Update sampling rate to 192kHz for devices
Change-Id: I0a83206be51d7ae18ccf85b94b2127356307be69
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 53b8c13..6cf2ca9 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -143,6 +143,16 @@
return required * (double)speed + 1 + 1; // accounting for rounding dependencies
}
+// Identifies sample rates that we associate with music
+// and thus eligible for better resampling and fast capture.
+// This is somewhat less than 44100 to allow for pitch correction
+// involving resampling as well as asynchronous resampling.
+#define AUDIO_PROCESSING_MUSIC_RATE 40000
+
+static inline bool isMusicRate(uint32_t sampleRate) {
+ return sampleRate >= AUDIO_PROCESSING_MUSIC_RATE;
+}
+
} // namespace android
// ---------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 7040af4..1348d08 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -708,11 +708,10 @@
// FIXME this is flawed for dynamic sample rates, as we choose the resampler
// quality level based on the initial ratio, but that could change later.
// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
- if (!((trackSampleRate == 44100 && devSampleRate == 48000) ||
- (trackSampleRate == 48000 && devSampleRate == 44100))) {
- quality = AudioResampler::DYN_LOW_QUALITY;
- } else {
+ if (isMusicRate(trackSampleRate)) {
quality = AudioResampler::DEFAULT_QUALITY;
+ } else {
+ quality = AudioResampler::DYN_LOW_QUALITY;
}
// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7feb63b..5ef017f 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5328,11 +5328,11 @@
}
initFastCapture =
// either capture sample rate is same as (a reasonable) primary output sample rate
- (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) &&
+ ((isMusicRate(primaryOutputSampleRate) &&
(mSampleRate == primaryOutputSampleRate)) ||
// or primary output sample rate is unknown, and capture sample rate is reasonable
((primaryOutputSampleRate == 0) &&
- ((mSampleRate == 44100 || mSampleRate == 48000)))) &&
+ isMusicRate(mSampleRate))) &&
// and the buffer size is < 12 ms
(mFrameCount * 1000) / mSampleRate < 12;
break;
@@ -6435,6 +6435,9 @@
return NO_ERROR;
}
+ ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
+ " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u",
+ srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
const bool valid =
audio_is_input_channel(srcChannelMask)
&& audio_is_input_channel(dstChannelMask)
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index a2327ee..4bda1f7 100755
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -20,7 +20,7 @@
// For mixed output and inputs, the policy will use max mixer sampling rates.
// Do not limit sampling rate otherwise
-#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_SAMPLING_RATE 192000
// For mixed output and inputs, the policy will use max mixer channel count.
// Do not limit channel count otherwise
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index 2e4d423..afcd073 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -611,9 +611,13 @@
// For mixed output and inputs, use max mixer sampling rates. Do not
// limit sampling rate otherwise
+ // For inputs, also see checkCompatibleSamplingRate().
if (mType != AUDIO_PORT_TYPE_MIX) {
maxRate = UINT_MAX;
}
+ // TODO: should mSamplingRates[] be ordered in terms of our preference
+ // and we return the first (and hence most preferred) match? This is of concern if
+ // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
for (size_t i = 0; i < mSamplingRates.size(); i ++) {
if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
samplingRate = mSamplingRates[i];
