Ticket #1104: mythtv_ac3.14.patch

libs/libmyth/libmyth.pro

@@ -34 +34 @@
 SOURCES += virtualkeyboard.cpp mythobservable.cpp
 
 INCLUDEPATH += ../libmythsamplerate ../libmythsoundtouch ../..
+INCLUDEPATH += ../libavutil ..
 DEPENDPATH += ../libmythsamplerate ../libmythsoundtouch
+DEPENDPATH += ../libavutil ../libavcodec
 
 
 LIBS += -L../libmythsamplerate -lmythsamplerate-$${LIBVERSION}
 LIBS += -L../libmythsoundtouch -lmythsoundtouch-$${LIBVERSION}
+LIBS += -L../libavcodec -lmythavcodec-$${LIBVERSION}
 
 isEmpty(QMAKE_EXTENSION_SHLIB) {
   QMAKE_EXTENSION_SHLIB=so

libs/libmyth/audiooutput.h

@@ -27 +27 @@
 
     // reconfigure sound out for new params
     virtual void Reconfigure(int audio_bits, 
-                             int audio_channels, int audio_samplerate) = 0;
+                             int audio_channels, 
+                             int audio_samplerate,
+                             void* audio_codec = NULL
+                             ) = 0;
     
     virtual void SetStretchFactor(float factor);
 
@@ -65 +68 @@
  protected:
     void Error(QString msg) 
      { lastError = msg; VERBOSE(VB_IMPORTANT, lastError); };
+    void ClearError()
+     { lastError = QString::null; };
 
  private:
     QString lastError;

libs/libmyth/audiooutputdx.h

@@ -21 +21 @@
 
     virtual void Reset(void);
     virtual void Reconfigure(int audio_bits, 
-                         int audio_channels, int audio_samplerate);
+                         int audio_channels, 
+                         int audio_samplerate
+                         AudioCodecMode aom = AUDIOCODECMODE_NORMAL);
     virtual void SetBlocking(bool blocking);
 
     virtual bool AddSamples(char *buffer, int samples, long long timecode);

libs/libmyth/audiooutputdx.cpp

@@ -119 +119 @@
 }
 
 void AudioOutputDX::Reconfigure(int audio_bits, 
-                                  int audio_channels, int audio_samplerate)
+                                int audio_channels, 
+                                int audio_samplerate,
+                                AudioCodecMode laom
+                                )
 {
     if (dsbuffer)
         DestroyDSBuffer();

libs/libmyth/audiooutputbase.h

@@ -12 +12 @@
 #include "samplerate.h"
 #include "SoundTouch.h"
 
-#define AUDBUFSIZE 768000
+struct AVCodecContext;
+class DigitalEncoder;
 
+//#define AUDBUFSIZE 768000
+//divisible by 12,10,8,6,4,2 and around 1024000
+//#define AUDBUFSIZE 1024080
+#define AUDBUFSIZE 1536000
+
 class AudioOutputBase : public AudioOutput
 {
  public:
@@ -24 +30 @@
 
     // reconfigure sound out for new params
     virtual void Reconfigure(int audio_bits, 
-                             int audio_channels, int audio_samplerate);
+                             int audio_channels, 
+                             int audio_samplerate,
+                             void* audio_codec = NULL);
     
     // do AddSamples calls block?
     virtual void SetBlocking(bool blocking);
@@ -52 +60 @@
     // Send output events showing current progress
     virtual void Status(void);
 
-    QString GetError() { return lastError; };
+    //QString GetError() { return lastError; };
 
     virtual void SetSourceBitrate(int rate);
 
@@ -108 +116 @@
 
     float audio_stretchfactor;
     AudioOutputSource source;
+    AVCodecContext *audio_codec;
 
     bool killaudio;
 
@@ -116 +125 @@
     bool buffer_output_data_for_use; //  used by AudioOutputNULL
     
  private:
-    QString lastError;
+    //QString lastError;
 
     // resampler
     bool need_resampler;
@@ -127 +136 @@
 
     // timestretch
     soundtouch::SoundTouch * pSoundStretch;
+    DigitalEncoder * encoder;
 
     bool blocking; // do AddSamples calls block?
 
@@ -142 +152 @@
 
     pthread_mutex_t avsync_lock; /* must hold avsync_lock to read or write
                                     'audiotime' and 'audiotime_updated' */
-    int audiotime; // timecode of audio leaving the soundcard (same units as
+    long long audiotime; // timecode of audio leaving the soundcard (same units as
                    //                                          timecodes) ...
     struct timeval audiotime_updated; // ... which was last updated at this time
 
     /* Audio circular buffer */
     unsigned char audiobuffer[AUDBUFSIZE];  /* buffer */
     int raud, waud;     /* read and write positions */
-    int audbuf_timecode;    /* timecode of audio most recently placed into
+    long long audbuf_timecode;    /* timecode of audio most recently placed into
                    buffer */
 
     int numlowbuffer;

libs/libmyth/audiooutputbase.cpp

@@ -12 +12 @@
 #include <sys/time.h>
 #include <unistd.h>
 
+extern "C" {
+#include "libavcodec/avcodec.h"
+#include "libavcodec/liba52/a52.h"
+}
 
+#if QT_VERSION < 0x030200
+#define LONGLONGCONVERT (long)
+#else
+#define LONGLONGCONVERT
+#endif
+
+#define LOC QString("DEnc: ");
+#define MAX_AC3_FRAME_SIZE 6144
+class DigitalEncoder
+{
+public:
+    DigitalEncoder();
+    ~DigitalEncoder();
+    void Dispose();
+    bool DigitalEncoder::Init(CodecID codec_id, int bitrate, int samplerate, int channels);
+    size_t Encode(short * buff);
+
+    // if needed
+    char * GetFrameBuffer() 
+    { 
+        if (!frame_buffer && av_context)
+        {
+            frame_buffer = new char [one_frame_bytes];
+        }
+        return frame_buffer; 
+    }    
+    size_t FrameSize() const { return one_frame_bytes; }
+    char * GetOutBuff() const { return outbuf; }
+
+    size_t audio_bytes_per_sample;
+private:
+    AVCodecContext *av_context;
+    char * outbuf;
+    char * frame_buffer;
+    int outbuf_size;
+    size_t one_frame_bytes;
+};
+
+DigitalEncoder::DigitalEncoder()
+{
+    av_context = NULL;
+    outbuf = NULL;
+    outbuf_size = 0;
+    one_frame_bytes = 0;
+    frame_buffer = NULL;
+}
+
+DigitalEncoder::~DigitalEncoder()
+{
+    Dispose();
+}
+
+void DigitalEncoder::Dispose()
+{
+    if (av_context)
+    {
+        avcodec_close(av_context);
+        av_free(av_context);
+        av_context = NULL;
+    }
+    if (outbuf)
+    {
+        delete [] outbuf;
+        outbuf = NULL;
+        outbuf_size = 0;
+    }
+    if (frame_buffer)
+    {
+        delete [] frame_buffer;
+        frame_buffer = NULL;
+        one_frame_bytes = 0;
+    }
+}
+
+//CODEC_ID_AC3
+bool DigitalEncoder::Init(CodecID codec_id, int bitrate, int samplerate, int channels)
+{
+    AVCodec * codec;
+    int ret;
+
+    VERBOSE(VB_AUDIO, QString("DigitalEncoder::Init codecid=%1, br=%2, sr=%3, ch=%4")
+            .arg(codec_id_string(codec_id))
+            .arg(bitrate)
+            .arg(samplerate)
+            .arg(channels));
+    //codec = avcodec_find_encoder(codec_id);
+    // always AC3 as there is no DTS encoder at the moment 2005/1/9
+    codec = avcodec_find_encoder(CODEC_ID_AC3);
+    if (!codec)
+    {
+        VERBOSE(VB_IMPORTANT,"Error: could not find codec");
+        return false;
+    }
+    av_context = avcodec_alloc_context();
+    av_context->bit_rate = bitrate;
+    av_context->sample_rate = samplerate;
+    av_context->channels = channels;
+    // open it */
+    if ((ret = avcodec_open(av_context, codec)) < 0) 
+    {
+        VERBOSE(VB_IMPORTANT,"Error: could not open codec, invalid bitrate or samplerate");
+        Dispose();
+        return false;
+    }
+
+    size_t bytes_per_frame = av_context->channels * sizeof(short);
+    audio_bytes_per_sample = bytes_per_frame;
+    one_frame_bytes = bytes_per_frame * av_context->frame_size;
+
+    outbuf_size = 16384;    // ok for AC3 but DTS?
+    outbuf = new char [outbuf_size];
+    VERBOSE(VB_AUDIO, QString("DigitalEncoder::Init fs=%1, bpf=%2 ofb=%3")
+            .arg(av_context->frame_size)
+            .arg(bytes_per_frame)
+            .arg(one_frame_bytes)
+           );
+
+    return true;
+}
+
+static int DTS_SAMPLEFREQS[16] =
+{
+    0,      8000,   16000,  32000,  64000,  128000, 11025,  22050,
+    44100,  88200,  176400, 12000,  24000,  48000,  96000,  192000
+};
+
+static int DTS_BITRATES[30] =
+{
+    32000,    56000,    64000,    96000,    112000,   128000,
+    192000,   224000,   256000,   320000,   384000,   448000,
+    512000,   576000,   640000,   768000,   896000,   1024000,
+    1152000,  1280000,  1344000,  1408000,  1411200,  1472000,
+    1536000,  1920000,  2048000,  3072000,  3840000,  4096000
+};
+
+static int dts_decode_header(uint8_t *indata_ptr, int *rate,
+                             int *nblks, int *sfreq)
+{
+    uint id = ((indata_ptr[0] << 24) | (indata_ptr[1] << 16) |
+               (indata_ptr[2] << 8)  | (indata_ptr[3]));
+
+    if (id != 0x7ffe8001)
+        return -1;
+
+    int ftype = indata_ptr[4] >> 7;
+
+    int surp = (indata_ptr[4] >> 2) & 0x1f;
+    surp = (surp + 1) % 32;
+
+    *nblks = (indata_ptr[4] & 0x01) << 6 | (indata_ptr[5] >> 2);
+    ++*nblks;
+
+    int fsize = (indata_ptr[5] & 0x03) << 12 |
+                (indata_ptr[6]         << 4) | (indata_ptr[7] >> 4);
+    ++fsize;
+
+    *sfreq = (indata_ptr[8] >> 2) & 0x0f;
+    *rate = (indata_ptr[8] & 0x03) << 3 | ((indata_ptr[9] >> 5) & 0x07);
+
+    if (ftype != 1)
+    {
+        VERBOSE(VB_IMPORTANT, LOC +
+                QString("DTS: Termination frames not handled (ftype %1)")
+                .arg(ftype));
+        return -1;
+    }
+
+    if (*sfreq != 13)
+    {
+        VERBOSE(VB_IMPORTANT, LOC +
+                QString("DTS: Only 48kHz supported (sfreq %1)").arg(*sfreq));
+        return -1;
+    }
+
+    if ((fsize > 8192) || (fsize < 96))
+    {
+        VERBOSE(VB_IMPORTANT, LOC +
+                QString("DTS: fsize: %1 invalid").arg(fsize));
+        return -1;
+    }
+
+    if (*nblks != 8 && *nblks != 16 && *nblks != 32 &&
+        *nblks != 64 && *nblks != 128 && ftype == 1)
+    {
+        VERBOSE(VB_IMPORTANT, LOC +
+                QString("DTS: nblks %1 not valid for normal frame")
+                .arg(*nblks));
+        return -1;
+    }
+
+    return fsize;
+}
+
+static int dts_syncinfo(uint8_t *indata_ptr, int * /*flags*/,
+                        int *sample_rate, int *bit_rate)
+{
+    int nblks;
+    int rate;
+    int sfreq;
+
+    int fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
+    if (fsize >= 0)
+    {
+        if (rate >= 0 && rate <= 29)
+            *bit_rate = DTS_BITRATES[rate];
+        else
+            *bit_rate = 0;
+        if (sfreq >= 1 && sfreq <= 15)
+            *sample_rate = DTS_SAMPLEFREQS[sfreq];
+        else
+            *sample_rate = 0;
+    }
+    return fsize;
+}
+
+static int encode_frame(
+        bool dts, 
+        unsigned char *data,
+        size_t &len)
+{
+    size_t enc_len;
+    int flags, sample_rate, bit_rate;
+
+    // we don't do any length/crc validation of the AC3 frame here; presumably
+    // the receiver will have enough sense to do that.  if someone has a
+    // receiver that doesn't, here would be a good place to put in a call
+    // to a52_crc16_block(samples+2, data_size-2) - but what do we do if the
+    // packet is bad?  we'd need to send something that the receiver would
+    // ignore, and if so, may as well just assume that it will ignore
+    // anything with a bad CRC...
+
+    uint nr_samples = 0, block_len;
+    if (dts)
+    {
+        enc_len = dts_syncinfo(data+8, &flags, &sample_rate, &bit_rate);
+        int rate, sfreq, nblks;
+        dts_decode_header(data+8, &rate, &nblks, &sfreq);
+        nr_samples = nblks * 32;
+        block_len = nr_samples * 2 * 2;
+    }
+    else
+    {
+        enc_len = a52_syncinfo(data+8, &flags, &sample_rate, &bit_rate);
+        block_len = MAX_AC3_FRAME_SIZE;
+    }
+
+    if (enc_len == 0 || enc_len > len)
+    {
+        int l = len;
+        len = 0;
+        return l;
+    }
+
+    enc_len = min((uint)enc_len, block_len - 8);
+
+    //uint32_t x = *(uint32_t*)(data+8);
+    // in place swab
+    swab(data+8, data+8, enc_len);
+    //VERBOSE(VB_AUDIO|VB_TIMESTAMP, 
+    //        QString("DigitalEncoder::Encode swab test %1 %2")
+    //        .arg(x,0,16).arg(*(uint32_t*)(data+8),0,16));
+
+    // the following values come from libmpcodecs/ad_hwac3.c in mplayer.
+    // they form a valid IEC958 AC3 header.
+    data[0] = 0x72;
+    data[1] = 0xF8;
+    data[2] = 0x1F;
+    data[3] = 0x4E;
+    data[4] = 0x01;
+    if (dts)
+    {
+        switch(nr_samples)
+        {
+            case 512:
+                data[4] = 0x0B;      /* DTS-1 (512-sample bursts) */
+                break;
+
+            case 1024:
+                data[4] = 0x0C;      /* DTS-2 (1024-sample bursts) */
+                break;
+
+            case 2048:
+                data[4] = 0x0D;      /* DTS-3 (2048-sample bursts) */
+                break;
+
+            default:
+                VERBOSE(VB_IMPORTANT, LOC +
+                        QString("DTS: %1-sample bursts not supported")
+                        .arg(nr_samples));
+                data[4] = 0x00;
+                break;
+        }
+    }
+    data[5] = 0x00;
+    data[6] = (enc_len << 3) & 0xFF;
+    data[7] = (enc_len >> 5) & 0xFF;
+    memset(data + 8 + enc_len, 0, block_len - 8 - enc_len);
+    len = block_len;
+
+    return enc_len;
+}
+
+// must have exactly 1 frames worth of data
+size_t DigitalEncoder::Encode(short * buff)
+{
+    int encsize = 0;
+    size_t outsize = 0;
+ 
+    // put data in the correct spot for encode frame
+    outsize = avcodec_encode_audio(
+                av_context, 
+                ((uchar*)outbuf)+8, 
+                outbuf_size-8, 
+                buff);
+    size_t tmpsize = outsize;
+
+    outsize = MAX_AC3_FRAME_SIZE;
+    encsize = encode_frame(
+            //av_context->codec_id==CODEC_ID_DTS,
+            false,
+            (unsigned char*)outbuf, outsize);
+    VERBOSE(VB_AUDIO|VB_TIMESTAMP, 
+            QString("DigitalEncoder::Encode len1=%1 len2=%2 finallen=%3")
+                .arg(tmpsize)
+                .arg(encsize)
+                .arg(outsize)
+           );
+
+    return outsize;
+}
+#undef LOC
+#define LOC QString("AO: ")
+
 AudioOutputBase::AudioOutputBase(QString audiodevice, int, 
                                  int, int,
                                  AudioOutputSource source, bool set_initial_vol)
@@ -29 +366 @@
     current_seconds = -1;
     source_bitrate = -1;
     audio_stretchfactor = 1.0;
+    audio_codec = NULL;
     pSoundStretch = NULL;
+    encoder = NULL;
     blocking = false;
     this->source = source;
     this->set_initial_vol = set_initial_vol;
@@ -78 +417 @@
             VERBOSE(VB_GENERAL, QString("Using time stretch %1")
                                         .arg(audio_stretchfactor));
             pSoundStretch = new soundtouch::SoundTouch();
-            pSoundStretch->setSampleRate(audio_samplerate);
-            pSoundStretch->setChannels(audio_channels);
+            if (audio_codec)
+            {
+                if (!encoder)
+                {
+                    VERBOSE(VB_AUDIO, LOC + QString("Creating Encoder for codec %1 origfs %2").arg(audio_codec->codec_id).arg(audio_codec->frame_size));
+                    encoder = new DigitalEncoder();
+                    if (!encoder->Init(audio_codec->codec_id,
+                                audio_codec->bit_rate,
+                                audio_codec->sample_rate,
+                                audio_codec->channels
+                                ))
+                    {
+                        // eeks
+                        delete encoder;
+                        encoder = NULL;
+                        VERBOSE(VB_AUDIO, LOC + QString("Failed to Create Encoder"));
+                    }
+                }
+            }
+            if (encoder)
+            {
+                pSoundStretch->setSampleRate(audio_codec->sample_rate);
+                pSoundStretch->setChannels(audio_codec->channels);
+            }
+            else
+            {
+                pSoundStretch->setSampleRate(audio_samplerate);
+                pSoundStretch->setChannels(audio_channels);
+            }
 
             pSoundStretch->setTempo(audio_stretchfactor);
             pSoundStretch->setSetting(SETTING_SEQUENCE_MS, 35);
@@ -102 +468 @@
 }
 
 void AudioOutputBase::Reconfigure(int laudio_bits, int laudio_channels, 
-                                 int laudio_samplerate)
+                                 int laudio_samplerate, void* laudio_codec)
 {
+    int codec_id = CODEC_ID_NONE;
+    int lcodec_id = CODEC_ID_NONE;
+    if (laudio_codec)
+    {
+        lcodec_id = ((AVCodecContext*)laudio_codec)->codec_id;
+        laudio_bits = 16;
+        laudio_channels = 2;
+        laudio_samplerate = 48000;
+    }
+    if (audio_codec)
+        codec_id = audio_codec->codec_id;
+    ClearError();
     if (laudio_bits == audio_bits && laudio_channels == audio_channels &&
-        laudio_samplerate == audio_samplerate && !need_resampler)
+        laudio_samplerate == audio_samplerate && !need_resampler &&
+        lcodec_id == codec_id)
         return;
 
     KillAudio();
@@ -120 +499 @@
     audio_channels = laudio_channels;
     audio_bits = laudio_bits;
     audio_samplerate = laudio_samplerate;
+    audio_codec = (AVCodecContext*)laudio_codec;
     if (audio_bits != 8 && audio_bits != 16)
     {
         Error("AudioOutput only supports 8 or 16bit audio.");
@@ -135 +515 @@
     
     numlowbuffer = 0;
 
-    VERBOSE(VB_GENERAL, QString("Opening audio device '%1'.")
-            .arg(audiodevice));
+    VERBOSE(VB_GENERAL, QString("Opening audio device '%1'. ch %2 sr %3")
+            .arg(audiodevice).arg(audio_channels).arg(audio_samplerate));
     
     // Actually do the device specific open call
     if (!OpenDevice()) {
         pthread_mutex_unlock(&avsync_lock);
         pthread_mutex_unlock(&audio_buflock);
+        if (GetError().isEmpty())
+            Error("Aborting reconfigure");
         VERBOSE(VB_AUDIO, "Aborting reconfigure");
         return;
     }
@@ -185 +567 @@
     }
 
     VERBOSE(VB_AUDIO, QString("Audio Stretch Factor: %1").arg(audio_stretchfactor));
+    VERBOSE(VB_AUDIO, QString("Audio Codec Used: %1")
+            .arg(audio_codec?codec_id_string(audio_codec->codec_id):"not set"));
 
     SetStretchFactorLocked(audio_stretchfactor);
     if (pSoundStretch)
     {
-        pSoundStretch->setSampleRate(audio_samplerate);
-        pSoundStretch->setChannels(audio_channels);
+        // if its passthru then we need to reencode
+        if (audio_codec)
+        {
+            if (!encoder)
+            {
+                VERBOSE(VB_AUDIO, LOC + QString("Creating Encoder for codec %1").arg(audio_codec->codec_id));
+                encoder = new DigitalEncoder();
+                if (!encoder->Init(audio_codec->codec_id,
+                            audio_codec->bit_rate,
+                            audio_codec->sample_rate,
+                            audio_codec->channels
+                            ))
+                {
+                    // eeks
+                    delete encoder;
+                    encoder = NULL;
+                    VERBOSE(VB_AUDIO, LOC + QString("Failed to Create Encoder"));
+                }
+            }
+        }
+        if (encoder)
+        {
+            pSoundStretch->setSampleRate(audio_codec->sample_rate);
+            pSoundStretch->setChannels(audio_codec->channels);
+        }
+        else
+        {
+            pSoundStretch->setSampleRate(audio_samplerate);
+            pSoundStretch->setChannels(audio_channels);
+        }
     }
 
     // Setup visualisations, zero the visualisations buffers
@@ -237 +649 @@
         pSoundStretch = NULL;
     }
 
+    if (encoder)
+    {
+        delete encoder;
+        encoder = NULL;
+    }
+
     CloseDevice();
 
     killAudioLock.unlock();
@@ -330 +748 @@
        The reason is that computing 'audiotime' requires acquiring the audio 
        lock, which the video thread should not do. So, we call 'SetAudioTime()'
        from the audio thread, and then call this from the video thread. */
-    int ret;
+    long long ret;
     struct timeval now;
 
     if (audiotime == 0)
@@ -342 +760 @@
 
     ret = (now.tv_sec - audiotime_updated.tv_sec) * 1000;
     ret += (now.tv_usec - audiotime_updated.tv_usec) / 1000;
-    ret = (int)(ret * audio_stretchfactor);
+    ret = (long long)(ret * audio_stretchfactor);
 
+#if 1
+    VERBOSE(VB_AUDIO|VB_TIMESTAMP, 
+            QString("GetAudiotime now=%1.%2, set=%3.%4, ret=%5, audt=%6 sf=%7")
+            .arg(now.tv_sec).arg(now.tv_usec)
+            .arg(audiotime_updated.tv_sec).arg(audiotime_updated.tv_usec)
+            .arg(ret)
+            .arg(audiotime)
+            .arg(audio_stretchfactor)
+           );
+#endif
+
     ret += audiotime;
 
     pthread_mutex_unlock(&avsync_lock);
-    return ret;
+    return (int)ret;
 }
 
 void AudioOutputBase::SetAudiotime(void)
@@ -384 +813 @@
     // include algorithmic latencies
     if (pSoundStretch)
     {
+        // if encoder is active, then use its idea of audiobytes
+        //size_t abps = encoder?encoder->audio_bytes_per_sample:audio_bytes_per_sample;
+
+        // add the effect of any unused but processed samples, AC3 reencode does this
+        totalbuffer += (int)(pSoundStretch->numSamples() * audio_bytes_per_sample);
         // add the effect of unprocessed samples in time stretch algo
         totalbuffer += (int)((pSoundStretch->numUnprocessedSamples() *
                               audio_bytes_per_sample) / audio_stretchfactor);
     }
-               
+
     audiotime = audbuf_timecode - (int)(totalbuffer * 100000.0 /
                                    (audio_bytes_per_sample * effdspstretched));
  
     gettimeofday(&audiotime_updated, NULL);
+#if 1
+    VERBOSE(VB_AUDIO|VB_TIMESTAMP, 
+            QString("SetAudiotime set=%1.%2, audt=%3 atc=%4 tb=%5 sb=%6 eds=%7 abps=%8 sf=%9")
+            .arg(audiotime_updated.tv_sec).arg(audiotime_updated.tv_usec)
+            .arg(audiotime)
+            .arg(audbuf_timecode)
+            .arg(totalbuffer)
+            .arg(soundcard_buffer)
+            .arg(effdspstretched)
+            .arg(audio_bytes_per_sample)
+            .arg(audio_stretchfactor)
+           );
+#endif
 
     pthread_mutex_unlock(&avsync_lock);
     pthread_mutex_unlock(&audio_buflock);
@@ -452 +899 @@
     // NOTE: This function is not threadsafe
 
     int afree = audiofree(true);
-    int len = samples * audio_bytes_per_sample;
+    int len = samples * (encoder?encoder->audio_bytes_per_sample:audio_bytes_per_sample);
 
     // Check we have enough space to write the data
     if (need_resampler && src_ctx)
         len = (int)ceilf(float(len) * src_data.src_ratio);
     if ((len > afree) && !blocking)
+    {
+        VERBOSE(VB_AUDIO|VB_TIMESTAMP, QString("AddSamples FAILED bytes=%1, used=%2, free=%3, timecode=%4")
+            .arg(len)
+            .arg(AUDBUFSIZE-afree).arg(afree).arg(LONGLONGCONVERT timecode));
         return false; // would overflow
+    }
 
     // resample input if necessary
     if (need_resampler && src_ctx) 
@@ -492 +944 @@
 
 int AudioOutputBase::WaitForFreeSpace(int samples)
 {
-    int len = samples * audio_bytes_per_sample;
+    int abps = encoder?encoder->audio_bytes_per_sample:audio_bytes_per_sample;
+    int len = samples * abps;
     int afree = audiofree(false);
 
     while (len > afree)
     {
         if (blocking)
         {
-            VERBOSE(VB_AUDIO, "Waiting for free space");
+            VERBOSE(VB_AUDIO|VB_TIMESTAMP, "Waiting for free space");
             // wait for more space
             pthread_cond_wait(&audio_bufsig, &audio_buflock);
             afree = audiofree(false);
         }
         else
         {
-            VERBOSE(VB_IMPORTANT, "Audio buffer overflow, audio data lost!");
-            samples = afree / audio_bytes_per_sample;
-            len = samples * audio_bytes_per_sample;
+            VERBOSE(VB_IMPORTANT, 
+                    QString("Audio buffer overflow, %1 audio samples lost!")
+                        .arg(samples-afree / abps));
+            samples = afree / abps;
+            len = samples * abps;
             if (src_ctx) 
             {
                 int error = src_reset(src_ctx);
@@ -533 +988 @@
     
     int afree = audiofree(false);
 
-    VERBOSE(VB_AUDIO, QString("_AddSamples bytes=%1, used=%2, free=%3, timecode=%4")
-            .arg(samples * audio_bytes_per_sample)
-            .arg(AUDBUFSIZE-afree).arg(afree).arg((long)timecode));
+    int abps = encoder?encoder->audio_bytes_per_sample:audio_bytes_per_sample;
+    VERBOSE(VB_AUDIO|VB_TIMESTAMP, QString("_AddSamples samples=%1 bytes=%2, used=%3, free=%4, timecode=%5")
+            .arg(samples)
+            .arg(samples * abps)
+            .arg(AUDBUFSIZE-afree).arg(afree).arg(LONGLONGCONVERT timecode));
     
     len = WaitForFreeSpace(samples);
 
@@ -572 +1029 @@
 
     if (pSoundStretch)
     {
+
         // does not change the timecode, only the number of samples
         // back to orig pos
         org_waud = waud;
         int bdiff = AUDBUFSIZE - org_waud;
-        int nSamplesToEnd = bdiff/audio_bytes_per_sample;
+        int nSamplesToEnd = bdiff/abps;
         if (bdiff < len)
         {
             pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)(audiobuffer + 
                                       org_waud), nSamplesToEnd);
             pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer,
-                                      (len - bdiff) / audio_bytes_per_sample);
+                                      (len - bdiff) / abps);
         }
         else
         {
             pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)(audiobuffer + 
-                                      org_waud), len / audio_bytes_per_sample);
+                                      org_waud), len / abps);
         }
 
-        int newLen = 0;
-        int nSamples;
-        len = WaitForFreeSpace(pSoundStretch->numSamples() * 
-                               audio_bytes_per_sample);
-        do 
+        if (encoder)
         {
-            int samplesToGet = len/audio_bytes_per_sample;
-            if (samplesToGet > nSamplesToEnd)
+            // pull out a packet's worth and reencode it until we dont have enough
+            // for any more packets
+            soundtouch::SAMPLETYPE* temp_buff = 
+                (soundtouch::SAMPLETYPE*)encoder->GetFrameBuffer();
+            size_t frameSize = encoder->FrameSize()/abps;
+            VERBOSE(VB_AUDIO|VB_TIMESTAMP,
+                    QString("_AddSamples Enc sfs=%1 bfs=%2 sss=%3")
+                    .arg(frameSize)
+                    .arg(encoder->FrameSize())
+                    .arg(pSoundStretch->numSamples())
+                   );
+            // process the same number of samples as it creates a full encoded buffer
+            // just like before
+            while (pSoundStretch->numSamples() >= frameSize)
             {
-                samplesToGet = nSamplesToEnd;    
+                int got = pSoundStretch->receiveSamples(temp_buff, frameSize);
+                int amount = encoder->Encode(temp_buff);
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP, 
+                        QString("_AddSamples Enc bytes=%1 got=%2 left=%3")
+                        .arg(amount)
+                        .arg(got)
+                        .arg(pSoundStretch->numSamples())
+                       );
+                if (amount == 0)
+                    continue;
+                //len = WaitForFreeSpace(amount);
+                char * ob = encoder->GetOutBuff();
+                if (amount >= bdiff)
+                {
+                    memcpy(audiobuffer + org_waud, ob, bdiff);
+                    ob += bdiff;
+                    amount -= bdiff;
+                    org_waud = 0;
+                }
+                if (amount > 0)
+                    memcpy(audiobuffer + org_waud, ob, amount);
+                bdiff = AUDBUFSIZE - amount;
+                org_waud += amount;
             }
-
-            nSamples = pSoundStretch->receiveSamples((soundtouch::SAMPLETYPE*)
-                                      (audiobuffer + org_waud), samplesToGet);
-            if (nSamples == nSamplesToEnd)
+        }
+        else
+        {
+            int newLen = 0;
+            int nSamples;
+            len = WaitForFreeSpace(pSoundStretch->numSamples() * 
+                                   audio_bytes_per_sample);
+            do 
             {
-                org_waud = 0;
-                nSamplesToEnd = AUDBUFSIZE/audio_bytes_per_sample;
-            }
-            else
-            {
-                org_waud += nSamples * audio_bytes_per_sample;
-                nSamplesToEnd -= nSamples;
-            }
+                int samplesToGet = len/audio_bytes_per_sample;
+                if (samplesToGet > nSamplesToEnd)
+                {
+                    samplesToGet = nSamplesToEnd;    
+                }
 
-            newLen += nSamples * audio_bytes_per_sample;
-            len -= nSamples * audio_bytes_per_sample;
-        } while (nSamples > 0);
+                nSamples = pSoundStretch->receiveSamples((soundtouch::SAMPLETYPE*)
+                                          (audiobuffer + org_waud), samplesToGet);
+                if (nSamples == nSamplesToEnd)
+                {
+                    org_waud = 0;
+                    nSamplesToEnd = AUDBUFSIZE/audio_bytes_per_sample;
+                }
+                else
+                {
+                    org_waud += nSamples * audio_bytes_per_sample;
+                    nSamplesToEnd -= nSamples;
+                }
+
+                newLen += nSamples * audio_bytes_per_sample;
+                len -= nSamples * audio_bytes_per_sample;
+            } while (nSamples > 0);
+        }
     }
 
     waud = org_waud;
@@ -687 +1190 @@
             space_on_soundcard = getSpaceOnSoundcard();
 
             if (space_on_soundcard != last_space_on_soundcard) {
-                VERBOSE(VB_AUDIO, QString("%1 bytes free on soundcard")
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP, QString("%1 bytes free on soundcard")
                         .arg(space_on_soundcard));
                 last_space_on_soundcard = space_on_soundcard;
             }
@@ -700 +1203 @@
                     WriteAudio(zeros, fragment_size);
                 } else {
                     // this should never happen now -dag
-                    VERBOSE(VB_AUDIO,
+                    VERBOSE(VB_AUDIO|VB_TIMESTAMP,
                             QString("waiting for space on soundcard "
                                     "to write zeros: have %1 need %2")
                             .arg(space_on_soundcard).arg(fragment_size));
@@ -736 +1239 @@
         if (fragment_size > audiolen(true))
         {
             if (audiolen(true) > 0)  // only log if we're sending some audio
-                VERBOSE(VB_AUDIO,
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP,
                         QString("audio waiting for buffer to fill: "
                                 "have %1 want %2")
                         .arg(audiolen(true)).arg(fragment_size));
 
-            VERBOSE(VB_AUDIO, "Broadcasting free space avail");
+            //VERBOSE(VB_AUDIO|VB_TIMESTAMP, "Broadcasting free space avail");
             pthread_mutex_lock(&audio_buflock);
             pthread_cond_broadcast(&audio_bufsig);
             pthread_mutex_unlock(&audio_buflock);
@@ -755 +1258 @@
         if (fragment_size > space_on_soundcard)
         {
             if (space_on_soundcard != last_space_on_soundcard) {
-                VERBOSE(VB_AUDIO,
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP,
                         QString("audio waiting for space on soundcard: "
                                 "have %1 need %2")
                         .arg(space_on_soundcard).arg(fragment_size));
@@ -817 +1320 @@
 
         /* update raud */
         raud = (raud + fragment_size) % AUDBUFSIZE;
-        VERBOSE(VB_AUDIO, "Broadcasting free space avail");
+        //VERBOSE(VB_AUDIO|VB_TIMESTAMP, "Broadcasting free space avail");
         pthread_cond_broadcast(&audio_bufsig);
 
         written_size = fragment_size;

libs/libmyth/audiooutputalsa.cpp

@@ -75 +75 @@
     }
     else
     {
-        fragment_size = 6144; // nicely divisible by 2,4,6,8 channels @ 16-bits
+        //fragment_size = 6144; // nicely divisible by 2,4,6,8 channels @ 16-bits
+        //fragment_size = 3072*audio_channels; // nicely divisible by 2,4,6,8 channels @ 16-bits
+        fragment_size = (audio_bits * audio_channels * audio_samplerate) / (8*30);
         buffer_time = 500000;  // .5 seconds
         period_time = buffer_time / 4;  // 4 interrupts per buffer
     }
@@ -148 +150 @@
     
     tmpbuf = aubuf;
 
-    VERBOSE(VB_AUDIO, QString("WriteAudio: Preparing %1 bytes (%2 frames)")
+    VERBOSE(VB_AUDIO|VB_TIMESTAMP, QString("WriteAudio: Preparing %1 bytes (%2 frames)")
             .arg(size).arg(frames));
     
     while (frames > 0) 

libs/libmythsoundtouch/TDStretch.cpp

@@ -96 +96 @@
 
     pMidBuffer = NULL;
     pRefMidBufferUnaligned = NULL;
+    midBufferLength = 0;
     overlapLength = 0;
 
     setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
@@ -108 +109 @@
 
 TDStretch::~TDStretch()
 {
-    delete[] pMidBuffer;
-    delete[] pRefMidBufferUnaligned;
+    if (midBufferLength)
+    {
+        delete[] pMidBuffer;
+        delete[] pRefMidBufferUnaligned;
+        midBufferLength = 0;
+    }
 }
 
 
@@ -196 +201 @@
 
 void TDStretch::clearMidBuffer()
 {
-    if (bMidBufferDirty) 
+    if (bMidBufferDirty && midBufferLength) 
     {
-        memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
+        memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength);
         bMidBufferDirty = FALSE;
     }
 }
@@ -239 +244 @@
 // Seeks for the optimal overlap-mixing position.
 uint TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
 {
+#ifdef MULTICHANNEL
+    if (channels > 2) 
+    {
+        // stereo sound
+        if (bQuickseek) 
+        {
+            return seekBestOverlapPositionMultiQuick(refPos);
+        } 
+        else 
+        {
+            return seekBestOverlapPositionMulti(refPos);
+        }
+    } 
+    else 
+#endif
     if (channels == 2) 
     {
         // stereo sound
@@ -272 +292 @@
 // of 'ovlPos'.
 inline void TDStretch::overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const
 {
+#ifdef MULTICHANNEL
+    if (channels > 2) 
+    {
+        overlapMulti(output, input + channels * ovlPos);
+    }
+    else 
+#endif
     if (channels == 2) 
     {
         // stereo sound
@@ -291 +318 @@
 // The best position is determined as the position where the two overlapped
 // sample sequences are 'most alike', in terms of the highest cross-correlation
 // value over the overlapping period
+uint TDStretch::seekBestOverlapPositionMulti(const SAMPLETYPE *refPos) 
+{
+    uint bestOffs;
+    LONG_SAMPLETYPE bestCorr, corr;
+    uint i;
+
+    // Slopes the amplitudes of the 'midBuffer' samples
+    precalcCorrReference();
+
+    bestCorr = INT_MIN;
+    bestOffs = 0;
+
+    // Scans for the best correlation value by testing each possible position
+    // over the permitted range.
+    for (i = 0; i < seekLength; i ++) 
+    {
+        // Calculates correlation value for the mixing position corresponding
+        // to 'i'
+        corr = calcCrossCorrMulti(refPos + channels * i, pRefMidBuffer);
+
+        // Checks for the highest correlation value
+        if (corr > bestCorr) 
+        {
+            bestCorr = corr;
+            bestOffs = i;
+        }
+    }
+    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+    clearCrossCorrState();
+
+    return bestOffs;
+}
+
+
+// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+uint TDStretch::seekBestOverlapPositionMultiQuick(const SAMPLETYPE *refPos) 
+{
+    uint j;
+    uint bestOffs;
+    LONG_SAMPLETYPE bestCorr, corr;
+    uint scanCount, corrOffset, tempOffset;
+
+    // Slopes the amplitude of the 'midBuffer' samples
+    precalcCorrReference();
+
+    bestCorr = INT_MIN;
+    bestOffs = 0;
+    corrOffset = 0;
+    tempOffset = 0;
+
+    // Scans for the best correlation value using four-pass hierarchical search.
+    //
+    // The look-up table 'scans' has hierarchical position adjusting steps.
+    // In first pass the routine searhes for the highest correlation with 
+    // relatively coarse steps, then rescans the neighbourhood of the highest
+    // correlation with better resolution and so on.
+    for (scanCount = 0;scanCount < 4; scanCount ++) 
+    {
+        j = 0;
+        while (scanOffsets[scanCount][j]) 
+        {
+            tempOffset = corrOffset + scanOffsets[scanCount][j];
+            if (tempOffset >= seekLength) break;
+
+            // Calculates correlation value for the mixing position corresponding
+            // to 'tempOffset'
+            corr = calcCrossCorrMulti(refPos + channels * tempOffset, pRefMidBuffer);
+
+            // Checks for the highest correlation value
+            if (corr > bestCorr) 
+            {
+                bestCorr = corr;
+                bestOffs = tempOffset;
+            }
+            j ++;
+        }
+        corrOffset = bestOffs;
+    }
+    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+    clearCrossCorrState();
+
+    return bestOffs;
+}
+
+
+// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
 uint TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos) 
 {
     uint bestOffs;
@@ -512 +635 @@
 void TDStretch::setChannels(uint numChannels)
 {
     if (channels == numChannels) return;
+#ifdef MULTICHANNEL
+    assert(numChannels >= 1 && numChannels <= MULTICHANNEL);
+#else
     assert(numChannels == 1 || numChannels == 2);
+#endif
 
     channels = numChannels;
     inputBuffer.setChannels(channels);
@@ -635 +762 @@
 /// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
 void TDStretch::acceptNewOverlapLength(uint newOverlapLength)
 {
-    uint prevOvl;
+    //uint prevOvl;
 
-    prevOvl = overlapLength;
+    //prevOvl = overlapLength;
     overlapLength = newOverlapLength;
 
-    if (overlapLength > prevOvl)
+    //if (overlapLength > prevOvl)
+    if (overlapLength*channels > midBufferLength)
     {
-        delete[] pMidBuffer;
-        delete[] pRefMidBufferUnaligned;
+        if (midBufferLength)
+        {
+            delete[] pMidBuffer;
+            delete[] pRefMidBufferUnaligned;
+            midBufferLength = 0;
+        }
 
-        pMidBuffer = new SAMPLETYPE[overlapLength * 2];
+        pMidBuffer = new SAMPLETYPE[overlapLength * channels];
         bMidBufferDirty = TRUE;
         clearMidBuffer();
+        midBufferLength = overlapLength * channels;
 
-        pRefMidBufferUnaligned = new SAMPLETYPE[2 * overlapLength + 16 / sizeof(SAMPLETYPE)];
+        pRefMidBufferUnaligned = new SAMPLETYPE[channels * overlapLength + 16 / sizeof(SAMPLETYPE)];
         // ensure that 'pRefMidBuffer' is aligned to 16 byte boundary for efficiency
         pRefMidBuffer = (SAMPLETYPE *)((((ulong)pRefMidBufferUnaligned) + 15) & -16);
     }
@@ -718 +851 @@
 
 #ifdef INTEGER_SAMPLES
 
+#ifdef MULTICHANNEL
 // Slopes the amplitude of the 'midBuffer' samples so that cross correlation
 // is faster to calculate
+void TDStretch::precalcCorrReference()
+{
+    int i,j;
+    int temp, temp2;
+    short *src = pMidBuffer;
+    short *dest = pRefMidBuffer;
+
+    for (i=0 ; i < (int)overlapLength ;i ++) 
+    {
+        temp = i * (overlapLength - i);
+
+        for(j=0;j<channels;j++)
+        {
+            temp2 = (*src++ * temp) / slopingDivider;
+            *dest++ = (short)(temp2);
+        }
+    }
+}
+#endif
+
+// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
+// is faster to calculate
 void TDStretch::precalcCorrReferenceStereo()
 {
     int i, cnt2;
@@ -772 +928 @@
     }
 }
 
+#ifdef MULTICHANNEL
+// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo' 
+// version of the routine.
+void TDStretch::overlapMulti(short *output, const short *input) const
+{
+    int i,j;
+    short temp;
+    //uint cnt2;
+    const short *ip = input;
+    short *op = output;
+    const short *md = pMidBuffer;
 
+    for (i = 0; i < (int)overlapLength ; i ++) 
+    {
+        temp = (short)(overlapLength - i);
+        for(j=0;j<channels;j++)
+            *op++ = (*ip++ * i + *md++ * temp )  / overlapLength;
+    }
+}
+#endif
+
+
 /// Calculates overlap period length in samples.
 /// Integer version rounds overlap length to closest power of 2
 /// for a divide scaling operation.
@@ -824 +1001 @@
     return corr;
 }
 
+#ifdef MULTICHANNEL
+long TDStretch::calcCrossCorrMulti(const short *mixingPos, const short *compare) const
+{
+    long corr;
+    uint i;
+
+    corr = 0;
+    for (i = channels; i < channels * overlapLength; i++) 
+    {
+        corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;
+    }
+
+    return corr;
+}
+#endif
+
 #endif // INTEGER_SAMPLES
 
 //////////////////////////////////////////////////////////////////////////////
@@ -931 +1124 @@
     return corr;
 }
 
-#endif // FLOAT_SAMPLES
- No newline at end of file
+#endif // FLOAT_SAMPLES

libs/libmythsoundtouch/TDStretch.h

@@ -100 +100 @@
     SAMPLETYPE *pMidBuffer;
     SAMPLETYPE *pRefMidBuffer;
     SAMPLETYPE *pRefMidBufferUnaligned;
+    uint midBufferLength;
     uint overlapLength;
     uint overlapDividerBits;
     uint slopingDivider;
@@ -123 +124 @@
     virtual void clearCrossCorrState();
     void calculateOverlapLength(uint overlapMs);
 
+#ifdef MULTICHANNEL
+    virtual LONG_SAMPLETYPE calcCrossCorrMulti(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+#endif
     virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
     virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
 
+#ifdef MULTICHANNEL
+    virtual uint seekBestOverlapPositionMulti(const SAMPLETYPE *refPos);
+    virtual uint seekBestOverlapPositionMultiQuick(const SAMPLETYPE *refPos);
+#endif
     virtual uint seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
     virtual uint seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
     virtual uint seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
     virtual uint seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
     uint seekBestOverlapPosition(const SAMPLETYPE *refPos);
 
+#ifdef MULTICHANNEL
+    virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+#endif
     virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
     virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
 
     void clearMidBuffer();
     void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
 
+#ifdef MULTICHANNEL
+    void precalcCorrReference();
+#endif
     void precalcCorrReferenceMono();
     void precalcCorrReferenceStereo();
 
@@ -225 +239 @@
     class TDStretchMMX : public TDStretch
     {
     protected:
+#ifdef MULTICHANNEL
+        //long calcCrossCorrMulti(const short *mixingPos, const short *compare) const;
+#endif
         long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;
         virtual void overlapStereo(short *output, const short *input) const;
         virtual void clearCrossCorrState();
@@ -237 +254 @@
     class TDStretch3DNow : public TDStretch
     {
     protected:
+#ifdef MULTICHANNEL
+        //double calcCrossCorrMulti(const float *mixingPos, const float *compare) const;
+#endif
         double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
     };
 #endif /// ALLOW_3DNOW
@@ -247 +267 @@
     class TDStretchSSE : public TDStretch
     {
     protected:
+#ifdef MULTICHANNEL
+        //double calcCrossCorrMulti(const float *mixingPos, const float *compare) const;
+#endif
         double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
     };
 

libs/libmythsoundtouch/RateTransposer.cpp

@@ -330 +330 @@
 {
     if (uChannels == numchannels) return;
 
+#ifdef MULTICHANNEL
+    assert(numchannels >= 1 && numchannels <= MULTICHANNEL);
+#else
     assert(numchannels == 1 || numchannels == 2);
+#endif
     uChannels = numchannels;
 
     storeBuffer.setChannels(uChannels);

libs/libmythsoundtouch/STTypes.h

@@ -61 +61 @@
     #define INTEGER_SAMPLES       //< 16bit integer samples
     //#define FLOAT_SAMPLES       //< 32bit float samples
 
+    #define MULTICHANNEL 6
 
     /// Define this to allow CPU-specific assembler optimizations. Notice that 
     /// having this enabled on non-x86 platforms doesn't matter; the compiler can 

libs/libmythsoundtouch/SoundTouch.cpp

@@ -140 +140 @@
 // Sets the number of channels, 1 = mono, 2 = stereo
 void SoundTouch::setChannels(uint numChannels)
 {
+#ifdef MULTICHANNEL
+    if (numChannels < 1 || numChannels > MULTICHANNEL)
+#else
     if (numChannels != 1 && numChannels != 2) 
+#endif
     {
         throw std::runtime_error("Illegal number of channels");
     }

programs/mythfrontend/globalsettings.cpp

@@ -36 +36 @@
         dev.setNameFilter("adsp*");
         gc->fillSelectionsFromDir(dev);
     }
+#ifdef USE_ALSA
+    gc->addSelection("ALSA:default", "ALSA:default");
+    gc->addSelection("ALSA:analog", "ALSA:analog");
+    gc->addSelection("ALSA:digital", "ALSA:digital");
+    gc->addSelection("ALSA:mixed-analog", "ALSA:mixed-analog");
+    gc->addSelection("ALSA:mixed-digital", "ALSA:mixed-digital");
+#endif
+#ifdef USE_ARTS
+    gc->addSelection("ARTS:", "ARTS:");
+#endif
+#ifdef USE_JACK
+    gc->addSelection("JACK:output", "JACK:output");
+#endif
+    gc->addSelection("NULL", "NULL");
 
     return gc;
 }
 
+static HostComboBox *MaxAudioChannels()
+{
+    HostComboBox *gc = new HostComboBox("MaxChannels",false);
+    gc->setLabel(QObject::tr("Max Audio Channels"));
+    //gc->addSelection(QObject::tr("Mono"), "1");
+    //gc->addSelection(QObject::tr("Stereo L+R"), "2", true); // default
+    //gc->addSelection(QObject::tr("3 Channel: L C R"), "3");
+    //gc->addSelection(QObject::tr("4 Channel: L R LS RS"), "4");
+    //gc->addSelection(QObject::tr("5 Channel: L C R LS RS"), "5");
+    //gc->addSelection(QObject::tr("6 Channel: L C R LS RS LFE"), "6");
+    gc->addSelection(QObject::tr("Stereo"), "2", true); // default
+    gc->addSelection(QObject::tr("6 Channel"), "6");
+    gc->setHelpText(
+            QObject::tr("Set the maximum number of audio channels to be decoded. "
+                "This is for multi-channel/surround audio playback."));
+    return gc;
+}
+
 static HostCheckBox *MythControlsVolume()
 {
     HostCheckBox *gc = new HostCheckBox("MythControlsVolume");
@@ -2092 +2124 @@
          setUseLabel(false);
 
          addChild(AudioOutputDevice());
+#if 0
+         ConfigurationGroup *hg = new HorizontalConfigurationGroup(false, false);
+         ConfigurationGroup* settingsLeft = new VerticalConfigurationGroup(false,false);
+         settingsLeft->addChild(AC3PassThrough());
+#ifdef CONFIG_DTS
+         settingsLeft->addChild(DTSPassThrough());
+#endif
+         settingsLeft->addChild(AggressiveBuffer());
+         hg->addChild(settingsLeft);
+
+         ConfigurationGroup* settingsRight = new VerticalConfigurationGroup(false,false);
+         settingsRight->addChild(MaxAudioChannels());
+         hg->addChild(settingsRight);
+
+         addChild(hg);
+#else
          addChild(AC3PassThrough());
 #ifdef CONFIG_DTS
          addChild(DTSPassThrough());
 #endif
          addChild(AggressiveBuffer());
+         addChild(MaxAudioChannels());
+#endif
 
          Setting* volumeControl = MythControlsVolume();
          addChild(volumeControl);

programs/mythtranscode/transcode.cpp

@@ -46 +46 @@
 
     // reconfigure sound out for new params
     virtual void Reconfigure(int audio_bits,
-                        int audio_channels, int audio_samplerate)
+                        int audio_channels, int audio_samplerate,
+                        void * = NULL)
     {
+        ClearError();
         (void)audio_samplerate;
         bits = audio_bits;
         channels = audio_channels;
         bytes_per_sample = bits * channels / 8;
+        if (channels>2)
+            Error("Invalid channel count");
     }
 
     // dsprate is in 100 * samples/second

libs/libmythtv/avformatdecoder.h

@@ -238 +238 @@
     bool              allow_ac3_passthru;
     bool              allow_dts_passthru;
     bool              disable_passthru;
+    int               max_channels;
 
     AudioInfo         audioIn;
     AudioInfo         audioOut;

libs/libmythtv/avformatdecoder.cpp

@@ -38 +38 @@
 #define MAX_AC3_FRAME_SIZE 6144
 
 /** Set to zero to allow any number of AC3 channels. */
+#define MAXCHANNELSELECT 1
+#if MAXCHANNELSELECT
+#define MAX_OUTPUT_CHANNELS compiler error
+#else
 #define MAX_OUTPUT_CHANNELS 2
+#endif
 
 static int dts_syncinfo(uint8_t *indata_ptr, int *flags,
                         int *sample_rate, int *bit_rate);
@@ -286 +291 @@
 #ifdef CONFIG_DTS
     allow_dts_passthru = gContext->GetNumSetting("DTSPassThru", false);
 #endif
+    max_channels = gContext->GetNumSetting("MaxChannels", 2);
 
     audioIn.sample_size = -32; // force SetupAudioStream to run once
 }
@@ -406 +412 @@
     framesPlayed = lastKey;
     framesRead = lastKey;
 
+    VERBOSE(VB_PLAYBACK, QString("AvFormatDecoder::DoFastForward newframe %5 frame %1 fps %2 ts %3 disc %4 cur_dts %6 adj %7 newts %8 fsa %9")
+        .arg(desiredFrame)
+        .arg(fps)
+        .arg(ts)
+        .arg(discardFrames)
+        .arg(framesPlayed)
+        .arg(st->cur_dts)
+        .arg(adj_cur_dts)
+        .arg(newts)
+        .arg(frameseekadjust)
+        );
+
     int normalframes = desiredFrame - framesPlayed;
 
+#if 0
+    if (!exactseeks)
+        normalframes = 0;
+#endif
+
     SeekReset(lastKey, normalframes, discardFrames, discardFrames);
 
     if (discardFrames)
@@ -739 +762 @@
 
     fmt->flags &= ~AVFMT_NOFILE;
 
+#if 1
+    if ((m_playbackinfo) || livetv || watchingrecording)
+    {
+        const char *name = ic->av_class->item_name(ic);
+        VERBOSE(VB_GENERAL, QString("libavformat type %1").arg(name));
+    }
+#endif
+ 
+    //struct timeval one, two, res;
+    //gettimeofday(&one, NULL);
+
     av_estimate_timings(ic);
     av_read_frame_flush(ic);
 
@@ -769 +803 @@
     // If we don't have a position map, set up ffmpeg for seeking
     if (!recordingHasPositionMap)
     {
+        const char *name = ic->av_class->item_name(ic);
         VERBOSE(VB_PLAYBACK, LOC +
-                "Recording has no position -- using libavformat seeking.");
+                QString("Recording has no position -- using libavformat seeking. %1").arg(name));
         int64_t dur = ic->duration / (int64_t)AV_TIME_BASE;
 
         if (dur > 0)
@@ -1048 +1083 @@
                             <<") already open, leaving it alone.");
                 }
                 assert(enc->codec_id);
+                VERBOSE(VB_GENERAL, QString("AVFD: codec %1 has %2 channels").arg(codec_id_string(enc->codec_id)).arg(enc->channels));
+#if 0
+                if (enc->channels > 2)
+                    enc->channels = 2;
+#endif
 
+#if 0
                 // HACK BEGIN REALLY UGLY HACK FOR DTS PASSTHRU
                 if (enc->codec_id == CODEC_ID_DTS)
                 {
@@ -1057 +1098 @@
                     // enc->bit_rate = what??;
                 }
                 // HACK END REALLY UGLY HACK FOR DTS PASSTHRU
+#endif
 
                 bitrate += enc->bit_rate;
                 break;
@@ -1557 +1599 @@
         {
             long long startpos = pkt->pos;
 
-            VERBOSE(VB_PLAYBACK, LOC + 
+            VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + 
                     QString("positionMap[ %1 ] == %2.")
                     .arg(prevgoppos / keyframedist)
                     .arg((int)startpos));
@@ -2268 +2310 @@
 
         AVStream *curstream = ic->streams[pkt->stream_index];
 
+#if 0
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + QString("timecode pts:%1 dts:%2 codec:%3")
+                .arg(pkt->pts)
+                .arg(pkt->dts)
+                .arg((curstream && curstream->codec)?curstream->codec->codec_type:-1)
+               );
+#endif
+
         if (pkt->dts != (int64_t)AV_NOPTS_VALUE)
             pts = (long long)(av_q2d(curstream->time_base) * pkt->dts * 1000);
 
@@ -2381 +2431 @@
                     if (!curstream->codec->channels)
                     {
                         QMutexLocker locker(&avcodeclock);
+#if MAXCHANNELSELECT
+                        VERBOSE(VB_IMPORTANT, LOC + QString("Setting channels to %1").arg(audioOut.channels));
+                        curstream->codec->channels = audioOut.channels;
+#else
                         curstream->codec->channels = MAX_OUTPUT_CHANNELS;
+#endif
                         ret = avcodec_decode_audio(
                             curstream->codec, audioSamples,
                             &data_size, ptr, len);
@@ -2432 +2487 @@
                     {
                         AVCodecContext *ctx = curstream->codec;
 
+#if MAXCHANNELSELECT
                         if ((ctx->channels == 0) ||
+                            (ctx->channels > audioOut.channels))
+                            ctx->channels = audioOut.channels;
+#else
+                        if ((ctx->channels == 0) ||
                             (ctx->channels > MAX_OUTPUT_CHANNELS))
                             ctx->channels = MAX_OUTPUT_CHANNELS;
+#endif
 
                         ret = avcodec_decode_audio(
                             ctx, audioSamples, &data_size, ptr, len);
@@ -2470 +2531 @@
                                 (curstream->codec->channels * 2) / 
                                 curstream->codec->sample_rate);
 
+                    VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + QString("audio timecode %1 %2 %3 %4")
+                            .arg(pkt->pts)
+                            .arg(pkt->dts)
+                            .arg(temppts).arg(lastapts));
                     GetNVP()->AddAudioData((char *)audioSamples, data_size,
                                            temppts);
 
@@ -2561 +2626 @@
                     else
                         temppts = lastvpts;
 
+                    VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + QString("video timecode %1 %2 %3 %4")
+                            .arg(pkt->pts)
+                            .arg(pkt->dts)
+                            .arg(temppts).arg(lastvpts));
 /* XXX: Broken.
                     if (mpa_pic.qscale_table != NULL && mpa_pic.qstride > 0 &&
                         context->height == picframe->height)
@@ -2655 +2724 @@
 
 void AvFormatDecoder::SetDisablePassThrough(bool disable)
 {
+#if MAXCHANNELSELECT
+    // can only disable never reenable as once tiemstretch is on its on for the session
+    if (disable_passthru)
+        return;
+#endif
     if (selectedAudioStream.av_stream_index < 0)
     {
         disable_passthru = disable;
         return;
     }
-
+ 
     if (disable != disable_passthru)
     {
         disable_passthru = disable;
@@ -2687 +2761 @@
     AVCodecContext *codec_ctx = NULL;
     AudioInfo old_in  = audioIn;
     AudioInfo old_out = audioOut;
+    bool using_passthru = false;
 
     if ((currentAudioTrack >= 0) &&
         (selectedAudioStream.av_stream_index <= ic->nb_streams) &&
@@ -2696 +2771 @@
         assert(curstream->codec);
         codec_ctx = curstream->codec;        
         bool do_ac3_passthru = (allow_ac3_passthru && !transcoding &&
-                                !disable_passthru &&
                                 (codec_ctx->codec_id == CODEC_ID_AC3));
         bool do_dts_passthru = (allow_dts_passthru && !transcoding &&
-                                !disable_passthru &&
                                 (codec_ctx->codec_id == CODEC_ID_DTS));
+        using_passthru = do_ac3_passthru || do_dts_passthru;
         info = AudioInfo(codec_ctx->codec_id,
                          codec_ctx->sample_rate, codec_ctx->channels,
-                         do_ac3_passthru || do_dts_passthru);
+                         using_passthru && !disable_passthru);
     }
 
     if (info == audioIn)
         return false; // no change
 
+    QString ptmsg = "";
+    if (using_passthru)
+    {
+        ptmsg = QString(" using passthru");
+    }
     VERBOSE(VB_AUDIO, LOC + "Initializing audio parms from " +
-            QString("audio track #%1").arg(currentAudioTrack+1));
+            QString("audio track #%1")
+                .arg(currentAudioTrack+1)
+            + ptmsg );
 
     audioOut = audioIn = info;
+#if MAXCHANNELSELECT
+    if (using_passthru)
+#else
     if (audioIn.do_passthru)
+#endif
     {
         // A passthru stream looks like a 48KHz 2ch (@ 16bit) to the sound card
-        audioOut.channels    = 2;
-        audioOut.sample_rate = 48000;
-        audioOut.sample_size = 4;
+        AudioInfo digInfo = audioOut;
+        if (!disable_passthru)
+        {
+            digInfo.channels    = 2;
+            digInfo.sample_rate = 48000;
+            digInfo.sample_size = 4;
+        }
+        if (audioOut.channels > max_channels)
+        {
+            audioOut.channels = max_channels;
+            audioOut.sample_size = audioOut.channels * 2;
+            codec_ctx->channels = audioOut.channels;
+        }
+#if MAXCHANNELSELECT
+        VERBOSE(VB_AUDIO, LOC + "Audio format changed digital passthrough " +
+                QString("%1\n\t\t\tfrom %2 ; %3\n\t\t\tto   %4 ; %5")
+                .arg(digInfo.toString())
+                .arg(old_in.toString()).arg(old_out.toString())
+                .arg(audioIn.toString()).arg(audioOut.toString()));
+
+        if (digInfo.sample_rate > 0)
+            GetNVP()->SetEffDsp(digInfo.sample_rate * 100);
+
+        //GetNVP()->SetAudioParams(audioOut.bps(), audioOut.channels,
+        //                         audioOut.sample_rate);
+        GetNVP()->SetAudioParams(digInfo.bps(), digInfo.channels,
+                                 digInfo.sample_rate);
+        // allow the audio stuff to reencode
+        GetNVP()->SetAudioCodec(codec_ctx);
+        GetNVP()->ReinitAudio();
+        return true;
+#endif
     }
+#if MAXCHANNELSELECT
     else
     {
+        if (audioOut.channels > max_channels)
+        {
+            audioOut.channels = max_channels;
+            audioOut.sample_size = audioOut.channels * 2;
+            codec_ctx->channels = audioOut.channels;
+        }
+    }
+    bool audiook;
+    do 
+    {
+#else
+    else
+    {
         if (audioOut.channels > MAX_OUTPUT_CHANNELS)
         {
             audioOut.channels = MAX_OUTPUT_CHANNELS;
@@ -2729 +2857 @@
             codec_ctx->channels = MAX_OUTPUT_CHANNELS;
         }
     }
+#endif
 
     VERBOSE(VB_AUDIO, LOC + "Audio format changed " +
             QString("\n\t\t\tfrom %1 ; %2\n\t\t\tto   %3 ; %4")
@@ -2740 +2869 @@
 
     GetNVP()->SetAudioParams(audioOut.bps(), audioOut.channels,
                              audioOut.sample_rate);
-    GetNVP()->ReinitAudio();
+    // allow the audio stuff to reencode
+    GetNVP()->SetAudioCodec(using_passthru?codec_ctx:NULL);
+    QString errMsg = GetNVP()->ReinitAudio();
+#if MAXCHANNELSELECT
+        audiook = errMsg.isEmpty();
+        if (!audiook)
+        {
+            switch (audioOut.channels)
+            {
+#if 0
+                case 8:
+                    audioOut.channels = 6;
+                    break;
+#endif
+                case 6:
+                    audioOut.channels = 5;
+                    break;
+                case 5:
+                    audioOut.channels = 4;
+                    break;
+                case 4:
+                    audioOut.channels = 3;
+                    break;
+                case 3:
+                    audioOut.channels = 2;
+                    break;
+                case 2:
+                    audioOut.channels = 1;
+                    break;
+                default:
+                    // failed to reconfigure under any circumstances
+                    audiook = true;
+                    audioOut.channels = 0;
+                    break;
+            }
+            audioOut.sample_size = audioOut.channels * 2;
+            codec_ctx->channels = audioOut.channels;
+        }
+    } while (!audiook);
+#endif
 
     return true;
 }

libs/libmythtv/NuppelVideoPlayer.h

@@ -107 +107 @@
     void SetVideoParams(int w, int h, double fps, int keydist,
                         float a = 1.33333, FrameScanType scan = kScan_Ignore);
     void SetAudioParams(int bits, int channels, int samplerate);
+    void SetAudioCodec(void *ac);
     void SetEffDsp(int dsprate);
     void SetFileLength(int total, int frames);
     void Zoom(int direction);
@@ -142 +143 @@
     bool    AtNormalSpeed(void) const         { return next_normal_speed; }
     bool    IsDecoderThreadAlive(void) const  { return decoder_thread_alive; }
     bool    IsNearEnd(long long framesRemaining = -1) const;
+    float   GetAudioStretchFactor() { return audio_stretchfactor; }
     bool    PlayingSlowForPrebuffer(void) const { return m_playing_slower; }
     bool    HasAudioIn(void) const            { return !no_audio_in; }
     bool    HasAudioOut(void) const           { return !no_audio_out; }
@@ -170 +172 @@
     bool Play(float speed = 1.0, bool normal = true,
               bool unpauseaudio = true);
     bool GetPause(void) const;
+    float GetNextPlaySpeed() { return next_play_speed; }
 
     // Seek stuff
     bool FastForward(float seconds);
@@ -492 +495 @@
     int      audio_bits;
     int      audio_samplerate;
     float    audio_stretchfactor;
+    void     *audio_codec;
 
     // Picture-in-Picture
     NuppelVideoPlayer *pipplayer;

libs/libmythtv/NuppelVideoPlayer.cpp

@@ -124 +124 @@
       audioOutput(NULL),            audiodevice("/dev/dsp"),
       audio_channels(2),            audio_bits(-1),
       audio_samplerate(44100),      audio_stretchfactor(1.0f),
+      audio_codec(NULL),
       // Picture-in-Picture
       pipplayer(NULL), setpipplayer(NULL), needsetpipplayer(false),
       // Preview window support
@@ -528 +529 @@
 
     if (audioOutput)
     {
-        audioOutput->Reconfigure(audio_bits, audio_channels, audio_samplerate);
+        audioOutput->Reconfigure(audio_bits, audio_channels, audio_samplerate, audio_codec);
         errMsg = audioOutput->GetError();
         if (!errMsg.isEmpty())
             audioOutput->SetStretchFactor(audio_stretchfactor);
@@ -661 +662 @@
         {
             VERBOSE(VB_IMPORTANT, "Video sync method can't support double "
                     "framerate (refresh rate too low for bob deint)");
+            //m_scan = kScan_Ignore;
+            //m_can_double = false;
             FallbackDeint();
         }
     }
@@ -1513 +1516 @@
     warpfactor_avg = (warpfactor + (warpfactor_avg * (WARPAVLEN - 1))) /
                       WARPAVLEN;
 
-    //cerr << "Divergence: " << divergence << "  Rate: " << rate
-    //<< "  Warpfactor: " << warpfactor << "  warpfactor_avg: "
-    //<< warpfactor_avg << endl;
+#if 1
+    VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, QString("A/V "
+        "Divergence: %1 "
+        "  Rate: %2" 
+        "  Warpfactor: %3" 
+        "  warpfactor_avg: %4")
+        .arg(divergence)
+        .arg(rate)
+        .arg(warpfactor)
+        .arg(warpfactor_avg)
+        );
+#endif
     return divergence;
 }
 
@@ -1596 +1608 @@
     if (diverge < -MAXDIVERGE)
     {
         // If video is way ahead of audio, adjust for it...
-        QString dbg = QString("Video is %1 frames ahead of audio, ")
+        QString dbg = QString("Audio is %1 frames ahead of video, ")
             .arg(-diverge);
 
         // Reset A/V Sync
@@ -1611 +1623 @@
             // decoding; display the frame, but don't wait for A/V Sync.
             videoOutput->PrepareFrame(buffer, ps);
             videoOutput->Show(m_scan);
-            VERBOSE(VB_PLAYBACK, LOC + dbg + "skipping A/V wait.");
+            VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + dbg + "skipping A/V wait.");
         }
         else
         {
             // If we are using software decoding, skip this frame altogether.
-            VERBOSE(VB_PLAYBACK, LOC + dbg + "dropping frame.");
+            VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + dbg + "dropping frame.");
         }
     }
     else if (!using_null_videoout)
@@ -1625 +1637 @@
         if (buffer)
             videoOutput->PrepareFrame(buffer, ps);
 
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, QString("AVSync waitforframe %1 %2").arg(avsync_adjustment).arg(m_double_framerate));
         videosync->WaitForFrame(avsync_adjustment);
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, "AVSync show");
         if (!resetvideo)
             videoOutput->Show(m_scan);
 
@@ -1645 +1659 @@
 
             // Display the second field
             videosync->AdvanceTrigger();
-            videosync->WaitForFrame(0);
+            videosync->WaitForFrame(avsync_adjustment);
             if (!resetvideo)
                 videoOutput->Show(kScan_Intr2ndField);
         }
@@ -1657 +1671 @@
 
     if (output_jmeter && output_jmeter->RecordCycleTime())
     {
-        //cerr << "avsync_delay: " << avsync_delay / 1000
-        //     << ", avsync_avg: " << avsync_avg / 1000
-        //     << ", warpfactor: " << warpfactor
-        //     << ", warpfactor_avg: " << warpfactor_avg << endl;
+#if 1
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, QString("A/V "
+            "avsync_delay: %1" 
+            ", avsync_avg: %2" 
+            ", warpfactor: %3" 
+            ", warpfactor_avg: %4")
+                .arg(avsync_delay / 1000)
+                .arg(avsync_avg / 1000)
+                .arg(warpfactor)
+                .arg(warpfactor_avg));
+#endif
     }
 
     videosync->AdvanceTrigger();
@@ -1671 +1692 @@
         // If audio is way ahead of video, adjust for it...
         // by cutting the frame rate in half for the length of this frame
 
-        avsync_adjustment = frame_interval;
+        //avsync_adjustment = frame_interval;
+        avsync_adjustment = refreshrate;
         lastsync = true;
-        VERBOSE(VB_PLAYBACK, LOC + 
-                QString("Audio is %1 frames ahead of video,\n"
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, LOC + 
+                QString("Video is %1 frames ahead of audio,\n"
                         "\t\t\tdoubling video frame interval.").arg(diverge));
     }
 
     if (audioOutput && normal_speed)
     {
         long long currentaudiotime = audioOutput->GetAudiotime();
-#if 0
-        VERBOSE(VB_PLAYBACK, QString(
+#if 1
+        VERBOSE(VB_PLAYBACK|VB_TIMESTAMP, QString(
                     "A/V timecodes audio %1 video %2 frameinterval %3 "
                     "avdel %4 avg %5 tcoffset %6")
                 .arg(currentaudiotime)
@@ -1968 +1990 @@
             {
                 VERBOSE(VB_IMPORTANT, "Video sync method can't support double "
                         "framerate (refresh rate too low for bob deint)");
+                //m_scan = kScan_Ignore;
+                //m_can_double = false;
                 FallbackDeint();
             }
         }
@@ -2672 +2696 @@
     audio_samplerate = samplerate;
 }
 
+void NuppelVideoPlayer::SetAudioCodec(void* ac)
+{
+    audio_codec = ac;
+}
+
 void NuppelVideoPlayer::SetEffDsp(int dsprate)
 {
     if (audioOutput)
@@ -2716 +2745 @@
         tc_avcheck_framecounter++;
         if (tc_avcheck_framecounter == 30)
         {
-#define AUTO_RESYNC 1
+#define AUTO_RESYNC 0
 #if AUTO_RESYNC
             // something's terribly, terribly wrong.
             if (tc_lastval[TC_AUDIO] < tc_lastval[TC_VIDEO] - 10000000 ||

libs/libavcodec/a52dec.c

@@ -149 +149 @@
     }
 }
 
+static inline int16_t convert (int32_t i)
+{
+    if (i > 0x43c07fff)
+        return 32767;
+    else if (i < 0x43bf8000)
+        return -32768;
+    else
+        return i - 0x43c00000;
+}
+
+void float2s16_2 (float * _f, int16_t * s16)
+{
+    int i;
+    int32_t * f = (int32_t *) _f;
+
+    for (i = 0; i < 256; i++) {
+        s16[2*i] = convert (f[i]);
+        s16[2*i+1] = convert (f[i+256]);
+    }
+}
+
+void float2s16_4 (float * _f, int16_t * s16)
+{
+    int i;
+    int32_t * f = (int32_t *) _f;
+
+    for (i = 0; i < 256; i++) {
+        s16[4*i] = convert (f[i]);
+        s16[4*i+1] = convert (f[i+256]);
+        s16[4*i+2] = convert (f[i+512]);
+        s16[4*i+3] = convert (f[i+768]);
+    }
+}
+
+void float2s16_5 (float * _f, int16_t * s16)
+{
+    int i;
+    int32_t * f = (int32_t *) _f;
+
+    for (i = 0; i < 256; i++) {
+        s16[5*i] = convert (f[i]);
+        s16[5*i+1] = convert (f[i+256]);
+        s16[5*i+2] = convert (f[i+512]);
+        s16[5*i+3] = convert (f[i+768]);
+        s16[5*i+4] = convert (f[i+1024]);
+    }
+}
+
+int channels_multi (int flags)
+{
+    if (flags & A52_LFE)
+        return 6;
+    else if (flags & 1) /* center channel */
+        return 5;
+    else if ((flags & A52_CHANNEL_MASK) == A52_2F2R)
+        return 4;
+    else
+        return 2;
+}
+
+void float2s16_multi (float * _f, int16_t * s16, int flags)
+{
+    int i;
+    int32_t * f = (int32_t *) _f;
+
+    switch (flags) {
+    case A52_MONO:
+        for (i = 0; i < 256; i++) {
+            s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
+            s16[5*i+4] = convert (f[i]);
+        }
+        break;
+    case A52_CHANNEL:
+    case A52_STEREO:
+    case A52_DOLBY:
+        float2s16_2 (_f, s16);
+        break;
+    case A52_3F:
+        for (i = 0; i < 256; i++) {
+            s16[5*i] = convert (f[i]);
+            s16[5*i+1] = convert (f[i+512]);
+            s16[5*i+2] = s16[5*i+3] = 0;
+            s16[5*i+4] = convert (f[i+256]);
+        }
+        break;
+    case A52_2F2R:
+        float2s16_4 (_f, s16);
+        break;
+    case A52_3F2R:
+        float2s16_5 (_f, s16);
+        break;
+    case A52_MONO | A52_LFE:
+        for (i = 0; i < 256; i++) {
+            s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
+            s16[6*i+4] = convert (f[i+256]);
+            s16[6*i+5] = convert (f[i]);
+        }
+        break;
+    case A52_CHANNEL | A52_LFE:
+    case A52_STEREO | A52_LFE:
+    case A52_DOLBY | A52_LFE:
+        for (i = 0; i < 256; i++) {
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+1] = convert (f[i+512]);
+            s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
+            s16[6*i+5] = convert (f[i]);
+        }
+        break;
+    case A52_3F | A52_LFE:
+        for (i = 0; i < 256; i++) {
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+1] = convert (f[i+768]);
+            s16[6*i+2] = s16[6*i+3] = 0;
+            s16[6*i+4] = convert (f[i+512]);
+            s16[6*i+5] = convert (f[i]);
+        }
+        break;
+    case A52_2F2R | A52_LFE:
+        for (i = 0; i < 256; i++) {
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+1] = convert (f[i+512]);
+            s16[6*i+2] = convert (f[i+768]);
+            s16[6*i+3] = convert (f[i+1024]);
+            s16[6*i+4] = 0;
+            s16[6*i+5] = convert (f[i]);
+        }
+        break;
+    case A52_3F2R | A52_LFE:
+        for (i = 0; i < 256; i++) {
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+1] = convert (f[i+768]);
+            s16[6*i+2] = convert (f[i+1024]);
+            s16[6*i+3] = convert (f[i+1280]);
+            s16[6*i+4] = convert (f[i+512]);
+            s16[6*i+5] = convert (f[i]);
+        }
+        break;
+    }
+}
+
+
 /**** end */
 
 #define HEADER_SIZE 7
@@ -212 +353 @@
             s->inbuf_ptr += len;
             buf_size -= len;
         } else {
+            int chans;
             flags = s->flags;
             if (avctx->channels == 1)
                 flags = A52_MONO;
-            else if (avctx->channels == 2)
-                flags = A52_STEREO;
+            else if (avctx->channels == 2) {
+                if (s->channels>2)
+                    flags = A52_DOLBY;
+                else
+                    flags = A52_STEREO;
+            }
             else
                 flags |= A52_ADJUST_LEVEL;
             level = 1;
+            chans = channels_multi(flags);
             if (s->a52_frame(s->state, s->inbuf, &flags, &level, 384)) {
             fail:
                 s->inbuf_ptr = s->inbuf;
@@ -229 +376 @@
             for (i = 0; i < 6; i++) {
                 if (s->a52_block(s->state))
                     goto fail;
-                float_to_int(s->samples, out_samples + i * 256 * avctx->channels, avctx->channels);
+                //float_to_int(s->samples, out_samples + i * 256 * avctx->channels, avctx->channels);
+                float2s16_multi(s->samples, out_samples + i * 256 * chans, flags);
             }
             s->inbuf_ptr = s->inbuf;
             s->frame_size = 0;
-            *data_size = 6 * avctx->channels * 256 * sizeof(int16_t);
+            //*data_size = 6 * avctx->channels * 256 * sizeof(int16_t);
+            *data_size = 6 * channels_multi(flags) * 256 * sizeof(int16_t);
             break;
         }
     }

libs/libavcodec/ac3enc.c

@@ -1362 +1362 @@
 {
     AC3EncodeContext *s = avctx->priv_data;
     int16_t *samples = data;
+    // expects L C R LS RS LFE
+    // audio format is L R LS RS C LFE
+    static int channel_index[6] = { 0, 4, 1, 2, 3, 5 };
+    /*
+     * A52->Analog->AC3Enc
+     * 1->0->0
+     * 3->1->2
+     * 4->2->3
+     * 5->3->4
+     * 2->4->1
+     * 0->5->5
+     */
     int i, j, k, v, ch;
     int16_t input_samples[N];
     int32_t mdct_coef[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
@@ -1382 +1394 @@
             /* compute input samples */
             memcpy(input_samples, s->last_samples[ch], N/2 * sizeof(int16_t));
             sinc = s->nb_all_channels;
-            sptr = samples + (sinc * (N/2) * i) + ch;
+            sptr = samples + (sinc * (N/2) * i) + channel_index[ch];
             for(j=0;j<N/2;j++) {
                 v = *sptr;
                 input_samples[j + N/2] = v;
@@ -1403 +1415 @@
             v = 14 - log2_tab(input_samples, N);
             if (v < 0)
                 v = 0;
-            exp_samples[i][ch] = v - 8;
+            exp_samples[i][ch] = v - 9;
             lshift_tab(input_samples, N, v);
 
             /* do the MDCT */