Enabling spatial layers in VP9Impl. Filter layers in the loopback test.

webrtc/test/layer_filtering_transport.cc

Index: webrtc/test/layer_filtering_transport.cc
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
new file mode 100644
index 0000000000000000000000000000000000000000..999baef9fb9ed7843999286948c3ee10a050b706
--- /dev/null
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/layer_filtering_transport.h"
+
+namespace webrtc {
+namespace test {
+
+LayerFilteringTransport::LayerFilteringTransport(
+    const FakeNetworkPipe::Config& config,
+    uint8_t vp8_video_payload_type,
+    uint8_t vp9_video_payload_type,
+    uint8_t tl_discard_threshold,
+    uint8_t sl_discard_threshold)
+    : test::DirectTransport(config),
+      vp8_video_payload_type_(vp8_video_payload_type),
+      vp9_video_payload_type_(vp9_video_payload_type),
+      tl_discard_threshold_(tl_discard_threshold),
+      sl_discard_threshold_(sl_discard_threshold),
+      current_seq_num_(10000) {
+}  // TODO(ivica): random seq num?
+
+bool LayerFilteringTransport::SendRtp(const uint8_t* packet, size_t length) {
+  bool set_marker_bit = false;
+  rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+  RTPHeader header;
+  parser->Parse(packet, length, &header);
+
+  if ((tl_discard_threshold_ > 0 || sl_discard_threshold_ > 0) &&

[sprang_webrtc, 2015/08/13 13:52:07]

Can't we just call test::DirectTransport::SendRtp(new_packet, length) and return
early if this is the case?

[ivica, 2015/08/13 15:20:30]

I've changed the code to immediately call SendRtp if both tl_discard_threshold
and sl_discard_threshold are zero. Otherwise, we must fix the sequence number,
so we have to process the packet.

+      (header.payloadType == vp8_video_payload_type_ ||
+       header.payloadType == vp9_video_payload_type_)) {
+    const uint8_t* payload = packet + header.headerLength;
+    DCHECK_GT(length, header.headerLength);
+    const size_t payload_length = length - header.headerLength;
+    DCHECK_GT(payload_length, header.paddingLength);
+    const size_t payload_data_length = payload_length - header.paddingLength;
+
+    const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
+    rtc::scoped_ptr<RtpDepacketizer> depacketizer(
+        RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
+    RtpDepacketizer::ParsedPayload parsed_payload;
+    if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
+      const uint8_t temporalIdx =
+          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
+                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx;
+      const uint8_t spatialIdx =
+          is_vp8 ? kNoSpatialIdx
+                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx;
+      if (sl_discard_threshold_ > 0 &&
+          spatialIdx == sl_discard_threshold_ - 1 &&
+          parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
+        // This layer is now the last in the superframe.
+        set_marker_bit = true;
+      }
+
+      if ((tl_discard_threshold_ > 0 && temporalIdx != kNoTemporalIdx &&
+           temporalIdx >= tl_discard_threshold_) ||
+          (sl_discard_threshold_ > 0 && spatialIdx != kNoSpatialIdx &&
+           spatialIdx >= sl_discard_threshold_)) {
+        return true;  // Discard the packet.
+      }
+    } else {
+      RTC_NOTREACHED() << "Parse error";
+    }
+  }
+
+  uint8_t* new_packet = new uint8_t[length];

[sprang_webrtc, 2015/08/13 13:52:07]

Stack allocate a temporary instead.
uint8_t temp_buffer[IP_PACKET_SIZE];

[ivica, 2015/08/13 15:20:30]

Done.

+  memcpy(new_packet, packet, length);
+
+  if (set_marker_bit) {
+    new_packet[1] |= kRtpMarkerBitMask;
+  }

[sprang_webrtc, 2015/08/13 13:52:07]

ByteWriter<uint16_t>::WriteBigEndian(&new_packet[2], current_seq_num_);
Also add a comment why you're updating the headers here.

[ivica, 2015/08/13 15:20:30]

Done.

+  new_packet[2] = (uint8_t)(current_seq_num_ >> 8);
+  new_packet[3] = (uint8_t)current_seq_num_;
+
+  ++current_seq_num_;  // Increase only if packet not discarded.
+
+  bool result = test::DirectTransport::SendRtp(new_packet, length);
+  delete[] new_packet;
+  return result;
+}
+
+}  // namespace test
+}  // namespace webrtc