Fix AC3 encoder following FFmpeg resync

AC3 encoder now expects planar audio data. This commit adds an utility method to de-interleave and convert interleaved data into planar data.

Fixes #11483

mythtv/libs/libmyth/audio/audiooutputdigitalencoder.cpp

@@ -23,7 +23,7 @@ extern "C" {
 AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) :
     av_context(NULL),
     out(NULL), out_size(0),
-    in(NULL), in_size(0),
+    in(NULL), inp(NULL), in_size(0),
     outlen(0), inlen(0),
     samples_per_frame(0),
     m_spdifenc(NULL)
@@ -38,6 +38,7 @@ AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) :
     {
         in_size = INBUFSIZE;
     }
+    inp = (inbuf_t *)av_malloc(INBUFSIZE);
 }
 
 AudioOutputDigitalEncoder::~AudioOutputDigitalEncoder()
@@ -62,6 +63,10 @@ void AudioOutputDigitalEncoder::Dispose()
         av_freep(&in);
         in_size = 0;
     }
+    if (inp)
+    {
+        av_freep(&inp);
+    }
     if (m_spdifenc)
     {
         delete m_spdifenc;
@@ -136,7 +141,7 @@ bool AudioOutputDigitalEncoder::Init(
             av_context->channel_layout = AV_CH_LAYOUT_5POINT1;
             break;
     }
-    av_context->sample_fmt     = AV_SAMPLE_FMT_S16;
+    av_context->sample_fmt     = AV_SAMPLE_FMT_S16P;
 
 // open it
     ret = avcodec_open2(av_context, codec, NULL);
@@ -205,20 +210,35 @@ size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, AudioFormat format)
         inlen += len;
     }
 
-    int frames = inlen / sizeof(inbuf_t) / samples_per_frame;
-    int i = 0;
-    AVFrame *frame = avcodec_alloc_frame();
+    int frames           = inlen / sizeof(inbuf_t) / samples_per_frame;
+    int i                = 0;
+    int channels         = av_context->channels;
+    AVFrame *frame       = avcodec_alloc_frame();
+    int size_channel     = av_context->frame_size *
+        AudioOutputSettings::SampleSize(FORMAT_S16);
+    frame->extended_data = frame->data;
+    frame->nb_samples    = av_context->frame_size;
+    frame->pts           = AV_NOPTS_VALUE;
+        // init AVFrame for planar data (input is interleaved)
+    for (int j = 0, jj = 0; j < channels; j++, jj += samples_per_frame)
+    {
+        frame->data[j] = (uint8_t*)(inp + jj);
+    }
 
     while (i < frames)
     {
         AVPacket pkt;
         av_init_packet(&pkt);
         pkt.data          = (uint8_t *)m_encodebuffer;
         pkt.size          = sizeof(m_encodebuffer);
-        frame->nb_samples = av_context->frame_size;
-        frame->data[0]    = (uint8_t *)(in + i * samples_per_frame);
-        frame->pts        = AV_NOPTS_VALUE;
         int got_packet    = 0;
+
+        AudioOutputUtil::DeinterleaveSamples(
+            FORMAT_S16, channels,
+            (uint8_t*)inp,
+            (uint8_t*)(in + i * samples_per_frame),
+            size_channel * channels);
+
         int ret           = avcodec_encode_audio2(av_context, &pkt, frame,
                                                   &got_packet);
         int outsize       = pkt.size;

mythtv/libs/libmyth/audio/audiooutputdigitalencoder.h

@@ -34,7 +34,7 @@ class AudioOutputDigitalEncoder
     AVCodecContext *av_context;
     outbuf_t       *out;
     size_t          out_size;
-    inbuf_t        *in;
+    inbuf_t        *in, *inp;
     size_t          in_size;
     int             outlen;
     int             inlen;

mythtv/libs/libmyth/audio/audiooutpututil.cpp

@@ -767,3 +767,45 @@ int AudioOutputUtil::DecodeAudio(AVCodecContext *ctx,
 
     return ret;
 }
+
+template <class AudioDataType>
+void _DeinterleaveSample(AudioDataType *out, AudioDataType *in, int channels, int frames)
+{
+    AudioDataType *outp[8];
+
+    for (int i = 0; i < channels; i++)
+    {
+        outp[i] = out + (i * channels * frames);
+    }
+
+    for (int i = 0; i < frames; i++)
+    {
+        for (int j = 0; j < channels; j++)
+        {
+            *(outp[j]++) = *(in++);
+        }
+    }
+}
+
+/**
+ * Deinterleave input samples
+ * Deinterleave audio samples and compact them
+ */
+void AudioOutputUtil::DeinterleaveSamples(AudioFormat format, int channels,
+                                          uint8_t *output, uint8_t *input,
+                                          int data_size)
+{
+    int bits = AudioOutputSettings::FormatToBits(format);
+    if (bits == 8)
+    {
+        _DeinterleaveSample((char *)output, (char *)input, channels, data_size/sizeof(char)/channels);
+    }
+    else if (bits == 16)
+    {
+        _DeinterleaveSample((short *)output, (short *)input, channels, data_size/sizeof(short)/channels);
+    }
+    else
+    {
+        _DeinterleaveSample((int *)output, (int *)input, channels, data_size/sizeof(int)/channels);
+    }
+}

mythtv/libs/libmyth/audio/audiooutpututil.h

@@ -19,6 +19,9 @@ class MPUBLIC AudioOutputUtil
     static int DecodeAudio(AVCodecContext *ctx,
                            uint8_t *buffer, int &data_size,
                            AVPacket *pkt);
+    static void DeinterleaveSamples(AudioFormat format, int channels,
+                                    uint8_t *output, uint8_t *input,
+                                    int data_size);
 };
 
 #endif