Various tweaks to wfd

related-to-bug: 7426218

Squashed commit of the following:

commit 1553f1a1c66af998674168f7f7a3be23fcb0c794
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 15:51:27 2012 -0700

    Add LPCM, AVC and AVC HRD descriptors as necessary.

    Change-Id: Ibc836fced0fe37e8a25574c2295e886765b9ea6f

commit 4e74db61d2d31ebe239acbdec8f110f88016a4ea
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 15:50:52 2012 -0700

    added copyright headers to Sender.{cpp,h}

    Change-Id: If615ccb8767e32bd83ed1f0f669acc39a72489f6

commit 7144bf8ae68c5cdb8faa6e219547aabbd750f04e
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 15:50:25 2012 -0700

    Reenable suspension of the RepeaterSource

    Change-Id: I765338fcde89c65e4b69be45a5949eba6bcdcf6f

commit 812164bcfa0699821d7d8eefcc0dff96b2e2cd08
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 14:03:50 2012 -0700

    Add 2 stuffing bytes to the PES headers for audio tracks.

    Change-Id: I8b9c634f6a565ab7fa7ecdb610f7d8557e0b139b

commit a084a741a63015d47c92d99fcd8b980fe615dc7d
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 13:19:38 2012 -0700

    Fix PCM audio packetization in WFD.

    Change-Id: I99a435f9fe6b4397f24d6c22afae5ae2505ffc14

commit c5cb9369585f701f34bce41534940d5f9b59248f
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 13:19:12 2012 -0700

    Support extraction of PCM audio from transport streams.

    Change-Id: I28a0516756ebcb5587325b6588df013ac871ffb9

commit b0a0512300ae037d6b39c2d04952d34b5fc12b2d
Author: Andreas Huber <andih@google.com>
Date:   Tue Oct 30 08:54:13 2012 -0700

    disable suspend of the RepeaterSource

    Change-Id: Ibf42a98185b0567f817ae582a82e6580f95d3d40

commit 4330e8b7668dc92a6d882b5622c0697cf292d04c
Author: Andreas Huber <andih@google.com>
Date:   Mon Oct 29 14:11:25 2012 -0700

    Better handling of datagrams in ANetworkSession

    reduce unnecessary copy overhead.

    Change-Id: I2ed8c767274ba07764f03e8d4913041168e5755f

commit a44e73c322ba3f2c336f7cc4e1d63d3a74faa75d
Author: Andreas Huber <andih@google.com>
Date:   Mon Oct 29 11:14:47 2012 -0700

    Network traffic is now handled on a separate thread.

    Audio and video are queued to ensure proper A/V interleaving.
    Scheduled packet sends according to capture timestamps to reduce
    send-jitter.

    Change-Id: Ibd6357c1e663086cf87bec0a98f8e54dfdfaa0e5
    related-to-bug: 7426218

Change-Id: Ia440129d656c35814abf18df06da50b73d5bb554

4 files changed, 83 insertions, 0 deletions

diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 27c7bf4..9faa6bc 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -131,6 +131,8 @@ private:
     sp<AnotherPacketSource> mSource;
     bool mPayloadStarted;
 
+    uint64_t mPrevPTS;
+
     ElementaryStreamQueue *mQueue;
 
     status_t flush();
@@ -458,6 +460,7 @@ ATSParser::Stream::Stream(
       mPCR_PID(PCR_PID),
       mExpectedContinuityCounter(-1),
       mPayloadStarted(false),
+      mPrevPTS(0),
       mQueue(NULL) {
     switch (mStreamType) {
         case STREAMTYPE_H264:
@@ -486,6 +489,11 @@ ATSParser::Stream::Stream(
                     ElementaryStreamQueue::MPEG4_VIDEO);
             break;
 
+        case STREAMTYPE_PCM_AUDIO:
+            mQueue = new ElementaryStreamQueue(
+                    ElementaryStreamQueue::PCM_AUDIO);
+            break;
+
         default:
             break;
     }
@@ -583,6 +591,7 @@ bool ATSParser::Stream::isAudio() const {
         case STREAMTYPE_MPEG1_AUDIO:
         case STREAMTYPE_MPEG2_AUDIO:
         case STREAMTYPE_MPEG2_AUDIO_ADTS:
+        case STREAMTYPE_PCM_AUDIO:
             return true;
 
         default:
@@ -827,6 +836,14 @@ status_t ATSParser::Stream::flush() {
 void ATSParser::Stream::onPayloadData(
         unsigned PTS_DTS_flags, uint64_t PTS, uint64_t DTS,
         const uint8_t *data, size_t size) {
+#if 0
+    ALOGI("payload streamType 0x%02x, PTS = 0x%016llx, dPTS = %lld",
+          mStreamType,
+          PTS,
+          (int64_t)PTS - mPrevPTS);
+    mPrevPTS = PTS;
+#endif
+
     ALOGV("onPayloadData mStreamType=0x%02x", mStreamType);
 
     int64_t timeUs = 0ll;  // no presentation timestamp available.
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index 5ccbab7..46edc45 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -87,6 +87,7 @@ struct ATSParser : public RefBase {
         STREAMTYPE_MPEG2_AUDIO_ADTS     = 0x0f,
         STREAMTYPE_MPEG4_VIDEO          = 0x10,
         STREAMTYPE_H264                 = 0x1b,
+        STREAMTYPE_PCM_AUDIO            = 0x83,
     };
 
 protected:
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index e58e9bf..82fb637 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -31,6 +31,8 @@
 
 #include "include/avc_utils.h"
 
+#include <netinet/in.h>
+
 namespace android {
 
 ElementaryStreamQueue::ElementaryStreamQueue(Mode mode, uint32_t flags)
@@ -248,6 +250,11 @@ status_t ElementaryStreamQueue::appendData(
                 break;
             }
 
+            case PCM_AUDIO:
+            {
+                break;
+            }
+
             default:
                 TRESPASS();
                 break;
@@ -324,12 +331,68 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() {
             return dequeueAccessUnitMPEGVideo();
         case MPEG4_VIDEO:
             return dequeueAccessUnitMPEG4Video();
+        case PCM_AUDIO:
+            return dequeueAccessUnitPCMAudio();
         default:
             CHECK_EQ((unsigned)mMode, (unsigned)MPEG_AUDIO);
             return dequeueAccessUnitMPEGAudio();
     }
 }
 
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
+    if (mBuffer->size() < 4) {
+        return NULL;
+    }
+
+    ABitReader bits(mBuffer->data(), 4);
+    CHECK_EQ(bits.getBits(8), 0xa0);
+    unsigned numAUs = bits.getBits(8);
+    bits.skipBits(8);
+    unsigned quantization_word_length = bits.getBits(2);
+    unsigned audio_sampling_frequency = bits.getBits(3);
+    unsigned num_channels = bits.getBits(3);
+
+    CHECK_EQ(audio_sampling_frequency, 2);  // 48kHz
+    CHECK_EQ(num_channels, 1u);  // stereo!
+
+    if (mFormat == NULL) {
+        mFormat = new MetaData;
+        mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+        mFormat->setInt32(kKeyChannelCount, 2);
+        mFormat->setInt32(kKeySampleRate, 48000);
+    }
+
+    static const size_t kFramesPerAU = 80;
+    size_t frameSize = 2 /* numChannels */ * sizeof(int16_t);
+
+    size_t payloadSize = numAUs * frameSize * kFramesPerAU;
+
+    if (mBuffer->size() < 4 + payloadSize) {
+        return NULL;
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(payloadSize);
+    memcpy(accessUnit->data(), mBuffer->data() + 4, payloadSize);
+
+    int64_t timeUs = fetchTimestamp(payloadSize + 4);
+    CHECK_GE(timeUs, 0ll);
+    accessUnit->meta()->setInt64("timeUs", timeUs);
+
+    int16_t *ptr = (int16_t *)accessUnit->data();
+    for (size_t i = 0; i < payloadSize / sizeof(int16_t); ++i) {
+        ptr[i] = ntohs(ptr[i]);
+    }
+
+    memmove(
+            mBuffer->data(),
+            mBuffer->data() + 4 + payloadSize,
+            mBuffer->size() - 4 - payloadSize);
+
+    mBuffer->setRange(0, mBuffer->size() - 4 - payloadSize);
+
+    return accessUnit;
+}
+
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAAC() {
     int64_t timeUs;
 
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 72aa2e7..66a8087 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -35,6 +35,7 @@ struct ElementaryStreamQueue {
         MPEG_AUDIO,
         MPEG_VIDEO,
         MPEG4_VIDEO,
+        PCM_AUDIO,
     };
 
     enum Flags {
@@ -69,6 +70,7 @@ private:
     sp<ABuffer> dequeueAccessUnitMPEGAudio();
     sp<ABuffer> dequeueAccessUnitMPEGVideo();
     sp<ABuffer> dequeueAccessUnitMPEG4Video();
+    sp<ABuffer> dequeueAccessUnitPCMAudio();
 
     // consume a logical (compressed) access unit of size "size",
     // returns its timestamp in us (or -1 if no time information).