AAudio: Fix channel rate conversion in flowgraph
In ag/23541584, we added sample rate conversion. However, we introduced
two bugs in regards to channel count.
1. The flowgraph uses the sink channel count instead of the source.
2. getBytesPerDeviceFrame() doesn't use the device channel count.
This CL fixes both issues.
Bug: 292246705
Bug: 219533889
Test: Test Output in OboeTester with a variety of CHs, SRs, FMTs
Test: OboeTester Data Paths test
Change-Id: I74fe810c450bdd03d2765dc5ba9857ba5d03bf97
diff --git a/media/libaaudio/src/client/AAudioFlowGraph.cpp b/media/libaaudio/src/client/AAudioFlowGraph.cpp
index 69be050..b7e0ae6 100644
--- a/media/libaaudio/src/client/AAudioFlowGraph.cpp
+++ b/media/libaaudio/src/client/AAudioFlowGraph.cpp
@@ -89,9 +89,9 @@
}
if (sourceSampleRate != sinkSampleRate) {
- mResampler.reset(aaudio::resampler::MultiChannelResampler::make(sinkChannelCount,
+ mResampler.reset(aaudio::resampler::MultiChannelResampler::make(sourceChannelCount,
sourceSampleRate, sinkSampleRate, resamplerQuality));
- mRateConverter = std::make_unique<SampleRateConverter>(sinkChannelCount,
+ mRateConverter = std::make_unique<SampleRateConverter>(sourceChannelCount,
*mResampler);
lastOutput->connect(&mRateConverter->input);
lastOutput = &mRateConverter->output;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index d75832f..9b1ad72 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -134,8 +134,6 @@
request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
- mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
-
mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput);
if (getServiceHandle() < 0
&& (request.getConfiguration().getSamplesPerFrame() == 1
@@ -179,8 +177,6 @@
setChannelMask(configurationOutput.getChannelMask());
}
- mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
-
setDeviceId(configurationOutput.getDeviceId());
setSessionId(configurationOutput.getSessionId());
setSharingMode(configurationOutput.getSharingMode());
@@ -205,6 +201,7 @@
// Save device format so we can do format conversion and volume scaling together.
setDeviceFormat(configurationOutput.getFormat());
+ setDeviceSamplesPerFrame(configurationOutput.getSamplesPerFrame());
setHardwareSamplesPerFrame(configurationOutput.getHardwareSamplesPerFrame());
setHardwareSampleRate(configurationOutput.getHardwareSampleRate());
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 0dc9995..a5981b1 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -138,8 +138,6 @@
// Calculate timeout for an operation involving framesPerOperation.
int64_t calculateReasonableTimeout(int32_t framesPerOperation);
- int32_t getDeviceChannelCount() const { return mDeviceChannelCount; }
-
/**
* @return true if running in audio service, versus in app process
*/
@@ -213,10 +211,6 @@
int64_t mServiceLatencyNanos = 0;
- // Sometimes the hardware is operating with a different channel count from the app.
- // Then we require conversion in AAudio.
- int32_t mDeviceChannelCount = 0;
-
int32_t mBufferSizeInFrames = 0; // local threshold to control latency
int32_t mDeviceBufferSizeInFrames = 0;
int32_t mBufferCapacityInFrames = 0;
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 47518d7..7d7b4ef 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -51,7 +51,7 @@
aaudio_result_t result = AudioStreamInternal::open(builder);
if (result == AAUDIO_OK) {
result = mFlowGraph.configure(getDeviceFormat(),
- getDeviceChannelCount(),
+ getDeviceSamplesPerFrame(),
getDeviceSampleRate(),
getFormat(),
getSamplesPerFrame(),
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 99aa910..ac927ae 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -54,7 +54,7 @@
getSamplesPerFrame(),
getSampleRate(),
getDeviceFormat(),
- getDeviceChannelCount(),
+ getDeviceSamplesPerFrame(),
getDeviceSampleRate(),
getRequireMonoBlend(),
useVolumeRamps,
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 1649eaf..f2f5cac 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -252,6 +252,10 @@
return mSamplesPerFrame;
}
+ aaudio_result_t getDeviceSamplesPerFrame() const {
+ return mDeviceSamplesPerFrame;
+ }
+
aaudio_result_t getHardwareSamplesPerFrame() const {
return mHardwareSamplesPerFrame;
}
@@ -334,10 +338,10 @@
}
/**
- * This is only valid after setChannelMask() and setDeviceFormat() have been called.
+ * This is only valid after setDeviceSamplesPerFrame() and setDeviceFormat() have been called.
*/
int32_t getBytesPerDeviceFrame() const {
- return getSamplesPerFrame() * audio_bytes_per_sample(getDeviceFormat());
+ return getDeviceSamplesPerFrame() * audio_bytes_per_sample(getDeviceFormat());
}
virtual int64_t getFramesWritten() = 0;
@@ -377,6 +381,11 @@
mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
}
+ void setDeviceSamplesPerFrame(int32_t deviceSamplesPerFrame) {
+ mDeviceSamplesPerFrame = deviceSamplesPerFrame;
+ }
+
+
/**
* @return true if data callback has been specified
*/
@@ -748,6 +757,7 @@
// These do not change after open().
int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+ int32_t mDeviceSamplesPerFrame = AAUDIO_UNSPECIFIED;
int32_t mHardwareSamplesPerFrame = AAUDIO_UNSPECIFIED;
aaudio_channel_mask_t mChannelMask = AAUDIO_UNSPECIFIED;
int32_t mSampleRate = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 7b4821f..fe4bf2c 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -208,6 +208,8 @@
setBufferCapacity(getBufferCapacityFromDevice());
setFramesPerBurst(getFramesPerBurstFromDevice());
+ // Use the same values for device values.
+ setDeviceSamplesPerFrame(getSamplesPerFrame());
setDeviceSampleRate(mAudioRecord->getSampleRate());
setDeviceBufferCapacity(getBufferCapacityFromDevice());
setDeviceFramesPerBurst(getFramesPerBurstFromDevice());
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 723b419..59fdabc 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -202,6 +202,9 @@
setSampleRate(mAudioTrack->getSampleRate());
setBufferCapacity(getBufferCapacityFromDevice());
setFramesPerBurst(getFramesPerBurstFromDevice());
+
+ // Use the same values for device values.
+ setDeviceSamplesPerFrame(getSamplesPerFrame());
setDeviceSampleRate(mAudioTrack->getSampleRate());
setDeviceBufferCapacity(getBufferCapacityFromDevice());
setDeviceFramesPerBurst(getFramesPerBurstFromDevice());