FlacDecoder: Enable float support

Test: MediaCodecTest#testFlacIdentity
Bug: 122117025
Change-Id: I9a80a61fbcfc615305b39befbbd9840748098a6a
diff --git a/media/codec2/components/flac/C2SoftFlacDec.cpp b/media/codec2/components/flac/C2SoftFlacDec.cpp
index f1e2f51..86b16e8 100644
--- a/media/codec2/components/flac/C2SoftFlacDec.cpp
+++ b/media/codec2/components/flac/C2SoftFlacDec.cpp
@@ -83,8 +83,21 @@
                 DefineParam(mInputMaxBufSize, C2_PARAMKEY_INPUT_MAX_BUFFER_SIZE)
                 .withConstValue(new C2StreamMaxBufferSizeInfo::input(0u, 32768))
                 .build());
+
+        addParameter(
+                DefineParam(mPcmEncodingInfo, C2_PARAMKEY_PCM_ENCODING)
+                .withDefault(new C2StreamPcmEncodingInfo::output(0u, C2Config::PCM_16))
+                .withFields({C2F(mPcmEncodingInfo, value).oneOf({
+                     C2Config::PCM_16,
+                     // C2Config::PCM_8,
+                     C2Config::PCM_FLOAT})
+                })
+                .withSetter((Setter<decltype(*mPcmEncodingInfo)>::StrictValueWithNoDeps))
+                .build());
     }
 
+    int32_t getPcmEncodingInfo() const { return mPcmEncodingInfo->value; }
+
 private:
     std::shared_ptr<C2StreamFormatConfig::input> mInputFormat;
     std::shared_ptr<C2StreamFormatConfig::output> mOutputFormat;
@@ -94,6 +107,7 @@
     std::shared_ptr<C2StreamChannelCountInfo::output> mChannelCount;
     std::shared_ptr<C2BitrateTuning::input> mBitrate;
     std::shared_ptr<C2StreamMaxBufferSizeInfo::input> mInputMaxBufSize;
+    std::shared_ptr<C2StreamPcmEncodingInfo::output> mPcmEncodingInfo;
 };
 
 C2SoftFlacDec::C2SoftFlacDec(
@@ -263,11 +277,11 @@
         return;
     }
 
-    size_t outSize;
-    if (mHasStreamInfo)
-        outSize = mStreamInfo.max_blocksize * mStreamInfo.channels * sizeof(short);
-    else
-        outSize = kMaxBlockSize * FLACDecoder::kMaxChannels * sizeof(short);
+    const bool outputFloat = mIntf->getPcmEncodingInfo() == C2Config::PCM_FLOAT;
+    const size_t sampleSize = outputFloat ? sizeof(float) : sizeof(short);
+    size_t outSize = mHasStreamInfo ?
+            mStreamInfo.max_blocksize * mStreamInfo.channels * sampleSize
+          : kMaxBlockSize * FLACDecoder::kMaxChannels * sampleSize;
 
     std::shared_ptr<C2LinearBlock> block;
     C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
@@ -284,9 +298,8 @@
         return;
     }
 
-    short *output = reinterpret_cast<short *>(wView.data());
     status_t decoderErr = mFLACDecoder->decodeOneFrame(
-                            input, inSize, output, &outSize);
+                            input, inSize, wView.data(), &outSize, outputFloat);
     if (decoderErr != OK) {
         ALOGE("process: FLACDecoder decodeOneFrame returns error %d", decoderErr);
         mSignalledError = true;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ceb8a13..2ea5286 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2183,7 +2183,8 @@
             err = setupG711Codec(encoder, sampleRate, numChannels);
         }
     } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_FLAC)) {
-        int32_t numChannels = 0, sampleRate = 0, compressionLevel = -1;
+        // numChannels needs to be set to properly communicate PCM values.
+        int32_t numChannels = 2, sampleRate = 44100, compressionLevel = -1;
         if (encoder &&
                 (!msg->findInt32("channel-count", &numChannels)
                         || !msg->findInt32("sample-rate", &sampleRate))) {
@@ -2209,7 +2210,7 @@
                 }
             }
             err = setupFlacCodec(
-                    encoder, numChannels, sampleRate, compressionLevel);
+                    encoder, numChannels, sampleRate, compressionLevel, pcmEncoding);
         }
     } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW)) {
         int32_t numChannels, sampleRate;
@@ -3033,8 +3034,8 @@
 }
 
 status_t ACodec::setupFlacCodec(
-        bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel) {
-
+        bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel,
+        AudioEncoding encoding) {
     if (encoder) {
         OMX_AUDIO_PARAM_FLACTYPE def;
         InitOMXParams(&def);
@@ -3057,7 +3058,8 @@
     return setupRawAudioFormat(
             encoder ? kPortIndexInput : kPortIndexOutput,
             sampleRate,
-            numChannels);
+            numChannels,
+            encoding);
 }
 
 status_t ACodec::setupRawAudioFormat(
@@ -3115,6 +3117,7 @@
     pcmParams.ePCMMode = OMX_AUDIO_PCMModeLinear;
 
     if (getOMXChannelMapping(numChannels, pcmParams.eChannelMapping) != OK) {
+        ALOGE("%s: incorrect numChannels: %d", __func__, numChannels);
         return OMX_ErrorNone;
     }
 
diff --git a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
index 4db0060..842a7ce 100644
--- a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
+++ b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
@@ -89,12 +89,12 @@
     def.eDir = OMX_DirOutput;
     def.nBufferCountMin = kNumOutputBuffers;
     def.nBufferCountActual = def.nBufferCountMin;
-    def.nBufferSize = 4096 * FLACDecoder::kMaxChannels;
+    def.nBufferSize = kNumSamplesPerFrame * FLACDecoder::kMaxChannels * sizeof(float);
     def.bEnabled = OMX_TRUE;
     def.bPopulated = OMX_FALSE;
     def.eDomain = OMX_PortDomainAudio;
     def.bBuffersContiguous = OMX_FALSE;
-    def.nBufferAlignment = 2;
+    def.nBufferAlignment = sizeof(float);
 
     def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
     def.format.audio.pNativeRender = NULL;
@@ -173,7 +173,7 @@
                 flacParams->nChannels = mStreamInfo.channels;
                 flacParams->nSampleRate = mStreamInfo.sample_rate;
             } else {
-                flacParams->nChannels = 1;
+                flacParams->nChannels = 2;
                 flacParams->nSampleRate = 44100;
             }
 
@@ -195,10 +195,10 @@
                 return OMX_ErrorBadPortIndex;
             }
 
-            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eNumData = mNumericalData;
             pcmParams->eEndian = OMX_EndianBig;
             pcmParams->bInterleaved = OMX_TRUE;
-            pcmParams->nBitPerSample = 16;
+            pcmParams->nBitPerSample = mBitsPerSample;
             pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
             pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
             pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
@@ -211,7 +211,7 @@
                 pcmParams->nChannels = mStreamInfo.channels;
                 pcmParams->nSamplingRate = mStreamInfo.sample_rate;
             } else {
-                pcmParams->nChannels = 1;
+                pcmParams->nChannels = 2;
                 pcmParams->nSamplingRate = 44100;
             }
 
@@ -281,6 +281,19 @@
                 return OMX_ErrorBadPortIndex;
             }
 
+            if (pcmParams->eNumData == OMX_NumericalDataFloat && pcmParams->nBitPerSample == 32) {
+                mNumericalData = OMX_NumericalDataFloat;
+                mBitsPerSample = 32;
+            } else if (pcmParams->eNumData == OMX_NumericalDataSigned
+                     && pcmParams->nBitPerSample == 16) {
+                mNumericalData = OMX_NumericalDataSigned;
+                mBitsPerSample = 16;
+            } else {
+                ALOGE("Invalid eNumData %d, nBitsPerSample %d",
+                        pcmParams->eNumData, pcmParams->nBitPerSample);
+                return OMX_ErrorUndefined;
+            }
+
             return OMX_ErrorNone;
         }
 
@@ -301,11 +314,13 @@
     List<BufferInfo *> &inQueue = getPortQueue(0);
     List<BufferInfo *> &outQueue = getPortQueue(1);
 
+    const bool outputFloat = mNumericalData == OMX_NumericalDataFloat;
+
     ALOGV("onQueueFilled %d/%d:", inQueue.empty(), outQueue.empty());
     while ((!inQueue.empty() || mSawInputEOS) && !outQueue.empty() && !mFinishedDecoder) {
         BufferInfo *outInfo = *outQueue.begin();
         OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
-        int16_t *outBuffer = reinterpret_cast<int16_t *>(outHeader->pBuffer + outHeader->nOffset);
+        void *outBuffer = reinterpret_cast<void *>(outHeader->pBuffer + outHeader->nOffset);
         size_t outBufferSize = outHeader->nAllocLen - outHeader->nOffset;
         int64_t timeStamp = 0;
 
@@ -374,7 +389,7 @@
             }
 
             status_t decoderErr = mFLACDecoder->decodeOneFrame(
-                    inBuffer, inBufferLength, outBuffer, &outBufferSize);
+                    inBuffer, inBufferLength, outBuffer, &outBufferSize, outputFloat);
             if (decoderErr != OK) {
                 ALOGE("onQueueFilled: FLACDecoder decodeOneFrame returns error %d", decoderErr);
                 mSignalledError = true;
@@ -393,7 +408,9 @@
                 continue;
             }
         } else if (mSawInputEOS) {
-            status_t decoderErr = mFLACDecoder->decodeOneFrame(NULL, 0, outBuffer, &outBufferSize);
+            status_t decoderErr = mFLACDecoder->decodeOneFrame(
+                    nullptr /* inBuffer */, 0 /* inBufferLen */,
+                    outBuffer, &outBufferSize, outputFloat);
             mFinishedDecoder = true;
             if (decoderErr != OK) {
                 ALOGE("onQueueFilled: FLACDecoder finish returns error %d", decoderErr);
@@ -456,7 +473,8 @@
             mOutputPortSettingsChange = AWAITING_ENABLED;
             PortInfo *info = editPortInfo(1 /* portIndex */);
             if (!info->mDef.bEnabled) {
-                info->mDef.nBufferSize = mStreamInfo.max_blocksize * mStreamInfo.channels * 2;
+                info->mDef.nBufferSize =
+                        mStreamInfo.max_blocksize * mStreamInfo.channels * sizeof(float);
             }
             break;
         }
diff --git a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.h b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.h
index b63f7ad..ba02074 100644
--- a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.h
+++ b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.h
@@ -45,10 +45,14 @@
     virtual void onReset() override;
 
 private:
+    static constexpr unsigned int kNumSamplesPerFrame = 2048; // adjusted based on stream.
+
     enum {
         kNumInputBuffers   = 4,
         kNumOutputBuffers  = 4,
     };
+    OMX_NUMERICALDATATYPE mNumericalData = OMX_NumericalDataSigned;
+    OMX_U32 mBitsPerSample = 16;
 
     FLACDecoder *mFLACDecoder;
     FLAC__StreamMetadata_StreamInfo mStreamInfo;
diff --git a/media/libstagefright/flac/dec/Android.bp b/media/libstagefright/flac/dec/Android.bp
index 751b053..307c9b0 100644
--- a/media/libstagefright/flac/dec/Android.bp
+++ b/media/libstagefright/flac/dec/Android.bp
@@ -27,11 +27,17 @@
     },
 
     static: {
-        whole_static_libs: ["libFLAC"],
+        whole_static_libs: [
+            "libFLAC",
+            "libaudioutils",
+        ],
     },
 
     shared: {
-        static_libs: ["libFLAC"],
+        static_libs: [
+            "libFLAC",
+            "libaudioutils",
+        ],
     },
 
     shared_libs: [
diff --git a/media/libstagefright/flac/dec/FLACDecoder.cpp b/media/libstagefright/flac/dec/FLACDecoder.cpp
index dfdc41c..cef0bc6 100644
--- a/media/libstagefright/flac/dec/FLACDecoder.cpp
+++ b/media/libstagefright/flac/dec/FLACDecoder.cpp
@@ -20,6 +20,7 @@
 
 #include "FLACDecoder.h"
 
+#include <audio_utils/primitives.h> // float_from_i32
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/MediaDefs.h>
@@ -117,104 +118,43 @@
     mErrorStatus = status;
 }
 
-// Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
+// Copy samples from FLAC native 32-bit non-interleaved to 16-bit signed
+// or 32-bit float interleaved.
+// TODO: Consider moving to audio_utils.  See similar code at FLACExtractor.cpp
 // These are candidates for optimization if needed.
-static void copyMono8(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
+static void copyTo16Signed(
+        short *dst,
+        const int *const *src,
         unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i] << 8;
-    }
-}
-
-static void copyStereo8(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i] << 8;
-        *dst++ = src[1][i] << 8;
-    }
-}
-
-static void copyMultiCh8(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned nChannels) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        for (unsigned c = 0; c < nChannels; ++c) {
-            *dst++ = src[c][i] << 8;
+        unsigned nChannels,
+        unsigned bitsPerSample) {
+    const int leftShift = 16 - (int)bitsPerSample; // cast to int to prevent unsigned overflow.
+    if (leftShift >= 0) {
+        for (unsigned i = 0; i < nSamples; ++i) {
+            for (unsigned c = 0; c < nChannels; ++c) {
+                *dst++ = src[c][i] << leftShift;
+            }
+        }
+    } else {
+        const int rightShift = -leftShift;
+        for (unsigned i = 0; i < nSamples; ++i) {
+            for (unsigned c = 0; c < nChannels; ++c) {
+                *dst++ = src[c][i] >> rightShift;
+            }
         }
     }
 }
 
-static void copyMono16(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
+static void copyToFloat(
+        float *dst,
+        const int *const *src,
         unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i];
-    }
-}
-
-static void copyStereo16(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i];
-        *dst++ = src[1][i];
-    }
-}
-
-static void copyMultiCh16(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned nChannels) {
+        unsigned nChannels,
+        unsigned bitsPerSample) {
+    const unsigned leftShift = 32 - bitsPerSample;
     for (unsigned i = 0; i < nSamples; ++i) {
         for (unsigned c = 0; c < nChannels; ++c) {
-            *dst++ = src[c][i];
-        }
-    }
-}
-
-// TODO: 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger
-static void copyMono24(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i] >> 8;
-    }
-}
-
-static void copyStereo24(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned /* nChannels */) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        *dst++ = src[0][i] >> 8;
-        *dst++ = src[1][i] >> 8;
-    }
-}
-
-static void copyMultiCh24(
-        int16_t *dst,
-        const int * src[FLACDecoder::kMaxChannels],
-        unsigned nSamples,
-        unsigned nChannels) {
-    for (unsigned i = 0; i < nSamples; ++i) {
-        for (unsigned c = 0; c < nChannels; ++c) {
-            *dst++ = src[c][i] >> 8;
+            *dst++ = float_from_i32(src[c][i] << leftShift);
         }
     }
 }
@@ -238,8 +178,7 @@
       mStreamInfoValid(false),
       mWriteRequested(false),
       mWriteCompleted(false),
-      mErrorStatus((FLAC__StreamDecoderErrorStatus) -1),
-      mCopy(nullptr) {
+      mErrorStatus((FLAC__StreamDecoderErrorStatus) -1) {
     ALOGV("ctor:");
     memset(&mStreamInfo, 0, sizeof(mStreamInfo));
     memset(&mWriteHeader, 0, sizeof(mWriteHeader));
@@ -379,6 +318,7 @@
         case 8:
         case 16:
         case 24:
+        case 32: // generally rare, but is supported in the framework
             break;
 
         default:
@@ -387,31 +327,6 @@
             return ERROR_MALFORMED;
     }
 
-    // configure the appropriate copy function, defaulting to trespass
-    static const struct {
-        unsigned mChannels;
-        unsigned mBitsPerSample;
-        void (*mCopy)(int16_t *dst, const int * src[kMaxChannels],
-                unsigned nSamples, unsigned nChannels);
-    } table[] = {
-        { 1,  8, copyMono8     },
-        { 2,  8, copyStereo8   },
-        { 8,  8, copyMultiCh8  },
-        { 1, 16, copyMono16    },
-        { 2, 16, copyStereo16  },
-        { 8, 16, copyMultiCh16 },
-        { 1, 24, copyMono24    },
-        { 2, 24, copyStereo24  },
-        { 8, 24, copyMultiCh24 },
-    };
-    for (const auto &entry : table) {
-        if (entry.mChannels >= getChannels() &&
-                entry.mBitsPerSample == getBitsPerSample()) {
-            mCopy = entry.mCopy;
-            break;
-        }
-    }
-
     // Now we have all metadata blocks.
     mBufferPos = 0;
     mBufferDataSize = 0;
@@ -420,7 +335,7 @@
 }
 
 status_t FLACDecoder::decodeOneFrame(const uint8_t *inBuffer, size_t inBufferLen,
-        int16_t *outBuffer, size_t *outBufferLen) {
+        void *outBuffer, size_t *outBufferLen, bool outputFloat) {
     ALOGV("decodeOneFrame: input size(%zu)", inBufferLen);
 
     if (!mStreamInfoValid) {
@@ -469,21 +384,33 @@
         return ERROR_MALFORMED;
     }
 
-    size_t bufferSize = blocksize * getChannels() * sizeof(int16_t);
+    const unsigned channels = getChannels();
+    const size_t sampleSize = outputFloat ? sizeof(float) : sizeof(int16_t);
+    const size_t frameSize = channels * sampleSize;
+    size_t bufferSize = blocksize * frameSize;
     if (bufferSize > *outBufferLen) {
         ALOGW("decodeOneFrame: output buffer holds only partial frame %zu:%zu",
                 *outBufferLen, bufferSize);
-        blocksize = *outBufferLen / (getChannels() * sizeof(int16_t));
-        bufferSize = blocksize * getChannels() * sizeof(int16_t);
+        blocksize = *outBufferLen / frameSize;
+        bufferSize = blocksize * frameSize;
     }
 
-    if (mCopy == nullptr) {
-        ALOGE("decodeOneFrame: format is not supported: channels(%d), BitsPerSample(%d)",
-                getChannels(), getBitsPerSample());
-        return ERROR_UNSUPPORTED;
-    }
     // copy PCM from FLAC write buffer to output buffer, with interleaving
-    (*mCopy)(outBuffer, mWriteBuffer, blocksize, getChannels());
+
+    const unsigned bitsPerSample = getBitsPerSample();
+    if (outputFloat) {
+        copyToFloat(reinterpret_cast<float*>(outBuffer),
+                    mWriteBuffer,
+                    blocksize,
+                    channels,
+                    bitsPerSample);
+    } else {
+        copyTo16Signed(reinterpret_cast<short*>(outBuffer),
+                       mWriteBuffer,
+                       blocksize,
+                       channels,
+                       bitsPerSample);
+    }
     *outBufferLen = bufferSize;
     return OK;
 }
diff --git a/media/libstagefright/flac/dec/FLACDecoder.h b/media/libstagefright/flac/dec/FLACDecoder.h
index af419a2..694fccb 100644
--- a/media/libstagefright/flac/dec/FLACDecoder.h
+++ b/media/libstagefright/flac/dec/FLACDecoder.h
@@ -41,7 +41,7 @@
 
     status_t parseMetadata(const uint8_t *inBuffer, size_t inBufferLen);
     status_t decodeOneFrame(const uint8_t *inBuffer, size_t inBufferLen,
-            int16_t *outBuffer, size_t *outBufferLen);
+            void *outBuffer, size_t *outBufferLen, bool outputFloat = false);
     void flush();
     virtual ~FLACDecoder();
 
@@ -89,8 +89,6 @@
     // most recent error reported by libFLAC decoder
     FLAC__StreamDecoderErrorStatus mErrorStatus;
 
-    void (*mCopy)(int16_t *dst, const int *src[kMaxChannels], unsigned nSamples, unsigned nChannels);
-
     status_t init();
 
     // FLAC stream decoder callbacks as C++ instance methods
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 9b2853e..9d46d2d 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -488,7 +488,8 @@
     status_t setupG711Codec(bool encoder, int32_t sampleRate, int32_t numChannels);
 
     status_t setupFlacCodec(
-            bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel);
+            bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel,
+            AudioEncoding encoding);
 
     status_t setupRawAudioFormat(
             OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels,