avaudenc: Add support for planar audio formats

diff --git a/ext/libav/gstavaudenc.c b/ext/libav/gstavaudenc.c
index 6996da7..d3f58cc 100644 (file)
--- a/ext/libav/gstavaudenc.c
+++ b/ext/libav/gstavaudenc.c
@@ -367,6 +367,7 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
   GstAudioInfo *info;
   AVPacket pkt;
   AVFrame frame;
+  gboolean planar;
 
   enc = GST_AUDIO_ENCODER (ffmpegaudenc);
 
@@ -378,11 +379,84 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
   memset (&frame, 0, sizeof (frame));
 
   info = gst_audio_encoder_get_audio_info (enc);
-  frame.data[0] = audio_in;
-  frame.linesize[0] = in_size;
-  frame.nb_samples = in_size / info->bpf;
+  planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt);
+
+  if (planar && info->channels > 1) {
+    gint channels, nsamples;
+    gint i, j;
+
+    nsamples = frame.nb_samples = in_size / info->bpf;
+    channels = info->channels;
+
+    frame.data[0] = g_malloc (in_size);
+    frame.linesize[0] = in_size / channels;
+    for (i = 1; i < channels; i++) {
+      frame.data[i] = frame.data[i - 1] + frame.linesize[0];
+      frame.linesize[i] = frame.linesize[0];
+    }
+
+    switch (info->finfo->width) {
+      case 8:{
+        const guint8 *idata = (const guint8 *) audio_in;
+
+        for (i = 0; i < nsamples; i++) {
+          for (j = 0; j < channels; j++) {
+            ((guint8 *) frame.data[j])[i] = idata[j];
+          }
+          idata += channels;
+        }
+        break;
+      }
+      case 16:{
+        const guint16 *idata = (const guint16 *) audio_in;
+
+        for (i = 0; i < nsamples; i++) {
+          for (j = 0; j < channels; j++) {
+            ((guint16 *) frame.data[j])[i] = idata[j];
+          }
+          idata += channels;
+        }
+        break;
+      }
+      case 32:{
+        const guint32 *idata = (const guint32 *) audio_in;
+
+        for (i = 0; i < nsamples; i++) {
+          for (j = 0; j < channels; j++) {
+            ((guint32 *) frame.data[j])[i] = idata[j];
+          }
+          idata += channels;
+        }
+
+        break;
+      }
+      case 64:{
+        const guint64 *idata = (const guint64 *) audio_in;
+
+        for (i = 0; i < nsamples; i++) {
+          for (j = 0; j < channels; j++) {
+            ((guint64 *) frame.data[j])[i] = idata[j];
+          }
+          idata += channels;
+        }
+
+        break;
+      }
+      default:
+        g_assert_not_reached ();
+        break;
+    }
+
+  } else {
+    frame.data[0] = audio_in;
+    frame.linesize[0] = in_size;
+    frame.nb_samples = in_size / info->bpf;
+  }
 
   res = avcodec_encode_audio2 (ctx, &pkt, &frame, have_data);
+  if (planar && info->channels > 1)
+    g_free (frame.data[0]);
+
   if (res < 0) {
     char error_str[128] = { 0, };
 

diff --git a/ext/libav/gstavcodecmap.c b/ext/libav/gstavcodecmap.c
index 7b30c93..75f3956 100644 (file)
--- a/ext/libav/gstavcodecmap.c
+++ b/ext/libav/gstavcodecmap.c
@@ -1747,19 +1747,25 @@ gst_ffmpeg_pixfmt_to_caps (enum PixelFormat pix_fmt, AVCodecContext * context,
 GstAudioFormat
 gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt)
 {
-  sample_fmt = av_get_packed_sample_fmt (sample_fmt);
-
   switch (sample_fmt) {
+    case AV_SAMPLE_FMT_U8:
+    case AV_SAMPLE_FMT_U8P:
+      return GST_AUDIO_FORMAT_U8;
+      break;
     case AV_SAMPLE_FMT_S16:
+    case AV_SAMPLE_FMT_S16P:
       return GST_AUDIO_FORMAT_S16;
       break;
     case AV_SAMPLE_FMT_S32:
+    case AV_SAMPLE_FMT_S32P:
       return GST_AUDIO_FORMAT_S32;
       break;
     case AV_SAMPLE_FMT_FLT:
+    case AV_SAMPLE_FMT_FLTP:
       return GST_AUDIO_FORMAT_F32;
       break;
     case AV_SAMPLE_FMT_DBL:
+    case AV_SAMPLE_FMT_DBLP:
       return GST_AUDIO_FORMAT_F64;
       break;
     default:
@@ -2201,25 +2207,105 @@ gst_ffmpeg_videoinfo_to_context (GstVideoInfo * info, AVCodecContext * context)
 void
 gst_ffmpeg_audioinfo_to_context (GstAudioInfo * info, AVCodecContext * context)
 {
+  const AVCodec *codec;
+  const enum AVSampleFormat *smpl_fmts;
+  enum AVSampleFormat smpl_fmt = -1;
+
   context->channels = info->channels;
   context->sample_rate = info->rate;
 
+  codec = context->codec;
+
+  smpl_fmts = codec->sample_fmts;
+
   switch (info->finfo->format) {
     case GST_AUDIO_FORMAT_F32:
-      context->sample_fmt = AV_SAMPLE_FMT_FLT;
+      if (smpl_fmts) {
+        while (*smpl_fmts != -1) {
+          if (*smpl_fmts == AV_SAMPLE_FMT_FLT) {
+            smpl_fmt = *smpl_fmts;
+            break;
+          } else if (*smpl_fmts == AV_SAMPLE_FMT_FLTP) {
+            smpl_fmt = *smpl_fmts;
+          }
+
+          smpl_fmts++;
+        }
+      } else {
+        smpl_fmt = AV_SAMPLE_FMT_FLT;
+      }
       break;
     case GST_AUDIO_FORMAT_F64:
-      context->sample_fmt = AV_SAMPLE_FMT_DBL;
+      if (smpl_fmts) {
+        while (*smpl_fmts != -1) {
+          if (*smpl_fmts == AV_SAMPLE_FMT_DBL) {
+            smpl_fmt = *smpl_fmts;
+            break;
+          } else if (*smpl_fmts == AV_SAMPLE_FMT_DBLP) {
+            smpl_fmt = *smpl_fmts;
+          }
+
+          smpl_fmts++;
+        }
+      } else {
+        smpl_fmt = AV_SAMPLE_FMT_DBL;
+      }
       break;
     case GST_AUDIO_FORMAT_S32:
-      context->sample_fmt = AV_SAMPLE_FMT_S32;
+      if (smpl_fmts) {
+        while (*smpl_fmts != -1) {
+          if (*smpl_fmts == AV_SAMPLE_FMT_S32) {
+            smpl_fmt = *smpl_fmts;
+            break;
+          } else if (*smpl_fmts == AV_SAMPLE_FMT_S32P) {
+            smpl_fmt = *smpl_fmts;
+          }
+
+          smpl_fmts++;
+        }
+      } else {
+        smpl_fmt = AV_SAMPLE_FMT_S32;
+      }
       break;
     case GST_AUDIO_FORMAT_S16:
-      context->sample_fmt = AV_SAMPLE_FMT_S16;
+      if (smpl_fmts) {
+        while (*smpl_fmts != -1) {
+          if (*smpl_fmts == AV_SAMPLE_FMT_S16) {
+            smpl_fmt = *smpl_fmts;
+            break;
+          } else if (*smpl_fmts == AV_SAMPLE_FMT_S16P) {
+            smpl_fmt = *smpl_fmts;
+          }
+
+          smpl_fmts++;
+        }
+      } else {
+        smpl_fmt = AV_SAMPLE_FMT_S16;
+      }
+      break;
+    case GST_AUDIO_FORMAT_U8:
+      if (smpl_fmts) {
+        while (*smpl_fmts != -1) {
+          if (*smpl_fmts == AV_SAMPLE_FMT_U8) {
+            smpl_fmt = *smpl_fmts;
+            break;
+          } else if (*smpl_fmts == AV_SAMPLE_FMT_U8P) {
+            smpl_fmt = *smpl_fmts;
+          }
+
+          smpl_fmts++;
+        }
+      } else {
+        smpl_fmt = AV_SAMPLE_FMT_U8;
+      }
       break;
     default:
       break;
   }
+
+  g_assert (smpl_fmt != -1);
+
+  context->sample_fmt = smpl_fmt;
 }
 
 /* Convert a GstCaps and a FFMPEG codec Type to a