBuffer;
- memcpy(previewBuffer->base(), (uint8_t *)heap->base() + offset, size);
+ void* previewBufferBase = previewBuffer->base();
+ void* heapBase = heap->base();
+
+ if (heapBase == MAP_FAILED) {
+ ALOGE("%s: Failed to mmap heap for preview frame.", __FUNCTION__);
+ mLock.unlock();
+ return;
+ } else if (previewBufferBase == MAP_FAILED) {
+ ALOGE("%s: Failed to mmap preview buffer for preview frame.", __FUNCTION__);
+ mLock.unlock();
+ return;
+ }
+
+ memcpy(previewBufferBase, (uint8_t *) heapBase + offset, size);
sp<MemoryBase> frame = new MemoryBase(previewBuffer, 0, size);
if (frame == 0) {
diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
index cb9aca6..9849f4d 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
@@ -445,11 +445,18 @@ CaptureSequencer::CaptureState CaptureSequencer::manageStandardPrecaptureWait(
if (mNewAEState) {
if (!mAeInPrecapture) {
// Waiting to see PRECAPTURE state
- if (mAETriggerId == mTriggerId &&
- mAEState == ANDROID_CONTROL_AE_STATE_PRECAPTURE) {
- ALOGV("%s: Got precapture start", __FUNCTION__);
- mAeInPrecapture = true;
- mTimeoutCount = kMaxTimeoutsForPrecaptureEnd;
+ if (mAETriggerId == mTriggerId) {
+ if (mAEState == ANDROID_CONTROL_AE_STATE_PRECAPTURE) {
+ ALOGV("%s: Got precapture start", __FUNCTION__);
+ mAeInPrecapture = true;
+ mTimeoutCount = kMaxTimeoutsForPrecaptureEnd;
+ } else if (mAEState == ANDROID_CONTROL_AE_STATE_CONVERGED ||
+ mAEState == ANDROID_CONTROL_AE_STATE_FLASH_REQUIRED) {
+ // It is legal to transit to CONVERGED or FLASH_REQUIRED
+ // directly after a trigger.
+ ALOGV("%s: AE is already in good state, start capture", __FUNCTION__);
+ return STANDARD_CAPTURE;
+ }
}
} else {
// Waiting to see PRECAPTURE state end
generated by cgit v1.1 at 2024-06-10 02:14:57 +0000