Merge "Update sampling rate to 192kHz for devices" into mnc-dev

5 files changed, 23 insertions, 7 deletions

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 @@ static inline size_t sourceFramesNeededWithTimestretch(
     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 959c140..586c737 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -708,11 +708,10 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam
                 // 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 2722d32..594ed05 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5345,11 +5345,11 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
         }
         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;
@@ -6452,6 +6452,9 @@ status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
         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 e6a767f..4eef02f 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 @@ uint32_t AudioPort::pickSamplingRate() const
 
     // 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];