avauddec: add support for decoding in non-interleaved layout

This removes the internal interleave loop and always negotiates
the native output layout of the libav decoder. Users can use
audioconvert to interleave if necessary.

Special care has been taken to leave the encoder unaffected by
the changes in avcodecmap, since GstAudioEncoder doesn't support
the non-interleaved layout yet.

https://bugzilla.gnome.org/show_bug.cgi?id=705977

diff --git a/ext/libav/gstavauddec.c b/ext/libav/gstavauddec.c
index a2518df..dbec03c 100644 (file)
--- a/ext/libav/gstavauddec.c
+++ b/ext/libav/gstavauddec.c
@@ -360,16 +360,17 @@ static gboolean
 settings_changed (GstFFMpegAudDec * ffmpegdec, AVFrame * frame)
 {
   GstAudioFormat format;
+  GstAudioLayout layout;
   gint channels = av_get_channel_layout_nb_channels (frame->channel_layout);
 
-  format = gst_ffmpeg_smpfmt_to_audioformat (frame->format);
+  format = gst_ffmpeg_smpfmt_to_audioformat (frame->format, &layout);
   if (format == GST_AUDIO_FORMAT_UNKNOWN)
     return TRUE;
 
-  return !(ffmpegdec->info.rate ==
-      frame->sample_rate &&
+  return !(ffmpegdec->info.rate == frame->sample_rate &&
       ffmpegdec->info.channels == channels &&
-      ffmpegdec->info.finfo->format == format);
+      ffmpegdec->info.finfo->format == format &&
+      ffmpegdec->info.layout == layout);
 }
 
 static gboolean
@@ -378,12 +379,13 @@ gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
 {
   GstFFMpegAudDecClass *oclass;
   GstAudioFormat format;
+  GstAudioLayout layout;
   gint channels;
   GstAudioChannelPosition pos[64] = { 0, };
 
   oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
 
-  format = gst_ffmpeg_smpfmt_to_audioformat (frame->format);
+  format = gst_ffmpeg_smpfmt_to_audioformat (frame->format, &layout);
   if (format == GST_AUDIO_FORMAT_UNKNOWN)
     goto no_caps;
   channels = av_get_channel_layout_nb_channels (frame->channel_layout);
@@ -396,9 +398,13 @@ gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
     return TRUE;
 
   GST_DEBUG_OBJECT (ffmpegdec,
-      "Renegotiating audio from %dHz@%dchannels (%d) to %dHz@%dchannels (%d)",
+      "Renegotiating audio from %dHz@%dchannels (%d, interleaved=%d) "
+      "to %dHz@%dchannels (%d, interleaved=%d)",
       ffmpegdec->info.rate, ffmpegdec->info.channels,
-      ffmpegdec->info.finfo->format, frame->sample_rate, channels, format);
+      ffmpegdec->info.finfo->format,
+      ffmpegdec->info.layout == GST_AUDIO_LAYOUT_INTERLEAVED,
+      frame->sample_rate, channels, format,
+      layout == GST_AUDIO_LAYOUT_INTERLEAVED);
 
   gst_ffmpeg_channel_layout_to_gst (frame->channel_layout, channels, pos);
   memcpy (ffmpegdec->ffmpeg_layout, pos,
@@ -410,6 +416,7 @@ gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
       memcmp (pos, ffmpegdec->ffmpeg_layout, sizeof (pos[0]) * channels) != 0;
   gst_audio_info_set_format (&ffmpegdec->info, format,
       frame->sample_rate, channels, pos);
+  ffmpegdec->info.layout = layout;
 
   if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (ffmpegdec),
           &ffmpegdec->info))
@@ -472,6 +479,7 @@ gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
   if (res >= 0) {
     gint nsamples, channels, byte_per_sample;
     gsize output_size;
+    gboolean planar;
 
     if (!gst_ffmpegauddec_negotiate (ffmpegdec, ffmpegdec->context,
             ffmpegdec->frame, FALSE)) {
@@ -485,81 +493,33 @@ gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
     channels = ffmpegdec->info.channels;
     nsamples = ffmpegdec->frame->nb_samples;
     byte_per_sample = ffmpegdec->info.finfo->width / 8;
+    planar = av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt);
+
+    g_return_val_if_fail (ffmpegdec->info.layout == (planar ?
+            GST_AUDIO_LAYOUT_NON_INTERLEAVED : GST_AUDIO_LAYOUT_INTERLEAVED),
+        GST_FLOW_NOT_NEGOTIATED);
+
+    GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
 
     /* ffmpegdec->frame->linesize[0] might contain padding, allocate only what's needed */
     output_size = nsamples * byte_per_sample * channels;
 
-    GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
-    if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)
-        && channels > 1) {
-      gint i, j;
-      GstMapInfo minfo;
-
-      /* note: linesize[0] might contain padding, allocate only what's needed */
-      *outbuf =
-          gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
-          (ffmpegdec), output_size);
-
-      gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
-
-      switch (ffmpegdec->info.finfo->width) {
-        case 8:{
-          guint8 *odata = minfo.data;
-
-          for (i = 0; i < nsamples; i++) {
-            for (j = 0; j < channels; j++) {
-              odata[j] =
-                  ((const guint8 *) ffmpegdec->frame->extended_data[j])[i];
-            }
-            odata += channels;
-          }
-          break;
-        }
-        case 16:{
-          guint16 *odata = (guint16 *) minfo.data;
-
-          for (i = 0; i < nsamples; i++) {
-            for (j = 0; j < channels; j++) {
-              odata[j] =
-                  ((const guint16 *) ffmpegdec->frame->extended_data[j])[i];
-            }
-            odata += channels;
-          }
-          break;
-        }
-        case 32:{
-          guint32 *odata = (guint32 *) minfo.data;
-
-          for (i = 0; i < nsamples; i++) {
-            for (j = 0; j < channels; j++) {
-              odata[j] =
-                  ((const guint32 *) ffmpegdec->frame->extended_data[j])[i];
-            }
-            odata += channels;
-          }
-          break;
-        }
-        case 64:{
-          guint64 *odata = (guint64 *) minfo.data;
-
-          for (i = 0; i < nsamples; i++) {
-            for (j = 0; j < channels; j++) {
-              odata[j] =
-                  ((const guint64 *) ffmpegdec->frame->extended_data[j])[i];
-            }
-            odata += channels;
-          }
-          break;
-        }
-        default:
-          g_assert_not_reached ();
-          break;
+    *outbuf =
+        gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
+        (ffmpegdec), output_size);
+
+    if (planar) {
+      gint i;
+      GstAudioMeta *meta;
+
+      meta = gst_buffer_add_audio_meta (*outbuf, &ffmpegdec->info, nsamples,
+          NULL);
+
+      for (i = 0; i < channels; i++) {
+        gst_buffer_fill (*outbuf, meta->offsets[i],
+            ffmpegdec->frame->extended_data[i], nsamples * byte_per_sample);
       }
-      gst_buffer_unmap (*outbuf, &minfo);
     } else {
-      *outbuf =
-          gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
-          (ffmpegdec), output_size);
       gst_buffer_fill (*outbuf, 0, ffmpegdec->frame->data[0], output_size);
     }
 

diff --git a/ext/libav/gstavcodecmap.c b/ext/libav/gstavcodecmap.c
index 3315ccb..19fef71 100644 (file)
--- a/ext/libav/gstavcodecmap.c
+++ b/ext/libav/gstavcodecmap.c
@@ -474,11 +474,14 @@ get_nbits_set (guint64 n)
 
 static void
 gst_ffmpeg_audio_set_sample_fmts (GstCaps * caps,
-    const enum AVSampleFormat *fmts)
+    const enum AVSampleFormat *fmts, gboolean always_interleaved)
 {
   GValue va = { 0, };
+  GValue vap = { 0, };
   GValue v = { 0, };
   GstAudioFormat format;
+  GstAudioLayout layout;
+  GstCaps *caps_copy = NULL;
 
   if (!fmts || fmts[0] == -1) {
     gint i;
@@ -486,38 +489,73 @@ gst_ffmpeg_audio_set_sample_fmts (GstCaps * caps,
     g_value_init (&va, GST_TYPE_LIST);
     g_value_init (&v, G_TYPE_STRING);
     for (i = 0; i <= AV_SAMPLE_FMT_DBL; i++) {
-      format = gst_ffmpeg_smpfmt_to_audioformat (i);
+      format = gst_ffmpeg_smpfmt_to_audioformat (i, NULL);
       if (format == GST_AUDIO_FORMAT_UNKNOWN)
         continue;
       g_value_set_string (&v, gst_audio_format_to_string (format));
       gst_value_list_append_value (&va, &v);
     }
     gst_caps_set_value (caps, "format", &va);
+    if (!always_interleaved) {
+      g_value_init (&vap, GST_TYPE_LIST);
+      g_value_set_string (&v, "interleaved");
+      gst_value_list_append_value (&vap, &v);
+      g_value_set_string (&v, "non-interleaved");
+      gst_value_list_append_value (&vap, &v);
+      gst_caps_set_value (caps, "layout", &vap);
+      g_value_unset (&vap);
+    } else {
+      gst_caps_set_simple (caps, "layout", G_TYPE_STRING, "interleaved", NULL);
+    }
     g_value_unset (&v);
     g_value_unset (&va);
     return;
   }
 
   g_value_init (&va, GST_TYPE_LIST);
+  g_value_init (&vap, GST_TYPE_LIST);
   g_value_init (&v, G_TYPE_STRING);
   while (*fmts != -1) {
-    format = gst_ffmpeg_smpfmt_to_audioformat (*fmts);
+    format = gst_ffmpeg_smpfmt_to_audioformat (*fmts, &layout);
     if (format != GST_AUDIO_FORMAT_UNKNOWN) {
       g_value_set_string (&v, gst_audio_format_to_string (format));
       /* Only append values we don't have yet */
-      if (!_gst_value_list_contains (&va, &v))
-        gst_value_list_append_value (&va, &v);
+      if (layout == GST_AUDIO_LAYOUT_INTERLEAVED || always_interleaved) {
+        if (!_gst_value_list_contains (&va, &v))
+          gst_value_list_append_value (&va, &v);
+      } else {
+        if (!_gst_value_list_contains (&vap, &v))
+          gst_value_list_append_value (&vap, &v);
+      }
     }
     fmts++;
   }
+  if (gst_value_list_get_size (&va) >= 1 && gst_value_list_get_size (&vap) >= 1) {
+    caps_copy = gst_caps_copy (caps);
+  }
   if (gst_value_list_get_size (&va) == 1) {
-    /* The single value is still in v */
-    gst_caps_set_value (caps, "format", &v);
+    gst_caps_set_value (caps, "format", gst_value_list_get_value (&va, 0));
+    gst_caps_set_simple (caps, "layout", G_TYPE_STRING, "interleaved", NULL);
   } else if (gst_value_list_get_size (&va) > 1) {
     gst_caps_set_value (caps, "format", &va);
+    gst_caps_set_simple (caps, "layout", G_TYPE_STRING, "interleaved", NULL);
+  }
+  if (gst_value_list_get_size (&vap) == 1) {
+    gst_caps_set_value (caps_copy ? caps_copy : caps, "format",
+        gst_value_list_get_value (&vap, 0));
+    gst_caps_set_simple (caps_copy ? caps_copy : caps, "layout", G_TYPE_STRING,
+        "non-interleaved", NULL);
+  } else if (gst_value_list_get_size (&vap) > 1) {
+    gst_caps_set_value (caps_copy ? caps_copy : caps, "format", &vap);
+    gst_caps_set_simple (caps_copy ? caps_copy : caps, "layout", G_TYPE_STRING,
+        "non-interleaved", NULL);
+  }
+  if (caps_copy) {
+    gst_caps_append (caps, caps_copy);
   }
   g_value_unset (&v);
   g_value_unset (&va);
+  g_value_unset (&vap);
 }
 
 /* same for audio - now with channels/sample rate
@@ -2333,29 +2371,47 @@ gst_ffmpeg_pixfmt_to_caps (enum AVPixelFormat pix_fmt, AVCodecContext * context,
 }
 
 GstAudioFormat
-gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt)
+gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt,
+    GstAudioLayout * layout)
 {
+  if (layout)
+    *layout = GST_AUDIO_LAYOUT_NON_INTERLEAVED;
+
   switch (sample_fmt) {
     case AV_SAMPLE_FMT_U8:
+      if (layout)
+        *layout = GST_AUDIO_LAYOUT_INTERLEAVED;
     case AV_SAMPLE_FMT_U8P:
       return GST_AUDIO_FORMAT_U8;
       break;
+
     case AV_SAMPLE_FMT_S16:
+      if (layout)
+        *layout = GST_AUDIO_LAYOUT_INTERLEAVED;
     case AV_SAMPLE_FMT_S16P:
       return GST_AUDIO_FORMAT_S16;
       break;
+
     case AV_SAMPLE_FMT_S32:
+      if (layout)
+        *layout = GST_AUDIO_LAYOUT_INTERLEAVED;
     case AV_SAMPLE_FMT_S32P:
       return GST_AUDIO_FORMAT_S32;
       break;
     case AV_SAMPLE_FMT_FLT:
+      if (layout)
+        *layout = GST_AUDIO_LAYOUT_INTERLEAVED;
     case AV_SAMPLE_FMT_FLTP:
       return GST_AUDIO_FORMAT_F32;
       break;
+
     case AV_SAMPLE_FMT_DBL:
+      if (layout)
+        *layout = GST_AUDIO_LAYOUT_INTERLEAVED;
     case AV_SAMPLE_FMT_DBLP:
       return GST_AUDIO_FORMAT_F64;
       break;
+
     default:
       /* .. */
       return GST_AUDIO_FORMAT_UNKNOWN;
@@ -2376,13 +2432,16 @@ gst_ffmpeg_smpfmt_to_caps (enum AVSampleFormat sample_fmt,
 {
   GstCaps *caps = NULL;
   GstAudioFormat format;
+  GstAudioLayout layout;
 
-  format = gst_ffmpeg_smpfmt_to_audioformat (sample_fmt);
+  format = gst_ffmpeg_smpfmt_to_audioformat (sample_fmt, &layout);
 
   if (format != GST_AUDIO_FORMAT_UNKNOWN) {
     caps = gst_ff_aud_caps_new (context, codec, codec_id, TRUE, "audio/x-raw",
         "format", G_TYPE_STRING, gst_audio_format_to_string (format),
-        "layout", G_TYPE_STRING, "interleaved", NULL);
+        "layout", G_TYPE_STRING,
+        (layout == GST_AUDIO_LAYOUT_INTERLEAVED) ?
+        "interleaved" : "non-interleaved", NULL);
     GST_LOG ("caps for sample_fmt=%d: %" GST_PTR_FORMAT, sample_fmt, caps);
   } else {
     GST_LOG ("No caps found for sample_fmt=%d", sample_fmt);
@@ -2426,10 +2485,10 @@ gst_ffmpeg_codectype_to_audio_caps (AVCodecContext * context,
         codec_id);
   } else {
     caps = gst_ff_aud_caps_new (context, codec, codec_id, encode, "audio/x-raw",
-        "layout", G_TYPE_STRING, "interleaved", NULL);
+        NULL);
     if (!caps_has_field (caps, "format"))
       gst_ffmpeg_audio_set_sample_fmts (caps,
-          codec ? codec->sample_fmts : NULL);
+          codec ? codec->sample_fmts : NULL, encode);
   }
 
   return caps;
@@ -2470,6 +2529,8 @@ gst_ffmpeg_caps_to_smpfmt (const GstCaps * caps,
   const gchar *fmt;
   GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
   gint bitrate;
+  const gchar *layout;
+  gboolean interleaved;
 
   g_return_if_fail (gst_caps_get_size (caps) == 1);
 
@@ -2490,18 +2551,25 @@ gst_ffmpeg_caps_to_smpfmt (const GstCaps * caps,
     }
   }
 
+  layout = gst_structure_get_string (structure, "layout");
+  interleaved = ! !g_strcmp0 (layout, "non-interleaved");
+
   switch (format) {
     case GST_AUDIO_FORMAT_F32:
-      context->sample_fmt = AV_SAMPLE_FMT_FLT;
+      context->sample_fmt =
+          interleaved ? AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_FLTP;
       break;
     case GST_AUDIO_FORMAT_F64:
-      context->sample_fmt = AV_SAMPLE_FMT_DBL;
+      context->sample_fmt =
+          interleaved ? AV_SAMPLE_FMT_DBL : AV_SAMPLE_FMT_DBLP;
       break;
     case GST_AUDIO_FORMAT_S32:
-      context->sample_fmt = AV_SAMPLE_FMT_S32;
+      context->sample_fmt =
+          interleaved ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S32P;
       break;
     case GST_AUDIO_FORMAT_S16:
-      context->sample_fmt = AV_SAMPLE_FMT_S16;
+      context->sample_fmt =
+          interleaved ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P;
       break;
     default:
       break;

diff --git a/ext/libav/gstavcodecmap.h b/ext/libav/gstavcodecmap.h
index 14cdfce..56975b6 100644 (file)
--- a/ext/libav/gstavcodecmap.h
+++ b/ext/libav/gstavcodecmap.h
@@ -141,7 +141,8 @@ gst_ffmpeg_audioinfo_to_context (GstAudioInfo *info,
 GstVideoFormat gst_ffmpeg_pixfmt_to_videoformat (enum AVPixelFormat pixfmt);
 enum AVPixelFormat gst_ffmpeg_videoformat_to_pixfmt (GstVideoFormat format);
 
-GstAudioFormat gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt);
+GstAudioFormat gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt,
+                                                 GstAudioLayout * layout);
 
 /*
  * _formatid_to_caps () is meant for muxers/demuxers, it