From 228d8808fe6f8aa7390b73d2b248b3a8f092eb9e Mon Sep 17 00:00:00 2001
From: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
Date: Wed, 7 Mar 2012 12:59:28 +0100
Subject: [PATCH] opusenc: only request and process 1 frame at a time

... since it is specified in _finish_frame that input buffer may be invalidated
after calling it, and is as such not reliably available for further encoding.

Also, requesting or allowing several frames is only useful if subclass intends
to process these "in 1 run" (as in, 1 output buffer), not for having another
(inner) loop in subclass where the baseclass one will do just fine.
---
 ext/opus/gstopusenc.c | 72 +++++++++++++++++++------------------------
 1 file changed, 32 insertions(+), 40 deletions(-)

diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c
index a87e08b3f7..6a19ffd89f 100644
--- a/ext/opus/gstopusenc.c
+++ b/ext/opus/gstopusenc.c
@@ -384,7 +384,7 @@ gst_opus_enc_setup_base_class (GstOpusEnc * enc, GstAudioEncoder * benc)
       gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
   gst_audio_encoder_set_frame_samples_min (benc, enc->frame_samples);
   gst_audio_encoder_set_frame_samples_max (benc, enc->frame_samples);
-  gst_audio_encoder_set_frame_max (benc, 0);
+  gst_audio_encoder_set_frame_max (benc, 1);
 }
 
 static gint
@@ -792,6 +792,8 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
   gsize bsize, size;
   gsize bytes;
   gint ret = GST_FLOW_OK;
+  gint outsize;
+  GstBuffer *outbuf;
 
   g_mutex_lock (enc->property_lock);
 
@@ -816,52 +818,42 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
     goto done;
   }
 
+  g_assert (size == bytes);
 
-  while (size) {
-    gint outsize;
-    GstBuffer *outbuf;
+  ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
+      GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
+      GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
 
-    ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
-        GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
-        GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
+  if (GST_FLOW_OK != ret)
+    goto done;
 
-    if (GST_FLOW_OK != ret)
-      goto done;
+  GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
+      enc->frame_samples, (int) bytes);
 
-    GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
-        enc->frame_samples, (int) bytes);
+  outsize =
+      opus_multistream_encode (enc->state, (const gint16 *) data,
+      enc->frame_samples, GST_BUFFER_DATA (outbuf),
+      enc->max_payload_size * enc->n_channels);
 
-    outsize =
-        opus_multistream_encode (enc->state, (const gint16 *) data,
-        enc->frame_samples, GST_BUFFER_DATA (outbuf),
-        enc->max_payload_size * enc->n_channels);
-
-    if (outsize < 0) {
-      GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
-      ret = GST_FLOW_ERROR;
-      goto done;
-    } else if (outsize > enc->max_payload_size) {
-      GST_WARNING_OBJECT (enc,
-          "Encoded size %d is higher than max payload size (%d bytes)",
-          outsize, enc->max_payload_size);
-      ret = GST_FLOW_ERROR;
-      goto done;
-    }
-
-    GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
-    GST_BUFFER_SIZE (outbuf) = outsize;
-
-    ret =
-        gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
-        enc->frame_samples);
-
-    if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
-      goto done;
-
-    data += bytes;
-    size -= bytes;
+  if (outsize < 0) {
+    GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
+    ret = GST_FLOW_ERROR;
+    goto done;
+  } else if (outsize > enc->max_payload_size) {
+    GST_WARNING_OBJECT (enc,
+        "Encoded size %d is higher than max payload size (%d bytes)",
+        outsize, enc->max_payload_size);
+    ret = GST_FLOW_ERROR;
+    goto done;
   }
 
+  GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
+  GST_BUFFER_SIZE (outbuf) = outsize;
+
+  ret =
+      gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
+      enc->frame_samples);
+
 done:
 
   g_mutex_unlock (enc->property_lock);