Ticket #1104: mythtv_ac3.48.patch

configure

@@ -752 +752 @@
     libfaac
     libfaad
     libfaadbin
+    libfftw3
     libgsm
     libmp3lame
     libnut
@@ -2886 +2887 @@
 echo "libfaac enabled           ${libfaac-no}"
 echo "libfaad enabled           ${libfaad-no}"
 echo "libfaad dlopened          ${libfaadbin-no}"
+echo "libfftw3 support          ${liba52-no}"
 echo "libgsm enabled            ${libgsm-no}"
 echo "libmp3lame enabled        ${libmp3lame-no}"
 echo "libnut enabled            ${libnut-no}"

libs/libs.pro

@@ -6 +6 @@
 # Directories
 SUBDIRS += libavutil libavcodec libavformat libmythsamplerate 
 SUBDIRS += libmythsoundtouch libmythmpeg2 libmythdvdnav
+SUBDIRS += libmythfreesurround
 
 mingw : SUBDIRS += libmyth libmythupnp libmythui
 !mingw: SUBDIRS += libmythupnp libmythui libmyth

libs/libmyth/libmyth.pro

@@ -11 +11 @@
 
 # Input
 HEADERS += audiooutput.h audiooutputbase.h audiooutputnull.h
+HEADERS += audiooutputdigitalencoder.h
 HEADERS += backendselect.h dbsettings.h dialogbox.h
 HEADERS += DisplayRes.h DisplayResScreen.h exitcodes.h
 HEADERS += generictree.h httpcomms.h langsettings.h lcddevice.h
@@ -25 +26 @@
 HEADERS += volumebase.h volumecontrol.h virtualkeyboard.h visual.h xmlparse.h
 HEADERS += mythhdd.h mythcdrom.h
 HEADERS += compat.h
+HEADERS += audiooutputdigitalencoder.h
 
 SOURCES += audiooutput.cpp audiooutputbase.cpp audiooutputnull.cpp
+SOURCES += audiooutputdigitalencoder.cpp
 SOURCES += backendselect.cpp dbsettings.cpp dialogbox.cpp
 SOURCES += DisplayRes.cpp DisplayResScreen.cpp
 SOURCES += generictree.cpp httpcomms.cpp langsettings.cpp lcddevice.cpp
@@ -41 +44 @@
 SOURCES += volumebase.cpp volumecontrol.cpp virtualkeyboard.cpp xmlparse.cpp
 SOURCES += mythhdd.cpp mythcdrom.cpp
 
-INCLUDEPATH += ../libmythsamplerate ../libmythsoundtouch ../.. ../ ./
+INCLUDEPATH += ../libmythsamplerate ../libmythsoundtouch ../libmythfreesurround
+INCLUDEPATH += ../libavcodec ../libavutil
+INCLUDEPATH += ../.. ../ ./
 DEPENDPATH += ../libmythsamplerate ../libmythsoundtouch ../ ../libmythui
-DEPENDPATH += ../libmythupnp
+DEPENDPATH += ../libmythupnp ../libmythfreesurround ../libavcodec ../libavutil
 
-LIBS += -L../libmythsamplerate -lmythsamplerate-$${LIBVERSION}
-LIBS += -L../libmythsoundtouch -lmythsoundtouch-$${LIBVERSION}
-LIBS += -L../libmythui         -lmythui-$${LIBVERSION}
-LIBS += -L../libmythupnp       -lmythupnp-$${LIBVERSION}
 
+LIBS += -L../libmythsamplerate   -lmythsamplerate-$${LIBVERSION}
+LIBS += -L../libmythsoundtouch   -lmythsoundtouch-$${LIBVERSION}
+LIBS += -L../libmythui           -lmythui-$${LIBVERSION}
+LIBS += -L../libmythupnp         -lmythupnp-$${LIBVERSION}
+LIBS += -L../libmythfreesurround -lmythfreesurround-$${LIBVERSION}
+LIBS += -L../libavcodec          -lmythavcodec-$${LIBVERSION}
+LIBS += -L../libavutil           -lmythavutil-$${LIBVERSION}
+
 TARGETDEPS += ../libmythsamplerate/libmythsamplerate-$${MYTH_LIB_EXT}
 TARGETDEPS += ../libmythsoundtouch/libmythsoundtouch-$${MYTH_LIB_EXT}
+TARGETDEPS += ../libmythfreesurround/libmythfreesurround-$${MYTH_LIB_EXT}
 
 # Install headers so that plugins can compile independently
 inc.path = $${PREFIX}/include/mythtv/
@@ -221 +231 @@
 use_hidesyms {
     QMAKE_CXXFLAGS += -fvisibility=hidden
 }
+
+contains( CONFIG_LIBA52, yes ) {
+    LIBS += -la52
+}
+
+contains( CONFIG_LIBFFTW3, yes ) {
+    LIBS += -lfftw3f
+}

libs/libmyth/audiooutput.h

@@ -31 +31 @@
     virtual ~AudioOutput() { };
 
     // reconfigure sound out for new params
-    virtual void Reconfigure(int audio_bits, int audio_channels,
-                             int audio_samplerate, bool audio_passthru) = 0;
+    virtual void Reconfigure(int audio_bits, 
+                             int audio_channels, 
+                             int audio_samplerate,
+                             bool audio_passthru,
+                             void* audio_codec = NULL) = 0;
     
     virtual void SetStretchFactor(float factor);
+    virtual float GetStretchFactor(void) { return 1.0f; }
 
     // do AddSamples calls block?
     virtual void SetBlocking(bool blocking) = 0;
@@ -76 +80 @@
         lastError = msg;
         VERBOSE(VB_IMPORTANT, "AudioOutput Error: " + lastError);
     }
+    void ClearError(void) { lastError = QString::null; }
 
     void Warn(QString msg)
     {

libs/libmyth/audiooutputdx.h

@@ -35 +35 @@
     /// END HACK HACK HACK HACK
         
     virtual void Reset(void);
-    virtual void Reconfigure(int audio_bits,       int audio_channels,
-                             int audio_samplerate, int audio_passthru);
+    virtual void Reconfigure(int audio_bits, 
+                             int audio_channels, 
+                             int audio_samplerate,
+                             bool audio_passthru,
+                             AudioCodecMode aom = AUDIOCODECMODE_NORMAL);
     virtual void SetBlocking(bool blocking);
 
     virtual bool AddSamples(char *buffer, int samples, long long timecode);

libs/libmyth/audiooutputdx.cpp

@@ -130 +130 @@
     // FIXME: kedl: not sure what else could be required here?
 }
 
-void AudioOutputDX::Reconfigure(int audio_bits, int audio_channels,
-                                int audio_samplerate, int audio_passthru)
+void AudioOutputDX::Reconfigure(int audio_bits, 
+                                int audio_channels, 
+                                int audio_samplerate,
+                                int audio_passthru,
+                                AudioCodecMode laom)
 {
     if (dsbuffer)
         DestroyDSBuffer();

libs/libmyth/audiooutputbase.h

@@ -16 +16 @@
 // MythTV headers
 #include "audiooutput.h"
 #include "samplerate.h"
-#include "SoundTouch.h"
 
-#define AUDBUFSIZE 768000
+namespace soundtouch {
+class SoundTouch;
+};
+class FreeSurround;
+class AudioOutputDigitalEncoder;
+struct AVCodecContext;
+
 #define AUDIO_SRC_IN_SIZE   16384
 #define AUDIO_SRC_OUT_SIZE (16384*6)
 #define AUDIO_TMP_BUF_SIZE (16384*6)
 
+//#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:
@@ -35 +45 @@
     virtual ~AudioOutputBase();
 
     // reconfigure sound out for new params
-    virtual void Reconfigure(int audio_bits, int audio_channels,
-                             int audio_samplerate, bool audio_passthru);
+    virtual void Reconfigure(int audio_bits, 
+                             int audio_channels, 
+                             int audio_samplerate,
+                             bool audio_passthru,
+                             void* audio_codec = NULL);
     
     // do AddSamples calls block?
     virtual void SetBlocking(bool blocking);
@@ -45 +58 @@
     virtual void SetEffDsp(int dsprate);
 
     virtual void SetStretchFactor(float factor);
+    virtual float GetStretchFactor(void);
 
     virtual void Reset(void);
 
@@ -127 +141 @@
     bool audio_passthru;
 
     float audio_stretchfactor;
+    AVCodecContext *audio_codec;
     AudioOutputSource source;
 
     bool killaudio;
@@ -135 +150 @@
     bool set_initial_vol;
     bool buffer_output_data_for_use; //  used by AudioOutputNULL
     
+    int configured_audio_channels;
+
  private:
     // resampler
     bool need_resampler;
@@ -145 +162 @@
     short tmp_buff[AUDIO_TMP_BUF_SIZE];
 
     // timestretch
-    soundtouch::SoundTouch * pSoundStretch;
+    soundtouch::SoundTouch    *pSoundStretch;
+    AudioOutputDigitalEncoder *encoder;
+    FreeSurround              *upmixer;
 
+    int source_audio_channels;
+    int source_audio_bytes_per_sample;
+    bool needs_upmix;
+    int surround_mode;
+
     bool blocking; // do AddSamples calls block?
 
     int lastaudiolen;
@@ -164 +188 @@
 
     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
-                   //                                          timecodes) ...
+    /// timecode of audio leaving the soundcard (same units as timecodes)
+    long long audiotime;
     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
-                   buffer */
+    /// timecode of audio most recently placed into buffer
+    long long audbuf_timecode;
 
     int numlowbuffer;
 

libs/libmyth/audiooutputbase.cpp

@@ -15 +15 @@
 
 // MythTV headers
 #include "audiooutputbase.h"
+#include "audiooutputdigitalencoder.h"
+#include "SoundTouch.h"
+#include "freesurround.h"
 #include "compat.h"
 
 #define LOC QString("AO: ")
@@ -36 +39 @@
     audio_passthru_device(QDeepCopy<QString>(laudio_passthru_device)),
     audio_passthru(false),      audio_stretchfactor(1.0f),
 
+    audio_codec(NULL),
     source(lsource),            killaudio(false),
 
     pauseaudio(false),          audio_actually_paused(false),
@@ -47 +51 @@
 
     src_ctx(NULL),
 
-    pSoundStretch(NULL),        blocking(false),
+    pSoundStretch(NULL),        
+    encoder(NULL),
+    upmixer(NULL),
+    source_audio_channels(-1),
+    source_audio_bytes_per_sample(0),
+    needs_upmix(false),
+    surround_mode(FreeSurround::SurroundModePassive),
 
+    blocking(false),
+
     lastaudiolen(0),            samples_buffered(0),
 
     audio_thread_exists(false),
@@ -71 +83 @@
     memset(tmp_buff,           0, sizeof(short) * AUDIO_TMP_BUF_SIZE);
     memset(&audiotime_updated, 0, sizeof(audiotime_updated));
     memset(audiobuffer,        0, sizeof(char)  * AUDBUFSIZE);
+    configured_audio_channels = gContext->GetNumSetting("MaxChannels", 2);
 
     // You need to call Reconfigure from your concrete class.
     // Reconfigure(laudio_bits,       laudio_channels,
@@ -111 +124 @@
             VERBOSE(VB_GENERAL, LOC + 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 AudioOutputDigitalEncoder();
+                    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);
 
@@ -134 +179 @@
     pthread_mutex_unlock(&audio_buflock);
 }
 
+float AudioOutputBase::GetStretchFactor()
+{
+    return audio_stretchfactor;
+}
+
 void AudioOutputBase::Reconfigure(int laudio_bits, int laudio_channels, 
-                                 int laudio_samplerate, bool laudio_passthru)
+                                 int laudio_samplerate, bool laudio_passthru,
+                                 void* laudio_codec)
 {
-    if (laudio_bits == audio_bits && laudio_channels == audio_channels &&
-        laudio_samplerate == audio_samplerate &&
-        laudio_passthru == audio_passthru && !need_resampler)
+    int codec_id = CODEC_ID_NONE;
+    int lcodec_id = CODEC_ID_NONE;
+    int lcchannels = 0;
+    int cchannels = 0;
+    int lsource_audio_channels = laudio_channels;
+    bool lneeds_upmix = false;
+
+    if (laudio_codec)
+    {
+        lcodec_id = ((AVCodecContext*)laudio_codec)->codec_id;
+        laudio_bits = 16;
+        laudio_channels = 2;
+        lsource_audio_channels = laudio_channels;
+        laudio_samplerate = 48000;
+        lcchannels = ((AVCodecContext*)laudio_codec)->channels;
+    }
+
+    if (audio_codec)
+    {
+        codec_id = audio_codec->codec_id;
+        cchannels = ((AVCodecContext*)audio_codec)->channels;
+    }
+
+    if ((configured_audio_channels == 6) && 
+        !(laudio_codec || audio_codec))
+    {
+        laudio_channels = configured_audio_channels;
+        lneeds_upmix = true;
+        VERBOSE(VB_AUDIO,LOC + "Needs upmix");
+    }
+
+    ClearError();
+    bool general_deps = (laudio_bits == audio_bits && 
+        laudio_channels == audio_channels &&
+        laudio_samplerate == audio_samplerate && !need_resampler &&
+        laudio_passthru == audio_passthru &&
+        lneeds_upmix == needs_upmix &&
+        lcodec_id == codec_id && lcchannels == cchannels);
+    bool upmix_deps =
+        (lsource_audio_channels == source_audio_channels);
+    if (general_deps && upmix_deps)
+    {
+        VERBOSE(VB_AUDIO,LOC + "no change exiting");
         return;
+    }
 
+    if (general_deps && !upmix_deps && lneeds_upmix && upmixer)
+    {
+        upmixer->flush();
+        source_audio_channels = lsource_audio_channels;
+        VERBOSE(VB_AUDIO,LOC + QString("source channels changed to %1").arg(source_audio_channels));
+        return;
+    }
+
     KillAudio();
     
     pthread_mutex_lock(&audio_buflock);
@@ -151 +251 @@
     waud = raud = 0;
     audio_actually_paused = false;
     
+    bool redo_stretch = (pSoundStretch && audio_channels != laudio_channels);
     audio_channels = laudio_channels;
+    source_audio_channels = lsource_audio_channels;
     audio_bits = laudio_bits;
     audio_samplerate = laudio_samplerate;
+    audio_codec = (AVCodecContext*)laudio_codec;
     audio_passthru = laudio_passthru;
+    needs_upmix = lneeds_upmix;
+
     if (audio_bits != 8 && audio_bits != 16)
     {
         pthread_mutex_unlock(&avsync_lock);
@@ -162 +267 @@
         Error("AudioOutput only supports 8 or 16bit audio.");
         return;
     }
+
     audio_bytes_per_sample = audio_channels * audio_bits / 8;
+    source_audio_bytes_per_sample = source_audio_channels * audio_bits / 8;
     
     need_resampler = false;
     killaudio = false;
@@ -172 +279 @@
     
     numlowbuffer = 0;
 
+    VERBOSE(VB_GENERAL, QString("Opening audio device '%1'. ch %2(%3) sr %4")
+            .arg(audio_main_device).arg(audio_channels)
+            .arg(source_audio_channels).arg(audio_samplerate));
+ 
     // Actually do the device specific open call
     if (!OpenDevice())
     {
         VERBOSE(VB_AUDIO, LOC_ERR + "Aborting reconfigure");
         pthread_mutex_unlock(&avsync_lock);
         pthread_mutex_unlock(&audio_buflock);
+        if (GetError().isEmpty())
+            Error("Aborting reconfigure");
+        VERBOSE(VB_AUDIO, "Aborting reconfigure");
         return;
     }
 
@@ -200 +314 @@
     current_seconds = -1;
     source_bitrate = -1;
 
+    // NOTE: this won't do anything as above samplerate vars are set equal
     // Check if we need the resampler
     if (audio_samplerate != laudio_samplerate)
     {
@@ -222 +337 @@
         need_resampler = true;
     }
 
+    if (needs_upmix)
+    {
+        VERBOSE(VB_AUDIO, LOC + QString("create upmixer"));
+        if (configured_audio_channels == 6)
+        {
+            surround_mode = gContext->GetNumSetting("AudioUpmixType", 2);
+        }
+
+        upmixer = new FreeSurround(
+            audio_samplerate, 
+            source == AUDIOOUTPUT_VIDEO, 
+            (FreeSurround::SurroundMode)surround_mode);
+
+        VERBOSE(VB_AUDIO, LOC +
+                QString("create upmixer done with surround mode %1")
+                .arg(surround_mode));
+    }
+
     VERBOSE(VB_AUDIO, LOC + 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)
+    if (redo_stretch)
     {
-        pSoundStretch->setSampleRate(audio_samplerate);
-        pSoundStretch->setChannels(audio_channels);
+        float laudio_stretchfactor = audio_stretchfactor;
+        delete pSoundStretch;
+        pSoundStretch = NULL;
+        audio_stretchfactor = 0.0f;
+        SetStretchFactorLocked(laudio_stretchfactor);
     }
+    else
+    {
+        SetStretchFactorLocked(audio_stretchfactor);
+        if (pSoundStretch)
+        {
+            // 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 AudioOutputDigitalEncoder();
+                    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 + "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
     prepareVisuals();
 
@@ -290 +468 @@
         pSoundStretch = NULL;
     }
 
+    if (encoder)
+    {
+        delete encoder;
+        encoder = NULL;
+    }
+
+    if (upmixer)
+    {
+        delete upmixer;
+        upmixer = NULL;
+    }
+    needs_upmix = false;
+
     CloseDevice();
 
     killAudioLock.unlock();
@@ -303 +494 @@
 
 void AudioOutputBase::Pause(bool paused)
 {
+    VERBOSE(VB_AUDIO, LOC + QString("Pause %0").arg(paused));
     pauseaudio = paused;
     audio_actually_paused = false;
 }
@@ -385 +577 @@
        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)
@@ -397 +589 @@
 
     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)
@@ -439 +642 @@
     // include algorithmic latencies
     if (pSoundStretch)
     {
+        // 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);
     }
-               
+
+    if (upmixer && needs_upmix)
+    {
+        totalbuffer += upmixer->sampleLatency() * audio_bytes_per_sample;
+    }
+
     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);
@@ -458 +684 @@
 {
     // NOTE: This function is not threadsafe
     int afree = audiofree(true);
-    int abps = audio_bytes_per_sample;
+    int abps = (encoder) ?
+        encoder->audio_bytes_per_sample : audio_bytes_per_sample;
     int len = samples * abps;
 
     // Check we have enough space to write the data
     if (need_resampler && src_ctx)
         len = (int)ceilf(float(len) * src_data.src_ratio);
 
+    // include samples in upmix buffer that may be flushed
+    if (needs_upmix && upmixer)
+        len += upmixer->numUnprocessedSamples() * abps;
+
     if (pSoundStretch)
         len += (pSoundStretch->numUnprocessedSamples() +
                 (int)(pSoundStretch->numSamples()/audio_stretchfactor))*abps;
@@ -520 +751 @@
     // NOTE: This function is not threadsafe
 
     int afree = audiofree(true);
-    int abps = audio_bytes_per_sample;
+    int abps = (encoder) ?
+        encoder->audio_bytes_per_sample : audio_bytes_per_sample;
     int len = samples * abps;
 
     // Check we have enough space to write the data
     if (need_resampler && src_ctx)
         len = (int)ceilf(float(len) * src_data.src_ratio);
 
+    // include samples in upmix buffer that may be flushed
+    if (needs_upmix && upmixer)
+        len += upmixer->numUnprocessedSamples() * abps;
+ 
     if (pSoundStretch)
     {
         len += (pSoundStretch->numUnprocessedSamples() +
@@ -575 +811 @@
 
 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, LOC + "Waiting for free space " +
+            VERBOSE(VB_AUDIO|VB_TIMESTAMP, LOC + "Waiting for free space " +
                     QString("(need %1, available %2)").arg(len).arg(afree));
 
             // wait for more space
@@ -591 +829 @@
         }
         else
         {
-            VERBOSE(VB_IMPORTANT, LOC_ERR +
-                    "Audio buffer overflow, audio data lost!");
-            samples = afree / audio_bytes_per_sample;
-            len = samples * audio_bytes_per_sample;
+            VERBOSE(VB_IMPORTANT, LOC_ERR + 
+                    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);
@@ -619 +858 @@
     
     int afree = audiofree(false);
 
-    VERBOSE(VB_AUDIO|VB_TIMESTAMP,
-            LOC + 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, 
+            LOC + QString("_AddSamples samples=%1 bytes=%2, used=%3, "
+                          "free=%4, timecode=%5 needsupmix %6")
+            .arg(samples)
+            .arg(samples * abps)
+            .arg(AUDBUFSIZE-afree).arg(afree).arg(timecode)
+            .arg(needs_upmix));
     
-    len = WaitForFreeSpace(samples);
-
-    if (interleaved) 
+    if (upmixer && needs_upmix)
     {
-        char *mybuf = (char*)buffer;
-        int bdiff = AUDBUFSIZE - org_waud;
-        if (bdiff < len)
+        int out_samples = 0;
+        int step = (interleaved)?source_audio_channels:1;
+        len = WaitForFreeSpace(samples);    // test
+        for (int itemp = 0; itemp < samples; )
         {
-            memcpy(audiobuffer + org_waud, mybuf, bdiff);
-            memcpy(audiobuffer, mybuf + bdiff, len - bdiff);
+            // just in case it does a processing cycle, release the lock
+            // to allow the output loop to do output
+            pthread_mutex_unlock(&audio_buflock);
+            if (audio_bytes == 2)
+            {
+                itemp += upmixer->putSamples(
+                    (short*)buffer + itemp * step,
+                    samples - itemp,
+                    source_audio_channels,
+                    (interleaved) ? 0 : samples);
+            }
+            else
+            {
+                itemp += upmixer->putSamples(
+                    (char*)buffer + itemp * step,
+                    samples - itemp,
+                    source_audio_channels,
+                    (interleaved) ? 0 : samples);
+            }
+            pthread_mutex_lock(&audio_buflock);
+
+            int copy_samples = upmixer->numSamples();
+            if (copy_samples)
+            {
+                int copy_len = copy_samples * abps;
+                out_samples += copy_samples;
+                if (out_samples > samples)
+                    len = WaitForFreeSpace(out_samples);
+                int bdiff = AUDBUFSIZE - org_waud;
+                if (bdiff < copy_len) 
+                {
+                    int bdiff_samples = bdiff/abps;
+                    upmixer->receiveSamples(
+                        (short*)(audiobuffer + org_waud), bdiff_samples);
+                    upmixer->receiveSamples(
+                        (short*)(audiobuffer), (copy_samples - bdiff_samples));
+                }
+                else
+                {
+                    upmixer->receiveSamples(
+                        (short*)(audiobuffer + org_waud), copy_samples);
+                }
+                org_waud = (org_waud + copy_len) % AUDBUFSIZE;
+            }
         }
-        else
-            memcpy(audiobuffer + org_waud, mybuf, len);
- 
-        org_waud = (org_waud + len) % AUDBUFSIZE;
-    } 
-    else 
+
+        if (samples > 0)
+            len = WaitForFreeSpace(out_samples);
+
+        samples = out_samples;
+    }
+    else
     {
-        char **mybuf = (char**)buffer;
-        for (int itemp = 0; itemp < samples * audio_bytes; itemp += audio_bytes)
+        len = WaitForFreeSpace(samples);
+
+        if (interleaved) 
         {
-            for (int chan = 0; chan < audio_channels; chan++)
+            char *mybuf = (char*)buffer;
+            int bdiff = AUDBUFSIZE - org_waud;
+            if (bdiff < len)
             {
-                audiobuffer[org_waud++] = mybuf[chan][itemp];
-                if (audio_bits == 16)
-                    audiobuffer[org_waud++] = mybuf[chan][itemp+1];
+                memcpy(audiobuffer + org_waud, mybuf, bdiff);
+                memcpy(audiobuffer, mybuf + bdiff, len - bdiff);
+            }
+            else
+            {
+                memcpy(audiobuffer + org_waud, mybuf, len);
+            }
+     
+            org_waud = (org_waud + len) % AUDBUFSIZE;
+        } 
+        else 
+        {
+            char **mybuf = (char**)buffer;
+            for (int itemp = 0; itemp < samples * audio_bytes;
+                 itemp += audio_bytes)
+            {
+                for (int chan = 0; chan < audio_channels; chan++)
+                {
+                    audiobuffer[org_waud++] = mybuf[chan][itemp];
+                    if (audio_bits == 16)
+                        audiobuffer[org_waud++] = mybuf[chan][itemp+1];
 
-                if (org_waud >= AUDBUFSIZE)
-                    org_waud -= AUDBUFSIZE;
+                    if (org_waud >= AUDBUFSIZE)
+                        org_waud -= AUDBUFSIZE;
+                }
             }
         }
     }
 
-    if (pSoundStretch)
+    if (samples > 0)
     {
-        // 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;
-        if (bdiff < len)
+        if (pSoundStretch)
         {
-            pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)(audiobuffer + 
-                                      org_waud), nSamplesToEnd);
-            pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer,
-                                      (len - bdiff) / audio_bytes_per_sample);
-        }
-        else
-        {
-            pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)(audiobuffer + 
-                                      org_waud), len / audio_bytes_per_sample);
-        }
 
-        int newLen = 0;
-        int nSamples;
-        len = WaitForFreeSpace(pSoundStretch->numSamples() * 
-                               audio_bytes_per_sample);
-        do 
-        {
-            int samplesToGet = len/audio_bytes_per_sample;
-            if (samplesToGet > nSamplesToEnd)
+            // 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/abps;
+            if (bdiff < len)
             {
-                samplesToGet = nSamplesToEnd;    
+                pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)
+                                          (audiobuffer + 
+                                           org_waud), nSamplesToEnd);
+                pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer,
+                                          (len - bdiff) / abps);
             }
+            else
+            {
+                pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)
+                                          (audiobuffer + org_waud),
+                                          len / abps);
+            }
 
-            nSamples = pSoundStretch->receiveSamples((soundtouch::SAMPLETYPE*)
-                                      (audiobuffer + org_waud), samplesToGet);
-            if (nSamples == nSamplesToEnd)
+            if (encoder)
             {
-                org_waud = 0;
-                nSamplesToEnd = AUDBUFSIZE/audio_bytes_per_sample;
+                // pull out a packet's worth and reencode it until we
+                // don't 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)
+                {
+                    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)
+                        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;
+                }
             }
             else
             {
-                org_waud += nSamples * audio_bytes_per_sample;
-                nSamplesToEnd -= nSamples;
+                int newLen = 0;
+                int nSamples;
+                len = WaitForFreeSpace(pSoundStretch->numSamples() * 
+                                       audio_bytes_per_sample);
+                do 
+                {
+                    int samplesToGet = len/audio_bytes_per_sample;
+                    if (samplesToGet > nSamplesToEnd)
+                    {
+                        samplesToGet = nSamplesToEnd;    
+                    }
+
+                    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);
             }
+        }
 
-            newLen += nSamples * audio_bytes_per_sample;
-            len -= nSamples * audio_bytes_per_sample;
-        } while (nSamples > 0);
-    }
+        waud = org_waud;
+        lastaudiolen = audiolen(false);
 
-    waud = org_waud;
-    lastaudiolen = audiolen(false);
+        if (timecode < 0)
+        {
+            // mythmusic doesn't give timestamps..
+            timecode = (int)((samples_buffered * 100000.0) / effdsp);
+        }
+        
+        samples_buffered += samples;
+        
+        /* we want the time at the end -- but the file format stores
+           time at the start of the chunk. */
+        // even with timestretch, timecode is still calculated from original
+        // sample count
+        audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp);
 
-    samples_buffered += samples;
-    
-    if (timecode < 0)
-    {
-        // mythmusic doesn't give timestamps..
-        timecode = (int)((samples_buffered * 100000.0) / effdsp);
+        if (interleaved)
+        {
+            dispatchVisual((unsigned char *)buffer, len, timecode,
+                           source_audio_channels, audio_bits);
+        }
     }
-    
-    /* we want the time at the end -- but the file format stores
-       time at the start of the chunk. */
-    // even with timestretch, timecode is still calculated from original
-    // sample count
-    audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp);
 
-    if (interleaved)
-        dispatchVisual((unsigned char *)buffer, len, timecode, audio_channels, audio_bits);
-
     pthread_mutex_unlock(&audio_buflock);
 }
 
@@ -739 +1108 @@
 
     if (source_bitrate == -1)
     {
-        source_bitrate = audio_samplerate * audio_channels * audio_bits;
+        source_bitrate = audio_samplerate * source_audio_channels * audio_bits;
     }
 
     if (ct / 1000 != current_seconds) 
@@ -747 +1116 @@
         current_seconds = ct / 1000;
         OutputEvent e(current_seconds, ct,
                       source_bitrate, audio_samplerate, audio_bits, 
-                      audio_channels);
+                      source_audio_channels);
         dispatch(e);
     }
 }
@@ -785 +1154 @@
 
             space_on_soundcard = getSpaceOnSoundcard();
 
-            if (space_on_soundcard != last_space_on_soundcard) {
-                VERBOSE(VB_AUDIO, LOC + QString("%1 bytes free on soundcard")
+            if (space_on_soundcard != last_space_on_soundcard)
+            {
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP,
+                        LOC + QString("%1 bytes free on soundcard")
                         .arg(space_on_soundcard));
+
                 last_space_on_soundcard = space_on_soundcard;
             }
 
@@ -799 +1171 @@
                     WriteAudio(zeros, fragment_size);
                 } else {
                     // this should never happen now -dag
-                    VERBOSE(VB_AUDIO, LOC +
+                    VERBOSE(VB_AUDIO|VB_TIMESTAMP, LOC + 
                             QString("waiting for space on soundcard "
                                     "to write zeros: have %1 need %2")
                             .arg(space_on_soundcard).arg(fragment_size));
@@ -835 +1207 @@
         if (fragment_size > audiolen(true))
         {
             if (audiolen(true) > 0)  // only log if we're sending some audio
-                VERBOSE(VB_AUDIO, LOC +
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP, LOC +
                         QString("audio waiting for buffer to fill: "
                                 "have %1 want %2")
                         .arg(audiolen(true)).arg(fragment_size));
 
-            VERBOSE(VB_AUDIO, LOC + "Broadcasting free space avail");
+            //VERBOSE(VB_AUDIO|VB_TIMESTAMP,
+            //LOC + "Broadcasting free space avail");
             pthread_mutex_lock(&audio_buflock);
             pthread_cond_broadcast(&audio_bufsig);
             pthread_mutex_unlock(&audio_buflock);
@@ -854 +1227 @@
         if (fragment_size > space_on_soundcard)
         {
             if (space_on_soundcard != last_space_on_soundcard) {
-                VERBOSE(VB_AUDIO, LOC +
+                VERBOSE(VB_AUDIO|VB_TIMESTAMP, LOC +
                         QString("audio waiting for space on soundcard: "
                                 "have %1 need %2")
                         .arg(space_on_soundcard).arg(fragment_size));
@@ -916 +1289 @@
 
         /* update raud */
         raud = (raud + fragment_size) % AUDBUFSIZE;
-        VERBOSE(VB_AUDIO, LOC + "Broadcasting free space avail");
+        //VERBOSE(VB_AUDIO|VB_TIMESTAMP, LOC + "Broadcasting free space avail");
         pthread_cond_broadcast(&audio_bufsig);
 
         written_size = fragment_size;

libs/libmyth/audiooutputalsa.cpp

@@ -89 +89 @@
     }
     else
     {
-        fragment_size = 6144; // nicely divisible by 2,4,6,8 channels @ 16-bits
-        buffer_time = 500000;  // .5 seconds
+        fragment_size =
+            (audio_bits * audio_channels * audio_samplerate) / (8*30);
+        buffer_time = 100000;
         period_time = buffer_time / 4;  // 4 interrupts per buffer
     }
 
@@ -162 +163 @@
     
     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/libmythfreesurround/el_processor.cpp

@@ -20 +20 @@
 #include <complex>
 #include <cmath>
 #include <vector>
+#ifdef USE_FFTW3
 #include "fftw3.h"
+#else
+extern "C" {
+#include "dsputil.h"
+};
+typedef FFTSample FFTComplexArray[2];
+#endif
 
 #define FILTERED_LFE
 
+#ifdef USE_FFTW3
 #pragma comment (lib,"libfftw3f-3.lib")
+#endif
 
 typedef std::complex<float> cfloat;
 
@@ -45 +54 @@
     // create an instance of the decoder
     //  blocksize is fixed over the lifetime of this object for performance reasons
     decoder_impl(unsigned blocksize=8192): N(blocksize), halfN(blocksize/2) {
+#ifdef USE_FFTW3
         // create FFTW buffers
         lt = (float*)fftwf_malloc(sizeof(float)*N);
         rt = (float*)fftwf_malloc(sizeof(float)*N);
@@ -55 +65 @@
         loadL = fftwf_plan_dft_r2c_1d(N, lt, dftL,FFTW_MEASURE);
         loadR = fftwf_plan_dft_r2c_1d(N, rt, dftR,FFTW_MEASURE);
         store = fftwf_plan_dft_c2r_1d(N, src, dst,FFTW_MEASURE);    
+#else
+        // create lavc fft buffers
+        lt = (float*)malloc(sizeof(FFTSample)*N);
+        rt = (float*)malloc(sizeof(FFTSample)*N);
+        dst = (float*)malloc(sizeof(FFTSample)*N);
+        dftL = (FFTComplexArray*)malloc(sizeof(FFTComplex)*N);
+        dftR = (FFTComplexArray*)malloc(sizeof(FFTComplex)*N);
+        src = (FFTComplexArray*)malloc(sizeof(FFTComplex)*N);
+        fftContextForward = (FFTContext*)malloc(sizeof(FFTContext));
+        memset(fftContextForward, 0, sizeof(FFTContext));
+        fftContextReverse = (FFTContext*)malloc(sizeof(FFTContext));
+        memset(fftContextReverse, 0, sizeof(FFTContext));
+        ff_fft_init(fftContextForward, 13, 0);
+        ff_fft_init(fftContextReverse, 13, 1);
+#endif
         // resize our own buffers
         frontR.resize(N);
         frontL.resize(N);
@@ -97 +122 @@
 
     // destructor
     ~decoder_impl() {
+#ifdef USE_FFTW3
         // clean up the FFTW stuff
         fftwf_destroy_plan(store);
         fftwf_destroy_plan(loadR);
@@ -107 +133 @@
         fftwf_free(dst);
         fftwf_free(rt);
         fftwf_free(lt);
+#else
+        ff_fft_end(fftContextForward);
+        ff_fft_end(fftContextReverse);
+        free(src); 
+        free(dftR);
+        free(dftL);
+        free(dst);
+        free(rt);
+        free(lt);
+        free(fftContextForward); 
+        free(fftContextReverse); 
+#endif
     }
 
     float ** getInputBuffers()
@@ -237 +275 @@
             }
         }
 
+#ifdef USE_FFTW3
         // ... and tranform it into the frequency domain
         fftwf_execute(loadL);
         fftwf_execute(loadR);
+#else
+        ff_fft_permuteRC(fftContextForward, &lt[0], (FFTComplex*)&dftL[0]);
+        ff_fft_calc(fftContextForward, (FFTComplex*)&dftL[0]);
+        ff_fft_permuteRC(fftContextForward, &rt[0], (FFTComplex*)&dftR[0]);
+        ff_fft_calc(fftContextForward, (FFTComplex*)&dftR[0]);
+#endif
 
         // 2. compare amplitude and phase of each DFT bin and produce the X/Y coordinates in the sound field
         //    but dont do DC or N/2 component
@@ -447 +492 @@
     // filter the complex source signal and add it to target
     void apply_filter(cfloat *signal, float *flt, float *target) {
         // filter the signal
-        for (unsigned f=0;f<=halfN;f++) {       
+        unsigned f;
+        for (f=0;f<=halfN;f++) {
             src[f][0] = signal[f].real() * flt[f];
             src[f][1] = signal[f].imag() * flt[f];
         }
+#ifdef USE_FFTW3
         // transform into time domain
         fftwf_execute(store);
+#else
+        // enforce odd symmetry
+        for (f=1;f<halfN-1;f++) {
+            src[N-f][0] = src[f][0];
+            src[N-f][1] = -src[f][1];   // complex conjugate
+        }
+        ff_fft_calc(fftContextReverse, (FFTComplex*)&src[0]);
+        ff_fft_permuteCR(fftContextReverse, (FFTComplex*)&src[0], &dst[0]);
+#endif
 
         float* pT1   = &target[current_buf*halfN];
         float* pWnd1 = &wnd[0];
@@ -470 +526 @@
         }
     }
 
+#ifndef USE_FFTW3
+    /**
+     *  * Do the permutation needed BEFORE calling ff_fft_calc()
+     *  special for freesurround that also copies
+     *   */
+    void ff_fft_permuteRC(FFTContext *s, FFTSample *r, FFTComplex *z)
+    {
+        int j, k, np;
+        FFTComplex tmp;
+        const uint16_t *revtab = s->revtab;
+
+        /* reverse */
+        np = 1 << s->nbits;
+        for(j=0;j<np;j++) {
+            k = revtab[j];
+            if (k < j) {
+                z[k].re = r[j];
+                z[k].im = 0.0;
+                z[j].re = r[k];
+                z[j].im = 0.0;
+            }
+        }
+    }
+
+    /**
+     *  * Do the permutation needed BEFORE calling ff_fft_calc()
+     *  special for freesurround that also copies and 
+     *  discards im component as it should be 0
+     *   */
+    void ff_fft_permuteCR(FFTContext *s, FFTComplex *z, FFTSample *r)
+    {
+        int j, k, np;
+        FFTComplex tmp;
+        const uint16_t *revtab = s->revtab;
+
+        /* reverse */
+        np = 1 << s->nbits;
+        for(j=0;j<np;j++) {
+            k = revtab[j];
+            if (k < j) {
+                r[k] = z[j].re;
+                r[j] = z[k].re;
+            }
+        }
+    }
+#endif
+
     unsigned int N;                    // the block size
     unsigned int halfN;                // half block size precalculated
+#ifdef USE_FFTW3
     // FFTW data structures
     float *lt,*rt,*dst;                // left total, right total (source arrays), destination array
     fftwf_complex *dftL,*dftR,*src;    // intermediate arrays (FFTs of lt & rt, processing source)
     fftwf_plan loadL,loadR,store;      // plans for loading the data into the intermediate format and back
+#else
+    FFTContext *fftContextForward, *fftContextReverse; 
+    FFTSample *lt,*rt,*dst;            // left total, right total (source arrays), destination array
+    FFTComplexArray *dftL,*dftR,*src;  // intermediate arrays (FFTs of lt & rt, processing source)
+#endif
     // buffers
     std::vector<cfloat> frontL,frontR,avg,surL,surR; // the signal (phase-corrected) in the frequency domain
 #ifdef FILTERED_LFE

libs/libmythfreesurround/libmythfreesurround.pro

@@ -19 +19 @@
 SOURCES += el_processor.cpp
 SOURCES += freesurround.cpp
 
-#required until its rewritten to use avcodec fft lib
-#LIBS += -lfftw3
-LIBS += -lfftw3f
-
+contains( CONFIG_LIBFFTW3, yes ) {
+    #required until its rewritten to use avcodec fft lib
+    LIBS += -lfftw3f
+    DEFINES += USE_FFTW3
+} else {
+    #required until its rewritten to use avcodec fft lib
+    DEPENDPATH += ../libavcodec
+    LIBS += -L../libavcodec -lavcodec
+    INCLUDEPATH += ../../libs/libavutil
+}

programs/mythfrontend/globalsettings.cpp

@@ -56 +56 @@
     }
 #endif
 #ifdef USING_ALSA
-    gc->addSelection("ALSA:default", "ALSA:default");
+    gc->addSelection("ALSA:default",       "ALSA:default");
+    gc->addSelection("ALSA:surround51",    "ALSA:surround51");
+    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 USING_ARTS
     gc->addSelection("ARTS:", "ARTS:");
@@ -78 +83 @@
     return gc;
 }
 
+static HostComboBox *MaxAudioChannels()
+{
+    HostComboBox *gc = new HostComboBox("MaxChannels",false);
+    gc->setLabel(QObject::tr("Max Audio Channels"));
+    gc->addSelection(QObject::tr("Stereo"), "2", true); // default
+    gc->addSelection(QObject::tr("5.1"), "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 HostComboBox *AudioUpmixType()
+{
+    HostComboBox *gc = new HostComboBox("AudioUpmixType",false);
+    gc->setLabel(QObject::tr("Upmix"));
+    gc->addSelection(QObject::tr("Passive"), "0");
+    gc->addSelection(QObject::tr("Active Simple"), "1");
+    gc->addSelection(QObject::tr("Active Linear"), "2", true); // default
+    gc->setHelpText(
+            QObject::tr(
+                "Set the audio upmix type for 2ch to 6ch conversion. "
+                "This is for multi-channel/surround audio playback."));
+    return gc;
+}
+
 static HostComboBox *PassThroughOutputDevice()
 {
     HostComboBox *gc = new HostComboBox("PassThruOutputDevice", true);
@@ -3202 +3234 @@
          vgrp0->addChild(AC3PassThrough());
          vgrp0->addChild(DTSPassThrough());
 
+         HorizontalConfigurationGroup *agrp =
+             new HorizontalConfigurationGroup(false, false, true, true);
+         agrp->addChild(MaxAudioChannels());
+         agrp->addChild(AudioUpmixType());
+         addChild(agrp);
+
          VerticalConfigurationGroup *vgrp1 =
              new VerticalConfigurationGroup(false, false, true, true);
          vgrp1->addChild(AggressiveBuffer());

programs/mythtranscode/transcode.cpp

@@ -55 +55 @@
 
     // reconfigure sound out for new params
     virtual void Reconfigure(int audio_bits, int audio_channels,
-                             int audio_samplerate, bool audio_passthru)
+                             int audio_samplerate, bool audio_passthru,
+                             void *audio_codec = NULL)
     {
+        ClearError();
         (void)audio_samplerate;
         (void)audio_passthru;
+        (void)audio_codec;
         bits = audio_bits;
         channels = audio_channels;
         bytes_per_sample = bits * channels / 8;
+        if ((uint)audio_channels > 2)
+            Error(QString("Invalid channel count %1").arg(channels));
     }
 
     // dsprate is in 100 * samples/second

libs/libmythtv/avformatdecoder.h

@@ -261 +261 @@
     bool              allow_ac3_passthru;
     bool              allow_dts_passthru;
     bool              disable_passthru;
+    uint              max_channels;
+
     VideoFrame       *dummy_frame;
 
     AudioInfo         audioIn;

libs/libmythtv/avformatdecoder.cpp

@@ -51 +51 @@
 
 #define MAX_AC3_FRAME_SIZE 6144
 
-/** Set to zero to allow any number of AC3 channels. */
-#define MAX_OUTPUT_CHANNELS 2
-
 static int cc608_parity(uint8_t byte);
 static int cc608_good_parity(const int *parity_table, uint16_t data);
 static void cc608_build_parity_table(int *parity_table);
@@ -400 +397 @@
       // Audio
       audioSamples(new short int[AVCODEC_MAX_AUDIO_FRAME_SIZE]),
       allow_ac3_passthru(false),    allow_dts_passthru(false),
-      disable_passthru(false),      dummy_frame(NULL),
+      disable_passthru(false),      max_channels(2),
+      dummy_frame(NULL),
       // DVD
       lastdvdtitle(-1), lastcellstart(0),
       dvdmenupktseen(false), indvdstill(false),
@@ -417 +415 @@
 
     allow_ac3_passthru = gContext->GetNumSetting("AC3PassThru", false);
     allow_dts_passthru = gContext->GetNumSetting("DTSPassThru", false);
+    max_channels = (uint) gContext->GetNumSetting("MaxChannels", 2);
 
     audioIn.sample_size = -32; // force SetupAudioStream to run once
     itv = GetNVP()->GetInteractiveTV();
@@ -1587 +1586 @@
                             <<") already open, leaving it alone.");
                 }
                 //assert(enc->codec_id);
+                VERBOSE(VB_GENERAL, LOC + QString("codec %1 has %2 channels")
+                        .arg(codec_id_string(enc->codec_id))
+                        .arg(enc->channels));
 
+#if 0
+                // HACK MULTICHANNEL DTS passthru disabled for multichannel,
+                // dont know how to handle this
                 // HACK BEGIN REALLY UGLY HACK FOR DTS PASSTHRU
                 if (enc->codec_id == CODEC_ID_DTS)
                 {
@@ -1596 +1601 @@
                     // enc->bit_rate = what??;
                 }
                 // HACK END REALLY UGLY HACK FOR DTS PASSTHRU
+#endif
 
                 bitrate += enc->bit_rate;
                 break;
@@ -3296 +3302 @@
 
                     // detect channels on streams that need
                     // to be decoded before we can know this
+                    int prev_channels = curstream->codec->channels;
+                    bool already_decoded = false;
                     if (!curstream->codec->channels)
                     {
                         QMutexLocker locker(&avcodeclock);
-                        curstream->codec->channels = MAX_OUTPUT_CHANNELS;
+                        VERBOSE(VB_IMPORTANT, LOC +
+                                QString("Setting channels to %1")
+                                .arg(audioOut.channels));
+
+                        bool do_ac3_passthru = (allow_ac3_passthru && !transcoding &&
+                                                (curstream->codec->codec_id == CODEC_ID_AC3));
+                        bool do_dts_passthru = (allow_dts_passthru && !transcoding &&
+                                                (curstream->codec->codec_id == CODEC_ID_DTS));
+                        bool using_passthru = do_ac3_passthru || do_dts_passthru;
+                        if (using_passthru)
+                        {
+                            // for passthru let it select the max number of channels
+                            curstream->codec->channels = 0;
+                            curstream->codec->request_channels = 0;
+                        }
+                        else
+                        {
+                            curstream->codec->channels = audioOut.channels;
+                            curstream->codec->request_channels = audioOut.channels;
+                        }
                         ret = avcodec_decode_audio(
                             curstream->codec, audioSamples,
                             &data_size, ptr, len);
+                        already_decoded = true;
 
                         reselectAudioTrack |= curstream->codec->channels;
                     }
@@ -3360 +3388 @@
                         AVCodecContext *ctx = curstream->codec;
 
                         if ((ctx->channels == 0) ||
-                            (ctx->channels > MAX_OUTPUT_CHANNELS))
-                            ctx->channels = MAX_OUTPUT_CHANNELS;
+                            (ctx->channels > audioOut.channels))
+                            ctx->channels = audioOut.channels;
 
-                        ret = avcodec_decode_audio(
-                            ctx, audioSamples, &data_size, ptr, len);
+                        if (!already_decoded)
+                        {
+                            curstream->codec->request_channels = audioOut.channels;
+                            ret = avcodec_decode_audio(
+                                ctx, audioSamples, &data_size, ptr, len);
+                        }
 
                         // When decoding some audio streams the number of
                         // channels, etc isn't known until we try decoding it.
@@ -3799 +3831 @@
 
 void AvFormatDecoder::SetDisablePassThrough(bool disable)
 {
+    // can only disable never reenable as once
+    // timestretch is on its on for the session
+    if (disable_passthru)
+        return;
+
     if (selectedTrack[kTrackTypeAudio].av_stream_index < 0)
     {
         disable_passthru = disable;
@@ -3831 +3868 @@
     AVCodecContext *codec_ctx = NULL;
     AudioInfo old_in  = audioIn;
     AudioInfo old_out = audioOut;
+    bool using_passthru = false;
 
     if ((currentTrack[kTrackTypeAudio] >= 0) &&
         (selectedTrack[kTrackTypeAudio].av_stream_index <=
@@ -3842 +3880 @@
         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 = (using_passthru) ? " using passthru" : "";
     VERBOSE(VB_AUDIO, LOC + "Initializing audio parms from " +
             QString("audio track #%1").arg(currentTrack[kTrackTypeAudio]+1));
 
     audioOut = audioIn = info;
-    if (audioIn.do_passthru)
+    if (using_passthru)
     {
         // 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 > (int) max_channels)
+        {
+            audioOut.channels = (int) max_channels;
+            audioOut.sample_size = audioOut.channels * 2;
+            codec_ctx->channels = audioOut.channels;
+        }
+        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(digInfo.bps(), digInfo.channels,
+                                 digInfo.sample_rate, audioIn.do_passthru);
+        // allow the audio stuff to reencode
+        GetNVP()->SetAudioCodec(codec_ctx);
+        GetNVP()->ReinitAudio();
+        return true;
     }
     else
     {
-        if (audioOut.channels > MAX_OUTPUT_CHANNELS)
+        if (audioOut.channels > (int) max_channels)
         {
-            audioOut.channels = MAX_OUTPUT_CHANNELS;
+            audioOut.channels = (int) max_channels;
             audioOut.sample_size = audioOut.channels * 2;
-            codec_ctx->channels = MAX_OUTPUT_CHANNELS;
+            codec_ctx->channels = audioOut.channels;
         }
     }
 
@@ -3887 +3950 @@
     GetNVP()->SetAudioParams(audioOut.bps(), audioOut.channels,
                              audioOut.sample_rate,
                              audioIn.do_passthru);
-    GetNVP()->ReinitAudio();
 
+    // allow the audio stuff to reencode
+    GetNVP()->SetAudioCodec(using_passthru?codec_ctx:NULL);
+    QString errMsg = GetNVP()->ReinitAudio();
+    bool audiook = errMsg.isEmpty();
+
     return true;
 }
 

libs/libmythtv/NuppelVideoPlayer.h

@@ -127 +127 @@
     void SetAudioInfo(const QString &main, const QString &passthru, uint rate);
     void SetAudioParams(int bits, int channels, int samplerate, bool passthru);
     void SetEffDsp(int dsprate);
+    void SetAudioCodec(void *ac);
 
     // Sets
     void SetParentWidget(QWidget *widget)     { parentWidget = widget; }
@@ -684 +685 @@
     int      audio_bits;
     int      audio_samplerate;
     float    audio_stretchfactor;
+    void    *audio_codec;
     bool     audio_passthru;
 
     // Picture-in-Picture

libs/libmythtv/NuppelVideoPlayer.cpp

@@ -207 +207 @@
       audio_passthru_device(QString::null),
       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
@@ -772 +773 @@
     if (audioOutput)
     {
         audioOutput->Reconfigure(audio_bits, audio_channels,
-                                 audio_samplerate, audio_passthru);
+                                 audio_samplerate, audio_passthru,
+                                 audio_codec);
         errMsg = audioOutput->GetError();
         if (!errMsg.isEmpty())
             audioOutput->SetStretchFactor(audio_stretchfactor);
@@ -3657 +3664 @@
     audio_passthru = passthru;
 }
 
+void NuppelVideoPlayer::SetAudioCodec(void *ac)
+{
+    audio_codec = ac;
+}
+
 void NuppelVideoPlayer::SetEffDsp(int dsprate)
 {
     if (audioOutput)

libs/libavcodec/liba52.c

@@ -134 +134 @@
     }
 }
 
+static inline int16_t convert(int32_t i)
+{
+    return av_clip_int16(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]);
+    }
+}
+
+#define LIKEAC3DEC 1
+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++) {
+#if LIKEAC3DEC
+            s16[6*i] = s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
+            s16[6*i+1] = convert (f[i+256]);
+            s16[6*i+5] = convert (f[i]);
+#else
+            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]);
+#endif
+        }
+        break;
+    case A52_CHANNEL | A52_LFE:
+    case A52_STEREO | A52_LFE:
+    case A52_DOLBY | A52_LFE:
+        for (i = 0; i < 256; i++) {
+#if LIKEAC3DEC
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+2] = convert (f[i+512]);
+            s16[6*i+1] = s16[6*i+3] = s16[6*i+4] = 0;
+            s16[6*i+5] = convert (f[i]);
+#else
+            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]);
+#endif
+        }
+        break;
+    case A52_3F | A52_LFE:
+        for (i = 0; i < 256; i++) {
+#if LIKEAC3DEC
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+2] = convert (f[i+768]);
+            s16[6*i+3] = s16[6*i+4] = 0;
+            s16[6*i+1] = convert (f[i+512]);
+            s16[6*i+5] = convert (f[i]);
+#else
+            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]);
+#endif
+        }
+        break;
+    case A52_2F2R | A52_LFE:
+        for (i = 0; i < 256; i++) {
+#if LIKEAC3DEC
+            s16[6*i] = convert (f[i+256]);
+            s16[6*i+1] = 0;
+            s16[6*i+2] = convert (f[i+512]);
+            s16[6*i+3] = convert (f[i+768]);
+            s16[6*i+4] = convert (f[i+1024]);
+            s16[6*i+5] = convert (f[i]);
+#else
+            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]);
+#endif
+        }
+        break;
+    case A52_3F2R | A52_LFE:
+        for (i = 0; i < 256; i++) {
+#if LIKEAC3DEC
+            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] = convert (f[i+1280]);
+            s16[6*i+5] = convert (f[i]);
+#else
+            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]);
+#endif
+        }
+        break;
+    }
+}
+
 /**** end */
 
 #define HEADER_SIZE 7
@@ -179 +354 @@
                     s->channels = ac3_channels[s->flags & 7];
                     if (s->flags & A52_LFE)
                         s->channels++;
+                    if (avctx->request_channels > 0)
+                    {
+                        avctx->channels = s->channels;
+                        if (s->channels > avctx->channels)
+                            avctx->channels = avctx->request_channels;
+                    }
                     if (avctx->channels == 0)
                         /* No specific number of channel requested */
                         avctx->channels = s->channels;
@@ -199 +380 @@
             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:
                 av_log(avctx, AV_LOG_ERROR, "Error decoding frame\n");
@@ -217 +404 @@
             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);
+                float2s16_multi(s->samples, out_samples + i * 256 * chans, flags);
             }
             s->inbuf_ptr = s->inbuf;
             s->frame_size = 0;

libs/libavcodec/ac3dec.c

@@ -1132 +1132 @@
 
     /* channel config */
     ctx->out_channels = ctx->nchans;
+    if (avctx->request_channels > 0)
+    {
+        avctx->channels = ctx->out_channels;
+        if (avctx->channels > avctx->request_channels)
+            avctx->channels = avctx->request_channels;
+    }
     if (avctx->channels == 0) {
         avctx->channels = ctx->out_channels;
     } else if(ctx->out_channels < avctx->channels) {

libs/libavcodec/dca.c

@@ -1159 +1159 @@
     avctx->bit_rate = s->bit_rate;
 
     channels = s->prim_channels + !!s->lfe;
-    avctx->channels = avctx->request_channels;
+    //avctx->channels = avctx->request_channels;
+    if (avctx->request_channels > 0)
+    {
+        avctx->channels = channels;
+        if (avctx->channels > avctx->request_channels)
+            avctx->channels = avctx->request_channels;
+    }
     if(avctx->channels == 0) {
         avctx->channels = channels;
     } else if(channels < avctx->channels) {