Extend downsampling ratios greater than 2:1

Also improve robustness to choice of sampling rate or buffer size
such that increasing either by 10x does not cause overflow.

Bug: 12979141
Bug: 15933066
Change-Id: If7989bd745d1bee3bdf811b8b7c978543ccafb65

4 files changed, 37 insertions, 8 deletions

diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
new file mode 100644
index 0000000..97847a0
--- /dev/null
+++ b/include/media/AudioResamplerPublic.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H
+#define ANDROID_AUDIO_RESAMPLER_PUBLIC_H
+
+// AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
+// audio sample rate and the target rate when downsampling,
+// as permitted in the audio framework, e.g. AudioTrack and AudioFlinger.
+// In practice, it is not recommended to downsample more than 6:1
+// for best audio quality, even though the audio framework permits a larger
+// downsampling ratio.
+// TODO: replace with an API
+#define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256
+
+#endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index b5c9125..d87e6f5 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -28,6 +28,7 @@
 #include <utils/Log.h>
 #include <private/media/AudioTrackShared.h>
 #include <media/IAudioFlinger.h>
+#include <media/AudioResamplerPublic.h>
 
 #define WAIT_PERIOD_MS                  10
 #define WAIT_STREAM_END_TIMEOUT_SEC     120
@@ -82,7 +83,7 @@ status_t AudioTrack::getMinFrameCount(
     }
 
     *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
-            afFrameCount * minBufCount * sampleRate / afSampleRate;
+            afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate;
     // The formula above should always produce a non-zero value, but return an error
     // in the unlikely event that it does not, as that's part of the API contract.
     if (*frameCount == 0) {
@@ -646,8 +647,7 @@ status_t AudioTrack::setSampleRate(uint32_t rate)
     if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
         return NO_INIT;
     }
-    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (rate == 0 || rate > afSamplingRate*2 ) {
+    if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
         return BAD_VALUE;
     }
 
@@ -1002,7 +1002,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
             minBufCount = nBuffering;
         }
 
-        size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
+        size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate;
         ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
                 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 159ab70..0eeb201 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -393,7 +393,7 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate)
     mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)
             * phaseWrapLimit / oldPhaseWrapLimit;
     mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.
-    mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit)
+    mPhaseIncrement = static_cast<uint32_t>(static_cast<uint64_t>(phaseWrapLimit)
             * inSampleRate / mSampleRate);
 
     // determine which resampler to use
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7d583bb..30cebf4 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -26,6 +26,7 @@
 #include <sys/stat.h>
 #include <cutils/properties.h>
 #include <media/AudioParameter.h>
+#include <media/AudioResamplerPublic.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 
@@ -1479,8 +1480,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
                 lStatus = BAD_VALUE;
                 goto Exit;
         }
-        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-        if (sampleRate > mSampleRate*2) {
+        if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
             ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
             lStatus = BAD_VALUE;
             goto Exit;
@@ -3500,7 +3500,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                 AudioMixer::TRACK,
                 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
             // limit track sample rate to 2 x output sample rate, which changes at re-configuration
-            uint32_t maxSampleRate = mSampleRate * 2;
+            uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
             uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
             if (reqSampleRate == 0) {
                 reqSampleRate = mSampleRate;