Move SPDIF/IEC958 encapsulation within the audio class so avfd doesn'…

…t depends directly on any of the audio code.

Fixes timestamp estimation when timestamps are missing or frames are dropped

mythtv/libs/libmyth/audiooutput.h

@@ -76,7 +76,9 @@ class MPUBLIC AudioOutput : public VolumeBase, public OutputListeners
 
     virtual void Reset(void) = 0;
 
-    virtual bool AddFrames(void *buffer, int samples, int64_t timecode) = 0;
+    virtual bool AddFrames(void *buffer, int frames, int64_t timecode) = 0;
+    virtual bool AddData(void *buffer, int len, int64_t timecode) = 0;
+    virtual int64_t LengthLastData(void) { return 0; }
 
     virtual void SetTimecode(int64_t timecode) = 0;
     virtual bool IsPaused(void) const = 0;

mythtv/libs/libmyth/audiooutputbase.cpp

@@ -17,6 +17,7 @@
 #include "audiooutputdownmix.h"
 #include "SoundTouch.h"
 #include "freesurround.h"
+#include "spdifencoder.h"
 
 #define LOC QString("AO: ")
 #define LOC_ERR QString("AO, ERROR: ")
@@ -97,7 +98,8 @@ AudioOutputBase::AudioOutputBase(const AudioSettings &settings) :
     src_out(NULL),              kAudioSRCOutputSize(0),
     memory_corruption_test2(0xdeadbeef),
     memory_corruption_test3(0xdeadbeef),
-    m_configure_succeeded(true)
+    m_configure_succeeded(true),m_length_last_data(0),
+    m_spdifenc(NULL)
 {
     src_in = (float *)AOALIGN(src_in_buf);
     // The following are not bzero() because MS Windows doesn't like it.
@@ -343,25 +345,25 @@ bool AudioOutputBase::ToggleUpmix(void)
 
 /*
  * Setup samplerate and number of channels for passthrough
+ * Create SPDIF encoder and true if successful
  */
-void AudioOutputBase::SetupPassthrough(AudioSettings &settings,
+bool AudioOutputBase::SetupPassthrough(int codec, int codec_profile,
                                        int &samplerate_tmp, int &channels_tmp)
 {
-    samplerate_tmp = settings.samplerate;
     channels_tmp = 2;
     QString log;
 
-    switch (settings.codec)
+    switch (codec)
     {
         case CODEC_ID_AC3:
             log = "AC3";
             break;
         case CODEC_ID_EAC3:
-            samplerate_tmp = settings.samplerate * 4;
+            samplerate_tmp = samplerate_tmp * 4;
             log = "Dolby Digital Plus (E-AC3)";
             break;
         case CODEC_ID_DTS:
-            switch(settings.codec_profile)
+            switch(codec_profile)
             {
                 case FF_PROFILE_DTS_ES:
                     log = "DTS-ES";
@@ -391,7 +393,7 @@ void AudioOutputBase::SetupPassthrough(AudioSettings &settings,
         case CODEC_ID_TRUEHD:
             channels_tmp = 8;
             log = "TrueHD";
-            switch(settings.samplerate)
+            switch(samplerate_tmp)
             {
                 case 48000:
                 case 96000:
@@ -412,6 +414,39 @@ void AudioOutputBase::SetupPassthrough(AudioSettings &settings,
             break;
     }
     VBAUDIO("Setting " + log + " passthrough");
+
+    if (m_spdifenc)
+    {
+        delete m_spdifenc;
+    }
+
+    m_spdifenc = new SPDIFEncoder("spdif", codec);
+    if (m_spdifenc->Succeeded() && codec == CODEC_ID_DTS)
+    {
+        switch(codec_profile)
+        {
+            case FF_PROFILE_DTS:
+            case FF_PROFILE_DTS_ES:
+            case FF_PROFILE_DTS_96_24:
+                m_spdifenc->SetMaxHDRate(0);
+                break;
+            case FF_PROFILE_DTS_HD_HRA:
+                m_spdifenc->SetMaxHDRate(
+                    OutputSettings(true)->canFeature(FEATURE_DTSHD) ?
+                    192000 : 0);
+                break;
+            case FF_PROFILE_DTS_HD_MA:
+                m_spdifenc->SetMaxHDRate(OutputSettings(true)->GetMaxHDRate());
+        }
+    }
+
+    if (!m_spdifenc->Succeeded())
+    {
+        delete m_spdifenc;
+        m_spdifenc = NULL;
+        return false;
+    }
+    return true;
 }
 
 AudioOutputSettings *AudioOutputBase::OutputSettings(bool digital)
@@ -495,7 +530,9 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings)
     int samplerate_tmp, channels_tmp;
     if (settings.use_passthru)
     {
-        SetupPassthrough(settings, samplerate_tmp, channels_tmp);
+        samplerate_tmp = settings.samplerate;
+        SetupPassthrough(settings.codec, settings.codec_profile,
+                         samplerate_tmp, channels_tmp);
         general_deps = samplerate == samplerate_tmp && channels == channels_tmp;
     }
 
@@ -859,7 +896,7 @@ void AudioOutputBase::Reset()
  * Set the timecode of the samples most recently added to the audiobuffer
  *
  * Used by mythmusic for seeking since it doesn't provide timecodes to
- * AddFrames()
+ * AddData()
  */
 void AudioOutputBase::SetTimecode(int64_t timecode)
 {
@@ -953,7 +990,7 @@ int64_t AudioOutputBase::GetAudiotime(void)
 
     /* timecode is the stretch adjusted version
        of major post-stretched buffer contents
-       processing latencies are catered for in AddFrames/SetAudiotime
+       processing latencies are catered for in AddData/SetAudiotime
        to eliminate race */
     audiotime = audbuf_timecode - (
         ((int64_t)(main_buffer + soundcard_buffer) * eff_stretchfactor) /
@@ -983,7 +1020,7 @@ int64_t AudioOutputBase::GetAudiotime(void)
 /**
  * Set the timecode of the top of the ringbuffer
  * Exclude all other processing elements as they dont vary
- * between AddFrames calls
+ * between AddData calls
  */
 void AudioOutputBase::SetAudiotime(int frames, int64_t timecode)
 {
@@ -1193,27 +1230,35 @@ int AudioOutputBase::CopyWithUpmix(char *buffer, int frames, int &org_waud)
 bool AudioOutputBase::AddFrames(void *in_buffer, int in_frames,
                                 int64_t timecode)
 {
-    int org_waud = waud,               afree = audiofree();
-    int frames   = in_frames;
+    return AddData(in_buffer, in_frames * source_bytes_per_frame, timecode);
+}
+
+/**
+ * Add data to the audiobuffer and perform any required processing
+ *
+ * Returns false if there's not enough space right now
+ */
+bool AudioOutputBase::AddData(void *in_buffer, int in_len,
+                              int64_t timecode)
+{
+    int org_waud = waud;
+    int afree    = audiofree();
+    int frames   = in_len / source_bytes_per_frame;
     void *buffer = in_buffer;
     int bpf      = bytes_per_frame;
-    int len      = frames * source_bytes_per_frame;
+    int len      = in_len;
     int used     = kAudioRingBufferSize - afree;
     bool music   = false;
     int bdiff;
 
     if (!m_configure_succeeded)
     {
-        VERBOSE(VB_IMPORTANT, LOC + "AddFrames called with audio framework not "
+        VERBOSE(VB_IMPORTANT, LOC + "AddData called with audio framework not "
                 "initialised");
+        m_length_last_data = 0;
         return false;
     }
 
-    VBAUDIOTS(QString("AddFrames frames=%1, bytes=%2, used=%3, free=%4, "
-                      "timecode=%5 needsupmix=%6")
-              .arg(frames).arg(len).arg(used).arg(afree).arg(timecode)
-              .arg(needs_upmix));
-
     /* See if we're waiting for new samples to be buffered before we unpause
        post channel change, seek, etc. Wait for 4 fragments to be buffered */
     if (unpause_when_ready && pauseaudio && audioready() > fragment_size << 2)
@@ -1222,6 +1267,28 @@ bool AudioOutputBase::AddFrames(void *in_buffer, int in_frames,
         Pause(false);
     }
 
+    if (passthru && m_spdifenc)
+    {
+        // mux into an IEC958 packet
+        m_spdifenc->WriteFrame((unsigned char *)in_buffer, len);
+        len = m_spdifenc->GetProcessedSize();
+        if (len > 0)
+        {
+            buffer = in_buffer = m_spdifenc->GetProcessedBuffer();
+            m_spdifenc->Reset();
+            frames = len / source_bytes_per_frame;
+        }
+        else
+            frames = 0;
+    }
+    m_length_last_data = (int64_t)
+        ((double)(len * 1000) / (source_samplerate * source_bytes_per_frame));
+
+    VBAUDIOTS(QString("AddData frames=%1, bytes=%2, used=%3, free=%4, "
+                      "timecode=%5 needsupmix=%6")
+              .arg(frames).arg(len).arg(used).arg(afree).arg(timecode)
+              .arg(needs_upmix));
+
     // Don't write new samples if we're resetting the buffer or reconfiguring
     QMutexLocker lock(&audio_buflock);
 
@@ -1263,11 +1330,11 @@ bool AudioOutputBase::AddFrames(void *in_buffer, int in_frames,
 
     if (len > afree)
     {
-        VBAUDIOTS("Buffer is full, AddFrames returning false");
+        VBAUDIOTS("Buffer is full, AddData returning false");
         return false; // would overflow
     }
 
-    int frames_remaining = in_frames;
+    int frames_remaining = frames;
     int frames_final = 0;
     int maxframes = (kAudioSRCInputSize /
                      (passthru ? channels : source_channels)) & ~0xf;

mythtv/libs/libmyth/audiooutputbase.h

@@ -43,6 +43,9 @@ class AsyncLooseLock
     int tail;
 };
 
+// Forward declaration of SPDIF encoder
+class SPDIFEncoder;
+
 class AudioOutputBase : public AudioOutput, public QThread
 {
  public:
@@ -75,6 +78,8 @@ class AudioOutputBase : public AudioOutput, public QThread
 
     // timecode is in milliseconds.
     virtual bool AddFrames(void *buffer, int frames, int64_t timecode);
+    virtual bool AddData(void *buffer, int len, int64_t timecode);
+    virtual int64_t LengthLastData(void) { return m_length_last_data; }
 
     virtual void SetTimecode(int64_t timecode);
     virtual bool IsPaused(void) const { return actually_paused; }
@@ -178,9 +183,9 @@ class AudioOutputBase : public AudioOutput, public QThread
     int src_quality;
 
  private:
-    void SetupPassthrough(AudioSettings &settings, int &samplerate_tmp,
-                          int &channels_tmp);
-    AudioOutputSettings* OutputSettings(bool digital);
+    bool SetupPassthrough(int codec, int codec_profile,
+                          int &samplerate_tmp, int &channels_tmp);
+    AudioOutputSettings* OutputSettings(bool digital = true);
     int CopyWithUpmix(char *buffer, int frames, int &org_waud);
     void SetAudiotime(int frames, int64_t timecode);
     AudioOutputSettings *output_settingsraw;
@@ -258,6 +263,10 @@ class AudioOutputBase : public AudioOutput, public QThread
     uchar audiobuffer[kAudioRingBufferSize];
     uint memory_corruption_test3;
     uint m_configure_succeeded;
+    int64_t m_length_last_data;
+
+    // SPDIF Encoder for digital passthrough
+    SPDIFEncoder     *m_spdifenc;
 };
 
 #endif

mythtv/libs/libmyth/audiooutputdigitalencoder.cpp

@@ -94,7 +94,7 @@ bool AudioOutputDigitalEncoder::Init(
         delete m_spdifenc;
     }
 
-    m_spdifenc = new SPDIFEncoder("spdif", av_context);
+    m_spdifenc = new SPDIFEncoder("spdif", CODEC_ID_AC3);
     if (!m_spdifenc->Succeeded())
     {
         Dispose();
@@ -144,7 +144,7 @@ size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, bool isFloat)
 
         if (!m_spdifenc)
         {
-            m_spdifenc = new SPDIFEncoder("spdif", av_context);
+            m_spdifenc = new SPDIFEncoder("spdif", CODEC_ID_AC3);
         }
         m_spdifenc->WriteFrame(m_encodebuffer, outsize);
         m_spdifenc->GetData(out + outlen, data_size);

mythtv/libs/libmyth/audiooutpututil.cpp

@@ -640,8 +640,8 @@ void AudioOutputUtil::MuteChannel(int obits, int channels, int ch,
 #define LE_INT(v)		(v)
 #endif
 
-char *AudioOutputUtil::GeneratePinkSamples(char *frames, int channels,
-                                           int channel, int count, int bits)
+char *AudioOutputUtil::GeneratePinkFrames(char *frames, int channels,
+                                          int channel, int count, int bits)
 {
     pink_noise_t pink;
 

mythtv/libs/libmyth/audiooutpututil.h

@@ -19,8 +19,8 @@ class MPUBLIC AudioOutputUtil
                              bool music, bool upmix);
     static void MuteChannel(int obits, int channels, int ch,
                             void *buffer, int bytes);
-    static char *GeneratePinkSamples(char *frames, int channels,
-                                     int channel, int count, int bits = 16);
+    static char *GeneratePinkFrames(char *frames, int channels,
+                                    int channel, int count, int bits = 16);
 };
 
 #endif

mythtv/libs/libmyth/spdifencoder.cpp

@@ -18,7 +18,7 @@
  *                         Use "adts" for ADTS encpsulation (AAC)
  *   AVCodecContext *ctx : CodecContext to be encaspulated
  */
-SPDIFEncoder::SPDIFEncoder(QString muxer, AVCodecContext *ctx)
+SPDIFEncoder::SPDIFEncoder(QString muxer, int codec_id)
     : m_complete(false), m_oc(NULL), m_stream(NULL), m_size(0)
 {
     QByteArray dev_ba     = muxer.toAscii();
@@ -102,21 +102,11 @@ SPDIFEncoder::SPDIFEncoder(QString muxer, AVCodecContext *ctx)
 
     AVCodecContext *codec = m_stream->codec;
 
-    codec->codec_type     = ctx->codec_type;
-    codec->codec_id       = ctx->codec_id;
-    codec->sample_rate    = ctx->sample_rate;
-    codec->sample_fmt     = ctx->sample_fmt;
-    codec->channels       = ctx->channels;
-    codec->bit_rate       = ctx->bit_rate;
-    codec->extradata      = new uint8_t[ctx->extradata_size];
-    codec->extradata_size = ctx->extradata_size;
-    memcpy(codec->extradata, ctx->extradata, ctx->extradata_size);
-
+    codec->codec_id       = (CodecID)codec_id;
     av_write_header(m_oc);
 
-    VERBOSE(VB_AUDIO, LOC + QString("Creating %1 encoder (%2, %3Hz)")
-            .arg(muxer).arg(ff_codec_id_string((CodecID)codec->codec_type))
-            .arg(codec->sample_rate));
+    VERBOSE(VB_AUDIO, LOC + QString("Creating %1 encoder (for %2)")
+            .arg(muxer).arg(ff_codec_id_string((CodecID)codec_id)));
 
     m_complete = true;
 }

mythtv/libs/libmyth/spdifencoder.h

@@ -15,10 +15,12 @@ extern "C" {
 class MPUBLIC SPDIFEncoder
 {
   public:
-    SPDIFEncoder(QString muxer, AVCodecContext *ctx);
+    SPDIFEncoder(QString muxer, int codec_id);
     ~SPDIFEncoder();
     void WriteFrame(unsigned char *data, int size);
     int  GetData(unsigned char *buffer, int &dest_size);
+    int  GetProcessedSize() { return m_size; };
+    unsigned char *GetProcessedBuffer() { return m_buffer; };
     void Reset();
     bool Succeeded()  { return m_complete; };
     bool SetMaxHDRate(int rate);

mythtv/libs/libmythtv/audioplayer.cpp

@@ -363,11 +363,19 @@ void AudioPlayer::AddAudioData(char *buffer, int len, int64_t timecode)
     if (samplesize <= 0)
         return;
 
-    if (!m_audioOutput->AddFrames(buffer, len / samplesize, timecode))
+    if (!m_audioOutput->AddData(buffer, len, timecode))
         VERBOSE(VB_PLAYBACK, LOC + "AddAudioData(): "
                 "Audio buffer overflow, audio data lost!");
 }
 
+int64_t AudioPlayer::LengthLastData(void)
+{
+    if (!m_audioOutput)
+        return 0;
+    else
+        return m_audioOutput->LengthLastData();
+}
+
 bool AudioPlayer::GetBufferStatus(uint &fill, uint &total)
 {
     fill = total = 0;

mythtv/libs/libmythtv/audioplayer.h

@@ -58,6 +58,7 @@ class MPUBLIC AudioPlayer
     MuteState IncrMuteState(void);
 
     void AddAudioData(char *buffer, int len, int64_t timecode);
+    int64_t LengthLastData(void);
     bool GetBufferStatus(uint &fill, uint &total);
     bool IsBufferAlmostFull(void);