[DO NOT MERGE] Support TS + AC3 for ATSC standard

Change-Id: I141667f3f54b242bafdf0ab9db86852c56f49ffa

4 files changed, 202 insertions, 0 deletions

diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 175a263..cb57a2f 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -506,6 +506,11 @@ ATSParser::Stream::Stream(
                     ElementaryStreamQueue::PCM_AUDIO);
             break;
 
+        case STREAMTYPE_AC3:
+            mQueue = new ElementaryStreamQueue(
+                    ElementaryStreamQueue::AC3);
+            break;
+
         default:
             break;
     }
@@ -614,6 +619,7 @@ bool ATSParser::Stream::isAudio() const {
         case STREAMTYPE_MPEG2_AUDIO:
         case STREAMTYPE_MPEG2_AUDIO_ADTS:
         case STREAMTYPE_PCM_AUDIO:
+        case STREAMTYPE_AC3:
             return true;
 
         default:
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index a10edc9..d4e30b4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -88,6 +88,10 @@ struct ATSParser : public RefBase {
         STREAMTYPE_MPEG2_AUDIO_ADTS     = 0x0f,
         STREAMTYPE_MPEG4_VIDEO          = 0x10,
         STREAMTYPE_H264                 = 0x1b,
+
+        // From ATSC A/53 Part 3:2009, 6.7.1
+        STREAMTYPE_AC3                  = 0x81,
+
         STREAMTYPE_PCM_AUDIO            = 0x83,
     };
 
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 8f9c9c8..ea79885 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -56,6 +56,122 @@ void ElementaryStreamQueue::clear(bool clearFormat) {
     }
 }
 
+// Parse AC3 header assuming the current ptr is start position of syncframe,
+// update metadata only applicable, and return the payload size
+static unsigned parseAC3SyncFrame(
+        const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 4, 4, 5, 6};
+    static const unsigned samplingRateTable[] = {48000, 44100, 32000};
+    static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256,
+            320, 384, 448, 512, 576, 640};
+
+    static const unsigned frameSizeTable[19][3] = {
+        { 64, 69, 96 },
+        { 80, 87, 120 },
+        { 96, 104, 144 },
+        { 112, 121, 168 },
+        { 128, 139, 192 },
+        { 160, 174, 240 },
+        { 192, 208, 288 },
+        { 224, 243, 336 },
+        { 256, 278, 384 },
+        { 320, 348, 480 },
+        { 384, 417, 576 },
+        { 448, 487, 672 },
+        { 512, 557, 768 },
+        { 640, 696, 960 },
+        { 768, 835, 1152 },
+        { 896, 975, 1344 },
+        { 1024, 1114, 1536 },
+        { 1152, 1253, 1728 },
+        { 1280, 1393, 1920 },
+    };
+
+    ABitReader bits(ptr, size);
+    unsigned syncStartPos = 0;  // in bytes
+    if (bits.numBitsLeft() < 16) {
+        return 0;
+    }
+    if (bits.getBits(16) != 0x0B77) {
+        return 0;
+    }
+
+    if (bits.numBitsLeft() < 16 + 2 + 6 + 5 + 3 + 3) {
+        ALOGV("Not enough bits left for further parsing");
+        return 0;
+    }
+    bits.skipBits(16);  // crc1
+
+    unsigned fscod = bits.getBits(2);
+    if (fscod == 3) {
+        ALOGW("Incorrect fscod in AC3 header");
+        return 0;
+    }
+
+    unsigned frmsizecod = bits.getBits(6);
+    if (frmsizecod > 37) {
+        ALOGW("Incorrect frmsizecod in AC3 header");
+        return 0;
+    }
+
+    unsigned bsid = bits.getBits(5);
+    if (bsid > 8) {
+        ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?");
+        return 0;
+    }
+
+    unsigned bsmod = bits.getBits(3);
+    unsigned acmod = bits.getBits(3);
+    unsigned cmixlev = 0;
+    unsigned surmixlev = 0;
+    unsigned dsurmod = 0;
+
+    if ((acmod & 1) > 0 && acmod != 1) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        cmixlev = bits.getBits(2);
+    }
+    if ((acmod & 4) > 0) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        surmixlev = bits.getBits(2);
+    }
+    if (acmod == 2) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        dsurmod = bits.getBits(2);
+    }
+
+    if (bits.numBitsLeft() < 1) {
+        return 0;
+    }
+    unsigned lfeon = bits.getBits(1);
+
+    unsigned samplingRate = samplingRateTable[fscod];
+    unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod];
+    if (fscod == 1) {
+        payloadSize += frmsizecod & 1;
+    }
+    payloadSize <<= 1;  // convert from 16-bit words to bytes
+
+    unsigned channelCount = channelCountTable[acmod] + lfeon;
+
+    if (metaData != NULL) {
+        (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+        (*metaData)->setInt32(kKeyChannelCount, channelCount);
+        (*metaData)->setInt32(kKeySampleRate, samplingRate);
+    }
+
+    return payloadSize;
+}
+
+static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) {
+    return parseAC3SyncFrame(ptr, size, NULL) > 0;
+}
+
 static bool IsSeeminglyValidADTSHeader(const uint8_t *ptr, size_t size) {
     if (size < 3) {
         // Not enough data to verify header.
@@ -224,6 +340,33 @@ status_t ElementaryStreamQueue::appendData(
                 break;
             }
 
+            case AC3:
+            {
+                uint8_t *ptr = (uint8_t *)data;
+
+                ssize_t startOffset = -1;
+                for (size_t i = 0; i < size; ++i) {
+                    if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) {
+                        startOffset = i;
+                        break;
+                    }
+                }
+
+                if (startOffset < 0) {
+                    return ERROR_MALFORMED;
+                }
+
+                if (startOffset > 0) {
+                    ALOGI("found something resembling an AC3 syncword at "
+                          "offset %d",
+                          startOffset);
+                }
+
+                data = &ptr[startOffset];
+                size -= startOffset;
+                break;
+            }
+
             case MPEG_AUDIO:
             {
                 uint8_t *ptr = (uint8_t *)data;
@@ -328,6 +471,8 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() {
             return dequeueAccessUnitH264();
         case AAC:
             return dequeueAccessUnitAAC();
+        case AC3:
+            return dequeueAccessUnitAC3();
         case MPEG_VIDEO:
             return dequeueAccessUnitMPEGVideo();
         case MPEG4_VIDEO:
@@ -340,6 +485,51 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() {
     }
 }
 
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() {
+    unsigned syncStartPos = 0;  // in bytes
+    unsigned payloadSize = 0;
+    sp<MetaData> format = new MetaData;
+    while (true) {
+        if (syncStartPos + 2 >= mBuffer->size()) {
+            return NULL;
+        }
+
+        payloadSize = parseAC3SyncFrame(
+                mBuffer->data() + syncStartPos,
+                mBuffer->size() - syncStartPos,
+                &format);
+        if (payloadSize > 0) {
+            break;
+        }
+        ++syncStartPos;
+    }
+
+    if (mBuffer->size() < syncStartPos + payloadSize) {
+        ALOGV("Not enough buffer size for AC3");
+        return NULL;
+    }
+
+    if (mFormat == NULL) {
+        mFormat = format;
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
+    memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
+
+    int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
+    CHECK_GE(timeUs, 0ll);
+    accessUnit->meta()->setInt64("timeUs", timeUs);
+
+    memmove(
+            mBuffer->data(),
+            mBuffer->data() + syncStartPos + payloadSize,
+            mBuffer->size() - syncStartPos - payloadSize);
+
+    mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize);
+
+    return accessUnit;
+}
+
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
     if (mBuffer->size() < 4) {
         return NULL;
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 66a8087..a2cca77 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -32,6 +32,7 @@ struct ElementaryStreamQueue {
     enum Mode {
         H264,
         AAC,
+        AC3,
         MPEG_AUDIO,
         MPEG_VIDEO,
         MPEG4_VIDEO,
@@ -67,6 +68,7 @@ private:
 
     sp<ABuffer> dequeueAccessUnitH264();
     sp<ABuffer> dequeueAccessUnitAAC();
+    sp<ABuffer> dequeueAccessUnitAC3();
     sp<ABuffer> dequeueAccessUnitMPEGAudio();
     sp<ABuffer> dequeueAccessUnitMPEGVideo();
     sp<ABuffer> dequeueAccessUnitMPEG4Video();