Add format conversion in AudioRecord.
In audio flinger, it is required that the client's format matches the
hardware format to get fast mode. However, format conversion is not a
time consuming operation. In that case, adding format conversion in
AudioRecord can help the client get fast mode if only format is
different from the hardware one. This can help improve latency.
Test: atest AudioRecordTest, capture audio in fast/non-fast mode
Bug: 79156275
Bug: 188843084
Change-Id: I1baaa9a6054bb84412c31a159c6d25ee498822cf
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index a02a373..d3c5231 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -22,6 +22,7 @@
#include <android-base/macros.h>
#include <sys/resource.h>
+#include <audio_utils/format.h>
#include <audiomanager/AudioManager.h>
#include <audiomanager/IAudioManager.h>
#include <binder/Binder.h>
@@ -858,6 +859,10 @@
mRoutedDeviceId = output.selectedDeviceId;
mSessionId = output.sessionId;
mSampleRate = output.sampleRate;
+ mServerConfig = output.serverConfig;
+ mServerFrameSize = audio_bytes_per_frame(
+ audio_channel_count_from_in_mask(mServerConfig.channel_mask), mServerConfig.format);
+ mServerSampleSize = audio_bytes_per_sample(mServerConfig.format);
if (output.cblk == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
@@ -921,6 +926,10 @@
mNotificationFramesReq, output.notificationFrameCount, output.frameCount);
}
mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
+ if (mServerConfig.format != mFormat && mCbf != nullptr) {
+ mFormatConversionBufRaw = std::make_unique<uint8_t[]>(mNotificationFramesAct * mFrameSize);
+ mFormatConversionBuffer.raw = mFormatConversionBufRaw.get();
+ }
//mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creation
if (mDeviceCallback != 0) {
@@ -947,7 +956,7 @@
}
// update proxy
- mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
+ mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mServerFrameSize);
mProxy->setEpoch(epoch);
mProxy->setMinimum(mNotificationFramesAct);
@@ -1077,7 +1086,7 @@
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mFrameSize;
+ audioBuffer->size = buffer.mFrameCount * mServerFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
@@ -1090,7 +1099,7 @@
{
// FIXME add error checking on mode, by adding an internal version
- size_t stepCount = audioBuffer->size / mFrameSize;
+ size_t stepCount = audioBuffer->frameCount;
if (stepCount == 0) {
return;
}
@@ -1152,8 +1161,9 @@
return ssize_t(err);
}
- size_t bytesRead = audioBuffer.size;
- memcpy(buffer, audioBuffer.i8, bytesRead);
+ size_t bytesRead = audioBuffer.frameCount * mFrameSize;
+ memcpy_by_audio_format(buffer, mFormat, audioBuffer.raw, mServerConfig.format,
+ audioBuffer.size / mServerSampleSize);
buffer = ((char *) buffer) + bytesRead;
userSize -= bytesRead;
read += bytesRead;
@@ -1350,9 +1360,19 @@
}
}
- size_t reqSize = audioBuffer.size;
- mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
- size_t readSize = audioBuffer.size;
+ Buffer* buffer = &audioBuffer;
+ if (mServerConfig.format != mFormat) {
+ buffer = &mFormatConversionBuffer;
+ buffer->frameCount = audioBuffer.frameCount;
+ buffer->size = buffer->frameCount * mFrameSize;
+ buffer->sequence = audioBuffer.sequence;
+ memcpy_by_audio_format(buffer->raw, mFormat, audioBuffer.raw,
+ mServerConfig.format, audioBuffer.size / mServerSampleSize);
+ }
+
+ size_t reqSize = buffer->size;
+ mCbf(EVENT_MORE_DATA, mUserData, buffer);
+ size_t readSize = buffer->size;
// Validate on returned size
if (ssize_t(readSize) < 0 || readSize > reqSize) {
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 5f12f71..6d2ec93 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -195,6 +195,8 @@
aidl.buffers = VALUE_OR_RETURN(legacy2aidl_NullableIMemory_SharedFileRegion(buffers));
aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(portId));
aidl.audioRecord = audioRecord;
+ aidl.serverConfig = VALUE_OR_RETURN(
+ legacy2aidl_audio_config_base_t_AudioConfigBase(serverConfig, true /*isInput*/));
return aidl;
}
@@ -215,6 +217,8 @@
legacy.buffers = VALUE_OR_RETURN(aidl2legacy_NullableSharedFileRegion_IMemory(aidl.buffers));
legacy.portId = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
legacy.audioRecord = aidl.audioRecord;
+ legacy.serverConfig = VALUE_OR_RETURN(
+ aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.serverConfig, true /*isInput*/));
return legacy;
}
diff --git a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
index d78b3fc..61195d0 100644
--- a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
@@ -16,6 +16,7 @@
package android.media;
+import android.media.AudioConfigBase;
import android.media.IAudioRecord;
import android.media.SharedFileRegion;
@@ -43,4 +44,5 @@
int portId;
/** The newly created record. */
@nullable IAudioRecord audioRecord;
+ AudioConfigBase serverConfig;
}
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index f17ee3a..a32c595 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -760,6 +760,13 @@
bool mTimestampRetrogradePositionReported = false; // reduce log spam
bool mTimestampRetrogradeTimeReported = false; // reduce log spam
+ // Format conversion. Maybe needed for adding fast tracks whose format is different from server.
+ audio_config_base_t mServerConfig;
+ size_t mServerFrameSize;
+ size_t mServerSampleSize;
+ std::unique_ptr<uint8_t[]> mFormatConversionBufRaw;
+ Buffer mFormatConversionBuffer;
+
private:
class DeathNotifier : public IBinder::DeathRecipient {
public:
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 8632abb..a2ce145 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -166,6 +166,7 @@
sp<IMemory> buffers;
audio_port_handle_t portId;
sp<media::IAudioRecord> audioRecord;
+ audio_config_base_t serverConfig;
ConversionResult<media::CreateRecordResponse> toAidl() const;
static ConversionResult<CreateRecordOutput>
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a1fb304..445de02 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2142,6 +2142,20 @@
goto Exit;
}
+ if (recordTrack->isFastTrack()) {
+ output.serverConfig = {
+ thread->sampleRate(),
+ thread->channelMask(),
+ thread->format()
+ };
+ } else {
+ output.serverConfig = {
+ recordTrack->sampleRate(),
+ recordTrack->channelMask(),
+ recordTrack->format()
+ };
+ }
+
// Check if one effect chain was awaiting for an AudioRecord to be created on this
// session and move it to this thread.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 9665424..a07b9b1 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8016,6 +8016,7 @@
if (
// we formerly checked for a callback handler (non-0 tid),
// but that is no longer required for TRANSFER_OBTAIN mode
+ // No need to match hardware format, format conversion will be done in client side.
//
// Frame count is not specified (0), or is less than or equal the pipe depth.
// It is OK to provide a higher capacity than requested.
@@ -8023,8 +8024,6 @@
(frameCount <= mPipeFramesP2) &&
// PCM data
audio_is_linear_pcm(format) &&
- // hardware format
- (format == mFormat) &&
// hardware channel mask
(channelMask == mChannelMask) &&
// hardware sample rate