opusdec: rewrite logic

Parameters such as frame size, etc, are variable. Pretty much
everything can change within a stream, so be prepared about it,
and do not cache parameters in the decoder.

diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c
index 0b30c38..d3528c8 100644 (file)
--- a/ext/opus/gstopusdec.c
+++ b/ext/opus/gstopusdec.c
@@ -48,9 +48,6 @@
 GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
 #define GST_CAT_DEFAULT opusdec_debug
 
-#define DEC_MAX_FRAME_SIZE 2000
-#define DEC_MAX_OUTPUT_BUFFER_SIZE (5760)
-
 static GstStaticPadTemplate opus_dec_src_factory =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
@@ -110,14 +107,13 @@ static void
 gst_opus_dec_reset (GstOpusDec * dec)
 {
   dec->packetno = 0;
-  dec->frame_size = 0;
-  dec->frame_samples = 960;
-  dec->frame_duration = 0;
   if (dec->state) {
     opus_decoder_destroy (dec->state);
     dec->state = NULL;
   }
 
+  dec->next_ts = 0;
+
   gst_buffer_replace (&dec->streamheader, NULL);
   gst_buffer_replace (&dec->vorbiscomment, NULL);
 }
@@ -176,7 +172,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
   GstBuffer *outbuf;
   gint16 *out_data;
   int n, err;
-  int samples_per_frame;
+  int samples;
   unsigned int packet_size;
 
   if (dec->state == NULL) {
@@ -192,8 +188,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
         "rate", G_TYPE_INT, dec->sample_rate,
         "channels", G_TYPE_INT, dec->n_channels, NULL);
 
-    GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
-        dec->sample_rate, dec->n_channels, dec->frame_size);
+    GST_DEBUG_OBJECT (dec, "rate=%d channels=%d",
+        dec->sample_rate, dec->n_channels);
 
     if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
       GST_ERROR ("nego failure");
@@ -214,12 +210,13 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
     size = 0;
   }
 
-  samples_per_frame =
-      opus_packet_get_samples_per_frame (data, dec->sample_rate);
+  samples =
+      opus_packet_get_samples_per_frame (data,
+      dec->sample_rate) * opus_packet_get_nb_frames (data, size);
   GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
-  GST_DEBUG ("samples_per_frame %d", samples_per_frame);
+  GST_DEBUG ("samples %d", samples);
 
-  packet_size = samples_per_frame * dec->n_channels * 2;
+  packet_size = samples * dec->n_channels * 2;
   outbuf = gst_buffer_new_and_alloc (packet_size);
   if (!outbuf) {
     goto buffer_failed;
@@ -227,34 +224,33 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
 
   out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
 
-  GST_LOG_OBJECT (dec, "decoding frame");
+  GST_LOG_OBJECT (dec, "decoding %d samples, in size %u", samples, size);
 
-  n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0);
+  n = opus_decode (dec->state, data, size, out_data, samples, 0);
   gst_buffer_unmap (buf, data, size);
+  gst_buffer_unmap (outbuf, out_data, out_size);
   if (n < 0) {
-    gst_buffer_unmap (outbuf, out_data, out_size);
     GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
     return GST_FLOW_ERROR;
   }
+  GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
 
-  if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
-    GST_WARNING_OBJECT (dec, "No timestamp in -> no timestamp out");
+  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+    GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+  } else {
+    GST_BUFFER_TIMESTAMP (outbuf) = dec->next_ts;
   }
 
-  GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
-      GST_TIME_ARGS (timestamp));
-
-  GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
-  GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
+  GST_BUFFER_DURATION (outbuf) =
+      gst_util_uint64_scale (n, GST_SECOND, dec->sample_rate);
+  dec->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
 
   GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
       GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
-      GST_TIME_ARGS (dec->frame_duration));
+      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
 
   res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
 
-  gst_buffer_unmap (outbuf, out_data, out_size);
-
   if (res != GST_FLOW_OK)
     GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
 
@@ -269,37 +265,6 @@ buffer_failed:
   return GST_FLOW_ERROR;
 }
 
-static gint
-gst_opus_dec_get_frame_samples (GstOpusDec * dec)
-{
-  gint frame_samples = 0;
-  switch (dec->frame_size) {
-    case 2:
-      frame_samples = dec->sample_rate / 400;
-      break;
-    case 5:
-      frame_samples = dec->sample_rate / 200;
-      break;
-    case 10:
-      frame_samples = dec->sample_rate / 100;
-      break;
-    case 20:
-      frame_samples = dec->sample_rate / 50;
-      break;
-    case 40:
-      frame_samples = dec->sample_rate / 25;
-      break;
-    case 60:
-      frame_samples = 3 * dec->sample_rate / 50;
-      break;
-    default:
-      GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
-      frame_samples = 0;
-      break;
-  }
-  return frame_samples;
-}
-
 static gboolean
 gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
 {
@@ -337,25 +302,6 @@ gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
     }
   }
 
-  if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
-    GST_WARNING_OBJECT (dec, "Frame size not included in caps");
-  }
-  if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
-    GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
-  }
-  if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
-    GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
-  }
-
-  dec->frame_samples = gst_opus_dec_get_frame_samples (dec);
-  dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
-      GST_SECOND, dec->sample_rate);
-
-  GST_INFO_OBJECT (dec,
-      "Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
-      GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
-      dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
-
   caps = gst_caps_new_simple ("audio/x-raw",
       "format", G_TYPE_STRING, "S16LE",
       "rate", G_TYPE_INT, dec->sample_rate,
@@ -387,6 +333,20 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
   return res;
 }
 
+static gboolean
+gst_opus_dec_is_header (GstBuffer * buf, const char *magic, guint magic_size)
+{
+  guint8 *data;
+  gsize size;
+  gboolean ret;
+
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+  if (!data)
+    return FALSE;
+  ret = (size >= magic_size && !memcmp (magic, data, magic_size));
+  gst_buffer_unmap (buf, data, size);
+  return ret;
+}
 
 static GstFlowReturn
 gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
@@ -424,16 +384,24 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
      * first two packets are the headers. */
     switch (dec->packetno) {
       case 0:
-        GST_DEBUG_OBJECT (dec, "counted streamheader");
-        res = GST_FLOW_OK;
-        res = gst_opus_dec_parse_header (dec, buf);
-        gst_audio_decoder_finish_frame (adec, NULL, 1);
+        if (gst_opus_dec_is_header (buf, "OpusHead", 8)) {
+          GST_DEBUG_OBJECT (dec, "found streamheader");
+          res = gst_opus_dec_parse_header (dec, buf);
+          gst_audio_decoder_finish_frame (adec, NULL, 1);
+        } else {
+          res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+              GST_BUFFER_DURATION (buf));
+        }
         break;
       case 1:
-        GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
-        res = GST_FLOW_OK;
-        res = gst_opus_dec_parse_comments (dec, buf);
-        gst_audio_decoder_finish_frame (adec, NULL, 1);
+        if (gst_opus_dec_is_header (buf, "OpusTags", 8)) {
+          GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
+          res = gst_opus_dec_parse_comments (dec, buf);
+          gst_audio_decoder_finish_frame (adec, NULL, 1);
+        } else {
+          res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+              GST_BUFFER_DURATION (buf));
+        }
         break;
       default:
       {

diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h
index 8389580..38dd279 100644 (file)
--- a/ext/opus/gstopusdec.h
+++ b/ext/opus/gstopusdec.h
@@ -45,11 +45,9 @@ struct _GstOpusDec {
   GstAudioDecoder       element;
 
   OpusDecoder          *state;
-  int frame_samples;
 
-  gint                  frame_size;
-  GstClockTime          frame_duration;
   guint64               packetno;
+  GstClockTime          next_ts;
 
   GstBuffer            *streamheader;
   GstBuffer            *vorbiscomment;