Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 191 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
new file mode 100644
index 0000000000..30a6a1c303
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2012 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 <stdlib.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
+#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
+
+namespace {
+
+const unsigned char kPayloadType = 100;
+
+};
+
+namespace webrtc {
+
+class RtpRtcpVideoTest : public ::testing::Test {
+ protected:
+  RtpRtcpVideoTest()
+      : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
+        test_ssrc_(3456),
+        test_timestamp_(4567),
+        test_sequence_number_(2345),
+        fake_clock(123456) {}
+  ~RtpRtcpVideoTest() {}
+
+  virtual void SetUp() {
+    transport_ = new LoopBackTransport();
+    receiver_ = new TestRtpReceiver();
+    receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
+    RtpRtcp::Configuration configuration;
+    configuration.audio = false;
+    configuration.clock = &fake_clock;
+    configuration.outgoing_transport = transport_;
+
+    video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
+    rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
+        &fake_clock, receiver_, NULL, &rtp_payload_registry_));
+
+    video_module_->SetRTCPStatus(RtcpMode::kCompound);
+    video_module_->SetSSRC(test_ssrc_);
+    rtp_receiver_->SetNACKStatus(kNackRtcp);
+    video_module_->SetStorePacketsStatus(true, 600);
+    EXPECT_EQ(0, video_module_->SetSendingStatus(true));
+
+    transport_->SetSendModule(video_module_, &rtp_payload_registry_,
+                              rtp_receiver_.get(), receive_statistics_.get());
+
+    VideoCodec video_codec;
+    memset(&video_codec, 0, sizeof(video_codec));
+    video_codec.plType = 123;
+    memcpy(video_codec.plName, "I420", 5);
+
+    EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
+    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
+                                                       video_codec.plType,
+                                                       90000,
+                                                       0,
+                                                       video_codec.maxBitrate));
+
+    payload_data_length_ = sizeof(video_frame_);
+
+    for (size_t n = 0; n < payload_data_length_; n++) {
+      video_frame_[n] = n%10;
+    }
+  }
+
+  size_t BuildRTPheader(uint8_t* dataBuffer,
+                         uint32_t timestamp,
+                         uint32_t sequence_number) {
+    dataBuffer[0] = static_cast<uint8_t>(0x80);  // version 2
+    dataBuffer[1] = static_cast<uint8_t>(kPayloadType);
+    ByteWriter<uint16_t>::WriteBigEndian(dataBuffer + 2, sequence_number);
+    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 4, timestamp);
+    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 8, 0x1234);  // SSRC.
+    size_t rtpHeaderLength = 12;
+    return rtpHeaderLength;
+  }
+
+  size_t PaddingPacket(uint8_t* buffer,
+                       uint32_t timestamp,
+                       uint32_t sequence_number,
+                       size_t bytes) {
+    // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
+    size_t max_length = 224;
+
+    size_t padding_bytes_in_packet = max_length;
+    if (bytes < max_length) {
+      padding_bytes_in_packet = (bytes + 16) & 0xffe0;  // Keep our modulus 32.
+    }
+    // Correct seq num, timestamp and payload type.
+    size_t header_length = BuildRTPheader(buffer, timestamp, sequence_number);
+    buffer[0] |= 0x20;  // Set padding bit.
+    int32_t* data =
+        reinterpret_cast<int32_t*>(&(buffer[header_length]));
+
+    // Fill data buffer with random data.
+    for (size_t j = 0; j < (padding_bytes_in_packet >> 2); j++) {
+      data[j] = rand();  // NOLINT
+    }
+    // Set number of padding bytes in the last byte of the packet.
+    buffer[header_length + padding_bytes_in_packet - 1] =
+        padding_bytes_in_packet;
+    return padding_bytes_in_packet + header_length;
+  }
+
+  virtual void TearDown() {
+    delete video_module_;
+    delete transport_;
+    delete receiver_;
+  }
+
+  int test_id_;
+  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  RTPPayloadRegistry rtp_payload_registry_;
+  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  RtpRtcp* video_module_;
+  LoopBackTransport* transport_;
+  TestRtpReceiver* receiver_;
+  uint32_t test_ssrc_;
+  uint32_t test_timestamp_;
+  uint16_t test_sequence_number_;
+  uint8_t  video_frame_[65000];
+  size_t payload_data_length_;
+  SimulatedClock fake_clock;
+};
+
+TEST_F(RtpRtcpVideoTest, BasicVideo) {
+  uint32_t timestamp = 3000;
+  EXPECT_EQ(0, video_module_->SendOutgoingData(kVideoFrameDelta, 123,
+                                               timestamp,
+                                               timestamp / 90,
+                                               video_frame_,
+                                               payload_data_length_));
+}
+
+TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
+  const size_t kPadSize = 255;
+  uint8_t padding_packet[kPadSize];
+  uint32_t seq_num = 0;
+  uint32_t timestamp = 3000;
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP8;
+  codec.plType = kPayloadType;
+  strncpy(codec.plName, "VP8", 4);
+  EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(codec.plName,
+                                                     codec.plType,
+                                                     90000,
+                                                     0,
+                                                     codec.maxBitrate));
+  for (int frame_idx = 0; frame_idx < 10; ++frame_idx) {
+    for (int packet_idx = 0; packet_idx < 5; ++packet_idx) {
+      size_t packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
+                                         kPadSize);
+      ++seq_num;
+      RTPHeader header;
+      rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+      EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header));
+      PayloadUnion payload_specific;
+      EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
+                                                            &payload_specific));
+      const uint8_t* payload = padding_packet + header.headerLength;
+      const size_t payload_length = packet_size - header.headerLength;
+      EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, payload,
+                                                   payload_length,
+                                                   payload_specific, true));
+      EXPECT_EQ(0u, receiver_->payload_size());
+      EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
+    }
+    timestamp += 3000;
+    fake_clock.AdvanceTimeMilliseconds(33);
+  }
+}
+
+}  // namespace webrtc