Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 25 insertions, 18 deletions

diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 603115cee7..d8fde670a0 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -38,19 +38,24 @@
 #include "talk/media/base/mediaengine.h"
 #include "talk/media/base/streamparams.h"
 #include "talk/media/base/videocapturer.h"
-#include "webrtc/p2p/base/transportcontroller.h"
-#include "webrtc/p2p/client/socketmonitor.h"
 #include "talk/session/media/audiomonitor.h"
 #include "talk/session/media/bundlefilter.h"
 #include "talk/session/media/mediamonitor.h"
 #include "talk/session/media/mediasession.h"
 #include "talk/session/media/rtcpmuxfilter.h"
 #include "talk/session/media/srtpfilter.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
+#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/p2p/client/socketmonitor.h"
+
+namespace webrtc {
+class AudioSinkInterface;
+}  // namespace webrtc
 
 namespace cricket {
 
@@ -174,8 +179,11 @@ class BaseChannel
   // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
   // true). Gets the transport channels from |transport_controller_|.
   bool SetTransport_w(const std::string& transport_name);
+
   void set_transport_channel(TransportChannel* transport);
-  void set_rtcp_transport_channel(TransportChannel* transport);
+  void set_rtcp_transport_channel(TransportChannel* transport,
+                                  bool update_writablity);
+
   bool was_ever_writable() const { return was_ever_writable_; }
   void set_local_content_direction(MediaContentDirection direction) {
     local_content_direction_ = direction;
@@ -213,6 +221,8 @@ class BaseChannel
                              int flags);
   void OnReadyToSend(TransportChannel* channel);
 
+  void OnDtlsState(TransportChannel* channel, DtlsTransportState state);
+
   bool PacketIsRtcp(const TransportChannel* channel, const char* data,
                     size_t len);
   bool SendPacket(bool rtcp,
@@ -235,8 +245,9 @@ class BaseChannel
   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
   bool SetupDtlsSrtp(bool rtcp_channel);
+  void MaybeSetupDtlsSrtp_w();
   // Set the DTLS-SRTP cipher policy on this channel as appropriate.
-  bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
+  bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);
 
   virtual void ChangeState() = 0;
 
@@ -282,9 +293,8 @@ class BaseChannel
   void OnMessage(rtc::Message* pmsg) override;
 
   // Handled in derived classes
-  // Get the SRTP ciphers to use for RTP media
-  virtual void GetSrtpCryptoSuiteNames(
-      std::vector<std::string>* ciphers) const = 0;
+  // Get the SRTP crypto suites to use for RTP media
+  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
   virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) = 0;
 
@@ -356,8 +366,6 @@ class VoiceChannel : public BaseChannel {
   // own ringing sound
   sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
 
-  // TODO(ronghuawu): Replace PressDTMF with InsertDtmf.
-  bool PressDTMF(int digit, bool playout);
   // Returns if the telephone-event has been negotiated.
   bool CanInsertDtmf();
   // Send and/or play a DTMF |event| according to the |flags|.
@@ -365,8 +373,11 @@ class VoiceChannel : public BaseChannel {
   // The |ssrc| should be either 0 or a valid send stream ssrc.
   // The valid value for the |event| are 0 which corresponding to DTMF
   // event 0-9, *, #, A-D.
-  bool InsertDtmf(uint32_t ssrc, int event_code, int duration, int flags);
+  bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
   bool SetOutputVolume(uint32_t ssrc, double volume);
+  void SetRawAudioSink(uint32_t ssrc,
+                       rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
+
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
@@ -402,12 +413,12 @@ class VoiceChannel : public BaseChannel {
                                   ContentAction action,
                                   std::string* error_desc);
   void HandleEarlyMediaTimeout();
-  bool InsertDtmf_w(uint32_t ssrc, int event, int duration, int flags);
+  bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
   bool GetStats_w(VoiceMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
@@ -456,8 +467,6 @@ class VideoChannel : public BaseChannel {
   // True if we've added a screencast.  Doesn't matter if the capturer
   // has been started or not.
   bool IsScreencasting();
-  int GetScreencastFps(uint32_t ssrc);
-  int GetScreencastMaxPixels(uint32_t ssrc);
   // Get statistics about the current media session.
   bool GetStats(VideoMediaInfo* stats);
 
@@ -476,7 +485,6 @@ class VideoChannel : public BaseChannel {
 
  private:
   typedef std::map<uint32_t, VideoCapturer*> ScreencastMap;
-  struct ScreencastDetailsData;
 
   // overrides from BaseChannel
   virtual void ChangeState();
@@ -493,11 +501,10 @@ class VideoChannel : public BaseChannel {
   bool RemoveScreencast_w(uint32_t ssrc);
   void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we);
   bool IsScreencasting_w() const;
-  void GetScreencastDetails_w(ScreencastDetailsData* d) const;
   bool GetStats_w(VideoMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
@@ -614,7 +621,7 @@ class DataChannel : public BaseChannel {
   virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(