AAudio: Add sample rate conversion to MMAP
This CL adds sample rate conversion to the MMAP path in AAudio.
AAudio no longer needs to use AudioFlinger's sample rate converter for
MMAP streams. This effectively reduces the latency from 100ms to 30ms
on speaker, 3.5mm, and USB if MMAP is enabled and sample rate conversion
is needed.
A sample rate converter is added to the AAudio flowgraph. Since input
and output sample rates are different for the flowgraph, a variable
number of frames must be used. 8 frames are passed through the flowgraph
at any given point of times and the output of the flowgraph pulls as
much data as there exists in the flowgraph.
Burst sizes, capacity, latency, frames read, and frames written are all
changed reflect the sample rate of the app.
Flowgraph was also added to input MMAP streams.
Bug: 219533889
Test: OboeTester Test Input/Output with a variety of sample rates and formats
Test: atest AAudioTests
Change-Id: I9b6a2694921ea558fb91a231f956f638d3342837
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index f5cc2be..47518d7 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -47,6 +47,27 @@
}
+aaudio_result_t AudioStreamInternalCapture::open(const AudioStreamBuilder &builder) {
+ aaudio_result_t result = AudioStreamInternal::open(builder);
+ if (result == AAUDIO_OK) {
+ result = mFlowGraph.configure(getDeviceFormat(),
+ getDeviceChannelCount(),
+ getDeviceSampleRate(),
+ getFormat(),
+ getSamplesPerFrame(),
+ getSampleRate(),
+ getRequireMonoBlend(),
+ false /* useVolumeRamps */,
+ getAudioBalance(),
+ aaudio::resampler::MultiChannelResampler::Quality::Medium);
+
+ if (result != AAUDIO_OK) {
+ safeReleaseClose();
+ }
+ }
+ return result;
+}
+
void AudioStreamInternalCapture::advanceClientToMatchServerPosition(int32_t serverMargin) {
int64_t readCounter = mAudioEndpoint->getDataReadCounter();
int64_t writeCounter = mAudioEndpoint->getDataWriteCounter() + serverMargin;
@@ -149,7 +170,8 @@
// Calculate frame position based off of the readCounter because
// the writeCounter might have just advanced in the background,
// causing us to sleep until a later burst.
- int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + getFramesPerBurst();
+ const int64_t nextPosition = mAudioEndpoint->getDataReadCounter() +
+ getDeviceFramesPerBurst();
wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
}
break;
@@ -166,42 +188,73 @@
aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer,
int32_t numFrames) {
- // ALOGD("readNowWithConversion(%p, %d)",
- // buffer, numFrames);
WrappingBuffer wrappingBuffer;
- uint8_t *destination = (uint8_t *) buffer;
- int32_t framesLeft = numFrames;
+ uint8_t *byteBuffer = (uint8_t *) buffer;
+ int32_t framesLeftInByteBuffer = numFrames;
+
+ if (framesLeftInByteBuffer > 0) {
+ // Pull data from the flowgraph in case there is residual data.
+ const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.pull(
+ (void *)byteBuffer,
+ framesLeftInByteBuffer);
+
+ const int32_t numBytesActuallyWrittenToByteBuffer =
+ framesActuallyWrittenToByteBuffer * getBytesPerFrame();
+ byteBuffer += numBytesActuallyWrittenToByteBuffer;
+ framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
+ }
mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);
- // Read data in one or two parts.
- for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
- int32_t framesToProcess = framesLeft;
- const int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
- if (framesAvailable <= 0) break;
+ // Write data in one or two parts.
+ int partIndex = 0;
+ int framesReadFromAudioEndpoint = 0;
+ while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
+ const int32_t totalFramesInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
+ int32_t framesAvailableInWrappingBuffer = totalFramesInWrappingBuffer;
+ uint8_t *currentWrappingBuffer = (uint8_t *) wrappingBuffer.data[partIndex];
- if (framesToProcess > framesAvailable) {
- framesToProcess = framesAvailable;
+ // Put data from the wrapping buffer into the flowgraph 8 frames at a time.
+ // Continuously pull as much data as possible from the flowgraph into the byte buffer.
+ // The return value of mFlowGraph.process is the number of frames actually pulled.
+ while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
+ const int32_t framesToReadFromWrappingBuffer = std::min(flowgraph::kDefaultBufferSize,
+ framesAvailableInWrappingBuffer);
+
+ const int32_t numBytesToReadFromWrappingBuffer = getBytesPerDeviceFrame() *
+ framesToReadFromWrappingBuffer;
+
+ // If framesActuallyWrittenToByteBuffer < framesLeftInByteBuffer, it is guaranteed
+ // that all the data is pulled. If there is no more space in the byteBuffer, the
+ // remaining data will be pulled in the following readNowWithConversion().
+ const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.process(
+ (void *)currentWrappingBuffer,
+ framesToReadFromWrappingBuffer,
+ (void *)byteBuffer,
+ framesLeftInByteBuffer);
+
+ const int32_t numBytesActuallyWrittenToByteBuffer =
+ framesActuallyWrittenToByteBuffer * getBytesPerFrame();
+ byteBuffer += numBytesActuallyWrittenToByteBuffer;
+ framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
+ currentWrappingBuffer += numBytesToReadFromWrappingBuffer;
+ framesAvailableInWrappingBuffer -= framesToReadFromWrappingBuffer;
+
+ //ALOGD("%s() numBytesActuallyWrittenToByteBuffer %d, framesLeftInByteBuffer %d"
+ // "framesAvailableInWrappingBuffer %d, framesReadFromAudioEndpoint %d"
+ // , __func__, numBytesActuallyWrittenToByteBuffer, framesLeftInByteBuffer,
+ // framesAvailableInWrappingBuffer, framesReadFromAudioEndpoint);
}
-
- const int32_t numBytes = getBytesPerFrame() * framesToProcess;
- const int32_t numSamples = framesToProcess * getSamplesPerFrame();
-
- const audio_format_t sourceFormat = getDeviceFormat();
- const audio_format_t destinationFormat = getFormat();
-
- memcpy_by_audio_format(destination, destinationFormat,
- wrappingBuffer.data[partIndex], sourceFormat, numSamples);
-
- destination += numBytes;
- framesLeft -= framesToProcess;
+ framesReadFromAudioEndpoint += totalFramesInWrappingBuffer -
+ framesAvailableInWrappingBuffer;
+ partIndex++;
}
- int32_t framesProcessed = numFrames - framesLeft;
- mAudioEndpoint->advanceReadIndex(framesProcessed);
+ // The audio endpoint should reference the number of frames written to the wrapping buffer.
+ mAudioEndpoint->advanceReadIndex(framesReadFromAudioEndpoint);
- //ALOGD("readNowWithConversion() returns %d", framesProcessed);
- return framesProcessed;
+ // The internal code should use the number of frames read from the app.
+ return numFrames - framesLeftInByteBuffer;
}
int64_t AudioStreamInternalCapture::getFramesWritten() {