Replace int bitDepth with audio_format_t in Resampler

Remove mBitDepth from class (not used).
Replace with audio_format_t in factory method to distinguish
between float and pcm 16-bit.

Change-Id: I166860796c68285077ef4458d8758d19b82523f9

11 files changed, 54 insertions, 59 deletions

diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index af312c4..4dbbc43 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -763,21 +763,9 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
                     quality = AudioResampler::DEFAULT_QUALITY;
                 }
 
-                int bits;
-                switch (mMixerInFormat) {
-                case AUDIO_FORMAT_PCM_16_BIT:
-                    bits = 16;
-                    break;
-                case AUDIO_FORMAT_PCM_FLOAT:
-                    bits = 32; // 32 bits to the AudioResampler::create() indicates float.
-                    break;
-                default:
-                    LOG_ALWAYS_FATAL("Invalid mMixerInFormat: %#x", mMixerInFormat);
-                    break;
-                }
                 ALOGVV("Creating resampler with %d bits\n", bits);
                 resampler = AudioResampler::create(
-                        bits,
+                        mMixerInFormat,
                         // the resampler sees the number of channels after the downmixer, if any
                         (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
                         devSampleRate, quality);
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index b8a0357..38c9061 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -40,8 +40,8 @@ namespace android {
 
 class AudioResamplerOrder1 : public AudioResampler {
 public:
-    AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
-        AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
+    AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
+        AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
     }
     virtual void resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
@@ -145,7 +145,7 @@ static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, s
 static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
 static uint32_t currentMHz = 0;
 
-AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
+AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount,
         int32_t sampleRate, src_quality quality) {
 
     bool atFinalQuality;
@@ -216,33 +216,40 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
     default:
     case LOW_QUALITY:
         ALOGV("Create linear Resampler");
-        resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
         break;
     case MED_QUALITY:
         ALOGV("Create cubic Resampler");
-        resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerCubic(inChannelCount, sampleRate);
         break;
     case HIGH_QUALITY:
         ALOGV("Create HIGH_QUALITY sinc Resampler");
-        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerSinc(inChannelCount, sampleRate);
         break;
     case VERY_HIGH_QUALITY:
         ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
-        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
+        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+        resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);
         break;
     case DYN_LOW_QUALITY:
     case DYN_MED_QUALITY:
     case DYN_HIGH_QUALITY:
         ALOGV("Create dynamic Resampler = %d", quality);
-        if (bitDepth == 32) { /* bitDepth == 32 signals float precision */
-            resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,
-                    sampleRate, quality);
-        } else if (quality == DYN_HIGH_QUALITY) {
-            resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,
+        if (format == AUDIO_FORMAT_PCM_FLOAT) {
+            resampler = new AudioResamplerDyn<float, float, float>(inChannelCount,
                     sampleRate, quality);
         } else {
-            resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,
-                    sampleRate, quality);
+            LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+            if (quality == DYN_HIGH_QUALITY) {
+                resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount,
+                        sampleRate, quality);
+            } else {
+                resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount,
+                        sampleRate, quality);
+            }
         }
         break;
     }
@@ -252,18 +259,17 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
     return resampler;
 }
 
-AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
+AudioResampler::AudioResampler(int inChannelCount,
         int32_t sampleRate, src_quality quality) :
-    mBitDepth(bitDepth), mChannelCount(inChannelCount),
-            mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
-            mPhaseFraction(0), mLocalTimeFreq(0),
-            mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
-    // sanity check on format
-    if ((bitDepth != 16 && (quality < DYN_LOW_QUALITY || bitDepth != 32))
-            || inChannelCount < 1
+        mChannelCount(inChannelCount),
+        mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
+        mPhaseFraction(0), mLocalTimeFreq(0),
+        mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
+
+    if (inChannelCount < 1
             || inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) {
-        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d bits, %d channels",
-                quality, bitDepth, inChannelCount);
+        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
+                quality, inChannelCount);
     }
     if (sampleRate <= 0) {
         LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
@@ -272,7 +278,6 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
     // initialize common members
     mVolume[0] = mVolume[1] = 0;
     mBuffer.frameCount = 0;
-
 }
 
 AudioResampler::~AudioResampler() {
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index b84567e..be747f6 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -22,6 +22,7 @@
 #include <cutils/compiler.h>
 
 #include <media/AudioBufferProvider.h>
+#include <system/audio.h>
 
 namespace android {
 // ----------------------------------------------------------------------------
@@ -46,7 +47,7 @@ public:
         DYN_HIGH_QUALITY=7,
     };
 
-    static AudioResampler* create(int bitDepth, int inChannelCount,
+    static AudioResampler* create(audio_format_t format, int inChannelCount,
             int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
 
     virtual ~AudioResampler();
@@ -86,7 +87,7 @@ protected:
     // multiplier to calculate fixed point phase increment
     static const double kPhaseMultiplier;
 
-    AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
+    AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
 
     // prevent copying
     AudioResampler(const AudioResampler&);
@@ -94,7 +95,6 @@ protected:
 
     int64_t calculateOutputPTS(int outputFrameIndex);
 
-    const int32_t mBitDepth;
     const int32_t mChannelCount;
     const int32_t mSampleRate;
     int32_t mInSampleRate;
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 203b933..b315da5 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -28,8 +28,8 @@ namespace android {
 
 class AudioResamplerCubic : public AudioResampler {
 public:
-    AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :
-        AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {
+    AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
+        AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
     }
     virtual void resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 7ca10c1..043c803 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -162,9 +162,9 @@ void AudioResamplerDyn<TC, TI, TO>::Constants::set(
 }
 
 template<typename TC, typename TI, typename TO>
-AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(int bitDepth,
+AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(
         int inChannelCount, int32_t sampleRate, src_quality quality)
-    : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+    : AudioResampler(inChannelCount, sampleRate, quality),
       mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
     mCoefBuffer(NULL)
 {
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 3dced8a..3044bfb 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -41,7 +41,7 @@ namespace android {
 template<typename TC, typename TI, typename TO>
 class AudioResamplerDyn: public AudioResampler {
 public:
-    AudioResamplerDyn(int bitDepth, int inChannelCount,
+    AudioResamplerDyn(int inChannelCount,
             int32_t sampleRate, src_quality quality);
 
     virtual ~AudioResamplerDyn();
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 35553ef..60ff88e 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -452,9 +452,9 @@ int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a)
 
 // ----------------------------------------------------------------------------
 
-AudioResamplerSinc::AudioResamplerSinc(int bitDepth,
+AudioResamplerSinc::AudioResamplerSinc(
         int inChannelCount, int32_t sampleRate, src_quality quality)
-    : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+    : AudioResampler(inChannelCount, sampleRate, quality),
     mState(0), mImpulse(0), mRingFull(0), mFirCoefs(0)
 {
     /*
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 1ea4474..97ae3d0 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -34,7 +34,7 @@ typedef int32_t (*readResampleFirLerpIntBitsFn)();
 
 class AudioResamplerSinc : public AudioResampler {
 public:
-    AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate,
+    AudioResamplerSinc(int inChannelCount, int32_t sampleRate,
             src_quality quality = HIGH_QUALITY);
 
     virtual ~AudioResamplerSinc();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 4fbb973..0ef9fe5 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1895,7 +1895,8 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
     if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
             channelCount <= FCC_2) {
         // sink SR
-        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
+        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+                thread->mChannelCount, sampleRate);
         // source SR
         mResampler->setSampleRate(thread->mSampleRate);
         mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT);
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index e14b4ae..78c9927 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -14,8 +14,6 @@
  * limitations under the License.
  */
 
-#include "AudioResampler.h"
-#include <media/AudioBufferProvider.h>
 #include <unistd.h>
 #include <stdio.h>
 #include <stdlib.h>
@@ -30,6 +28,8 @@
 #include <audio_utils/primitives.h>
 #include <audio_utils/sndfile.h>
 #include <utils/Vector.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioResampler.h"
 
 using namespace android;
 
@@ -329,7 +329,7 @@ int main(int argc, char* argv[]) {
         printf("%zu input frames\n", input_frames);
     }
 
-    int bit_depth = useFloat ? 32 : 16;
+    audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
     int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples
     size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t));
     size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq;
@@ -342,7 +342,7 @@ int main(int argc, char* argv[]) {
         //
         // On fast devices, filters should be generated between 0.1ms - 1ms.
         // (single threaded).
-        AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
+        AudioResampler* resampler = AudioResampler::create(format, channels,
                 8000, quality);
         int looplimit = 100;
         timespec start, end;
@@ -380,7 +380,7 @@ int main(int argc, char* argv[]) {
     }
 
     void* output_vaddr = malloc(output_size);
-    AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
+    AudioResampler* resampler = AudioResampler::create(format, channels,
             output_freq, quality);
 
 
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index d76c376..987162c 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -69,7 +69,7 @@ void testBufferIncrement(size_t channels, bool useFloat,
         unsigned inputFreq, unsigned outputFreq,
         enum android::AudioResampler::src_quality quality)
 {
-    const int bits = useFloat ? 32 : 16;
+    const audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
     // create the provider
     std::vector<int> inputIncr;
     SignalProvider provider;
@@ -92,7 +92,7 @@ void testBufferIncrement(size_t channels, bool useFloat,
     const int volumePrecision = 12; /* typical unity gain */
     android::AudioResampler* resampler;
 
-    resampler = android::AudioResampler::create(bits, channels, outputFreq, quality);
+    resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
 
@@ -109,7 +109,7 @@ void testBufferIncrement(size_t channels, bool useFloat,
     resampler->reset();
 #else
     delete resampler;
-    resampler = android::AudioResampler::create(bits, channels, outputFreq, quality);
+    resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
 #endif
@@ -174,7 +174,8 @@ void testStopbandDownconversion(size_t channels,
     const int volumePrecision = 12; /* typical unity gain */
     android::AudioResampler* resampler;
 
-    resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
+    resampler = android::AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+            channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);