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, 258 insertions, 0 deletions

diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
new file mode 100644
index 0000000000..88edc60d78
--- /dev/null
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -0,0 +1,258 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// This file contains fake implementations, for use in unit tests, of the
+// following classes:
+//
+//   webrtc::Call
+//   webrtc::AudioSendStream
+//   webrtc::AudioReceiveStream
+//   webrtc::VideoSendStream
+//   webrtc::VideoReceiveStream
+
+#ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
+#define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
+
+#include <vector>
+
+#include "webrtc/call.h"
+#include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio_send_stream.h"
+#include "webrtc/video_frame.h"
+#include "webrtc/video_receive_stream.h"
+#include "webrtc/video_send_stream.h"
+
+namespace cricket {
+
+class FakeAudioSendStream : public webrtc::AudioSendStream {
+ public:
+  explicit FakeAudioSendStream(
+      const webrtc::AudioSendStream::Config& config);
+
+  const webrtc::AudioSendStream::Config& GetConfig() const;
+  void SetStats(const webrtc::AudioSendStream::Stats& stats);
+
+ private:
+  // webrtc::SendStream implementation.
+  void Start() override {}
+  void Stop() override {}
+  void SignalNetworkState(webrtc::NetworkState state) override {}
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
+    return true;
+  }
+
+  // webrtc::AudioSendStream implementation.
+  webrtc::AudioSendStream::Stats GetStats() const override;
+
+  webrtc::AudioSendStream::Config config_;
+  webrtc::AudioSendStream::Stats stats_;
+};
+
+class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
+ public:
+  explicit FakeAudioReceiveStream(
+      const webrtc::AudioReceiveStream::Config& config);
+
+  const webrtc::AudioReceiveStream::Config& GetConfig() const;
+  void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
+  int received_packets() const { return received_packets_; }
+  void IncrementReceivedPackets();
+
+ private:
+  // webrtc::ReceiveStream implementation.
+  void Start() override {}
+  void Stop() override {}
+  void SignalNetworkState(webrtc::NetworkState state) override {}
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
+    return true;
+  }
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const webrtc::PacketTime& packet_time) override {
+    return true;
+  }
+
+  // webrtc::AudioReceiveStream implementation.
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
+  webrtc::AudioReceiveStream::Config config_;
+  webrtc::AudioReceiveStream::Stats stats_;
+  int received_packets_;
+};
+
+class FakeVideoSendStream : public webrtc::VideoSendStream,
+                            public webrtc::VideoCaptureInput {
+ public:
+  FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
+                      const webrtc::VideoEncoderConfig& encoder_config);
+  webrtc::VideoSendStream::Config GetConfig() const;
+  webrtc::VideoEncoderConfig GetEncoderConfig() const;
+  std::vector<webrtc::VideoStream> GetVideoStreams();
+
+  bool IsSending() const;
+  bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
+  bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
+
+  int GetNumberOfSwappedFrames() const;
+  int GetLastWidth() const;
+  int GetLastHeight() const;
+  int64_t GetLastTimestamp() const;
+  void SetStats(const webrtc::VideoSendStream::Stats& stats);
+
+ private:
+  void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
+
+  // webrtc::SendStream implementation.
+  void Start() override;
+  void Stop() override;
+  void SignalNetworkState(webrtc::NetworkState state) override {}
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
+    return true;
+  }
+
+  // webrtc::VideoSendStream implementation.
+  webrtc::VideoSendStream::Stats GetStats() override;
+  bool ReconfigureVideoEncoder(
+      const webrtc::VideoEncoderConfig& config) override;
+  webrtc::VideoCaptureInput* Input() override;
+
+  bool sending_;
+  webrtc::VideoSendStream::Config config_;
+  webrtc::VideoEncoderConfig encoder_config_;
+  bool codec_settings_set_;
+  union VpxSettings {
+    webrtc::VideoCodecVP8 vp8;
+    webrtc::VideoCodecVP9 vp9;
+  } vpx_settings_;
+  int num_swapped_frames_;
+  webrtc::VideoFrame last_frame_;
+  webrtc::VideoSendStream::Stats stats_;
+};
+
+class FakeVideoReceiveStream : public webrtc::VideoReceiveStream {
+ public:
+  explicit FakeVideoReceiveStream(
+      const webrtc::VideoReceiveStream::Config& config);
+
+  webrtc::VideoReceiveStream::Config GetConfig();
+
+  bool IsReceiving() const;
+
+  void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms);
+
+  void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
+
+ private:
+  // webrtc::ReceiveStream implementation.
+  void Start() override;
+  void Stop() override;
+  void SignalNetworkState(webrtc::NetworkState state) override {}
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
+    return true;
+  }
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const webrtc::PacketTime& packet_time) override {
+    return true;
+  }
+
+  // webrtc::VideoReceiveStream implementation.
+  webrtc::VideoReceiveStream::Stats GetStats() const override;
+
+  webrtc::VideoReceiveStream::Config config_;
+  bool receiving_;
+  webrtc::VideoReceiveStream::Stats stats_;
+};
+
+class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
+ public:
+  explicit FakeCall(const webrtc::Call::Config& config);
+  ~FakeCall() override;
+
+  webrtc::Call::Config GetConfig() const;
+  const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
+  const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
+
+  const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
+  const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
+  const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
+  const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
+
+  rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
+  webrtc::NetworkState GetNetworkState() const;
+  int GetNumCreatedSendStreams() const;
+  int GetNumCreatedReceiveStreams() const;
+  void SetStats(const webrtc::Call::Stats& stats);
+
+ private:
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
+  webrtc::AudioReceiveStream* CreateAudioReceiveStream(
+      const webrtc::AudioReceiveStream::Config& config) override;
+  void DestroyAudioReceiveStream(
+      webrtc::AudioReceiveStream* receive_stream) override;
+
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      const webrtc::VideoSendStream::Config& config,
+      const webrtc::VideoEncoderConfig& encoder_config) override;
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
+
+  webrtc::VideoReceiveStream* CreateVideoReceiveStream(
+      const webrtc::VideoReceiveStream::Config& config) override;
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStream* receive_stream) override;
+  webrtc::PacketReceiver* Receiver() override;
+
+  DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const webrtc::PacketTime& packet_time) override;
+
+  webrtc::Call::Stats GetStats() const override;
+
+  void SetBitrateConfig(
+      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  void SignalNetworkState(webrtc::NetworkState state) override;
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+
+  webrtc::Call::Config config_;
+  webrtc::NetworkState network_state_;
+  rtc::SentPacket last_sent_packet_;
+  webrtc::Call::Stats stats_;
+  std::vector<FakeVideoSendStream*> video_send_streams_;
+  std::vector<FakeAudioSendStream*> audio_send_streams_;
+  std::vector<FakeVideoReceiveStream*> video_receive_streams_;
+  std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
+
+  int num_created_send_streams_;
+  int num_created_receive_streams_;
+};
+
+}  // namespace cricket
+#endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_