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/AAudioFlowGraph.cpp b/media/libaaudio/src/client/AAudioFlowGraph.cpp
index 5444565..69be050 100644
--- a/media/libaaudio/src/client/AAudioFlowGraph.cpp
+++ b/media/libaaudio/src/client/AAudioFlowGraph.cpp
@@ -39,18 +39,21 @@
aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat,
int32_t sourceChannelCount,
+ int32_t sourceSampleRate,
audio_format_t sinkFormat,
int32_t sinkChannelCount,
+ int32_t sinkSampleRate,
bool useMonoBlend,
+ bool useVolumeRamps,
float audioBalance,
- bool isExclusive) {
+ aaudio::resampler::MultiChannelResampler::Quality resamplerQuality) {
FlowGraphPortFloatOutput *lastOutput = nullptr;
- // TODO change back to ALOGD
- ALOGI("%s() source format = 0x%08x, channels = %d, sink format = 0x%08x, channels = %d, "
- "useMonoBlend = %d, audioBalance = %f, isExclusive %d",
- __func__, sourceFormat, sourceChannelCount, sinkFormat, sinkChannelCount,
- useMonoBlend, audioBalance, isExclusive);
+ ALOGD("%s() source format = 0x%08x, channels = %d, sample rate = %d, "
+ "sink format = 0x%08x, channels = %d, sample rate = %d, "
+ "useMonoBlend = %d, audioBalance = %f, useVolumeRamps %d",
+ __func__, sourceFormat, sourceChannelCount, sourceSampleRate, sinkFormat,
+ sinkChannelCount, sinkSampleRate, useMonoBlend, audioBalance, useVolumeRamps);
switch (sourceFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
@@ -85,6 +88,15 @@
lastOutput = &mLimiter->output;
}
+ if (sourceSampleRate != sinkSampleRate) {
+ mResampler.reset(aaudio::resampler::MultiChannelResampler::make(sinkChannelCount,
+ sourceSampleRate, sinkSampleRate, resamplerQuality));
+ mRateConverter = std::make_unique<SampleRateConverter>(sinkChannelCount,
+ *mResampler);
+ lastOutput->connect(&mRateConverter->input);
+ lastOutput = &mRateConverter->output;
+ }
+
// Expand the number of channels if required.
if (sourceChannelCount == 1 && sinkChannelCount > 1) {
mChannelConverter = std::make_unique<MonoToMultiConverter>(sinkChannelCount);
@@ -95,8 +107,7 @@
return AAUDIO_ERROR_UNIMPLEMENTED;
}
- // Apply volume ramps for only exclusive streams.
- if (isExclusive) {
+ if (useVolumeRamps) {
// Apply volume ramps to set the left/right audio balance and target volumes.
// The signals will be decoupled, volume ramps will be applied, before the signals are
// combined again.
@@ -137,9 +148,14 @@
return AAUDIO_OK;
}
-void AAudioFlowGraph::process(const void *source, void *destination, int32_t numFrames) {
- mSource->setData(source, numFrames);
- mSink->read(destination, numFrames);
+int32_t AAudioFlowGraph::pull(void *destination, int32_t targetFramesToRead) {
+ return mSink->read(destination, targetFramesToRead);
+}
+
+int32_t AAudioFlowGraph::process(const void *source, int32_t numFramesToWrite, void *destination,
+ int32_t targetFramesToRead) {
+ mSource->setData(source, numFramesToWrite);
+ return mSink->read(destination, targetFramesToRead);
}
/**