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
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 @@
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 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 @@
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 @@
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 @@
// initialize common members
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
-
}
AudioResampler::~AudioResampler() {