Fixes use after free error when setting a new FrameEncryptor on ChannelSend.

This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be  deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.

To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.

ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.

Added Seth as TBR as this only introduces mocks.

TBR=shampson@webrtc.org

Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}

api/BUILD.gn

@@ -453,13 +453,50 @@ if (rtc_include_tests) {
     ]
   }
 
-  rtc_source_set("fake_frame_crypto") {
+  rtc_source_set("mock_frame_encryptor") {
+    testonly = true
+    sources = [
+      "test/mock_frame_encryptor.cc",
+      "test/mock_frame_encryptor.h",
+    ]
+    deps = [
+      ":libjingle_peerconnection_api",
+      "../test:test_support",
+    ]
+  }
+
+  rtc_source_set("mock_frame_decryptor") {
+    testonly = true
+    sources = [
+      "test/mock_frame_decryptor.cc",
+      "test/mock_frame_decryptor.h",
+    ]
+    deps = [
+      ":libjingle_peerconnection_api",
+      "../test:test_support",
+    ]
+  }
+
+  rtc_source_set("fake_frame_encryptor") {
+    testonly = true
+    sources = [
+      "test/fake_frame_encryptor.cc",
+      "test/fake_frame_encryptor.h",
+    ]
+    deps = [
+      ":array_view",
+      ":libjingle_peerconnection_api",
+      "..:webrtc_common",
+      "../rtc_base:checks",
+      "../rtc_base:rtc_base_approved",
+    ]
+  }
+
+  rtc_source_set("fake_frame_decryptor") {
     testonly = true
     sources = [
       "test/fake_frame_decryptor.cc",
       "test/fake_frame_decryptor.h",
-      "test/fake_frame_encryptor.cc",
-      "test/fake_frame_encryptor.h",
     ]
     deps = [
       ":array_view",

api/test/mock_frame_decryptor.cc

@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2018 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 "api/test/mock_frame_decryptor.h"
+
+namespace webrtc {
+
+MockFrameDecryptor::MockFrameDecryptor() = default;
+MockFrameDecryptor::~MockFrameDecryptor() = default;
+
+}  // namespace webrtc

api/test/mock_frame_decryptor.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2018 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.
+ */
+
+#ifndef API_TEST_MOCK_FRAME_DECRYPTOR_H_
+#define API_TEST_MOCK_FRAME_DECRYPTOR_H_
+
+#include <vector>
+
+#include "api/crypto/framedecryptorinterface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockFrameDecryptor : public FrameDecryptorInterface {
+ public:
+  MockFrameDecryptor();
+  ~MockFrameDecryptor() override;
+
+  MOCK_METHOD6(Decrypt,
+               int(cricket::MediaType,
+                   const std::vector<uint32_t>&,
+                   rtc::ArrayView<const uint8_t>,
+                   rtc::ArrayView<const uint8_t>,
+                   rtc::ArrayView<uint8_t>,
+                   size_t*));
+
+  MOCK_METHOD2(GetMaxPlaintextByteSize,
+               size_t(cricket::MediaType, size_t encrypted_frame_size));
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_FRAME_DECRYPTOR_H_

api/test/mock_frame_encryptor.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright 2018 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 "api/test/mock_frame_encryptor.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+MockFrameEncryptor::MockFrameEncryptor() = default;
+MockFrameEncryptor::~MockFrameEncryptor() = default;
+
+}  // namespace webrtc

api/test/mock_frame_encryptor.h

@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2018 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.
+ */
+
+#ifndef API_TEST_MOCK_FRAME_ENCRYPTOR_H_
+#define API_TEST_MOCK_FRAME_ENCRYPTOR_H_
+
+#include "api/crypto/frameencryptorinterface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockFrameEncryptor : public FrameEncryptorInterface {
+ public:
+  MockFrameEncryptor();
+  ~MockFrameEncryptor() override;
+
+  MOCK_METHOD6(Encrypt,
+               int(cricket::MediaType,
+                   uint32_t,
+                   rtc::ArrayView<const uint8_t>,
+                   rtc::ArrayView<const uint8_t>,
+                   rtc::ArrayView<uint8_t>,
+                   size_t*));
+
+  MOCK_METHOD2(GetMaxCiphertextByteSize,
+               size_t(cricket::MediaType media_type, size_t frame_size));
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_FRAME_ENCRYPTOR_H_

audio/BUILD.gn

@@ -135,6 +135,8 @@ if (rtc_include_tests) {
       ":audio",
       ":audio_end_to_end_test",
       "../api:mock_audio_mixer",
+      "../api:mock_frame_decryptor",
+      "../api:mock_frame_encryptor",
       "../api/audio:audio_frame_api",
       "../api/units:time_delta",
       "../call:mock_call_interfaces",

audio/audio_receive_stream_unittest.cc

@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "api/test/mock_audio_mixer.h"
+#include "api/test/mock_frame_decryptor.h"
 #include "audio/audio_receive_stream.h"
 #include "audio/conversion.h"
 #include "audio/mock_voe_channel_proxy.h"
@@ -373,5 +374,25 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
 
   recv_stream->Reconfigure(new_config);
 }
+
+TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
+  ConfigHelper helper;
+  auto recv_stream = helper.CreateAudioReceiveStream();
+
+  auto new_config_0 = helper.config();
+  rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
+      new rtc::RefCountedObject<MockFrameDecryptor>());
+  new_config_0.frame_decryptor = mock_frame_decryptor_0;
+
+  recv_stream->Reconfigure(new_config_0);
+
+  auto new_config_1 = helper.config();
+  rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
+      new rtc::RefCountedObject<MockFrameDecryptor>());
+  new_config_1.frame_decryptor = mock_frame_decryptor_1;
+  new_config_1.crypto_options.sframe.require_frame_encryption = true;
+  recv_stream->Reconfigure(new_config_1);
+}
+
 }  // namespace test
 }  // namespace webrtc

audio/audio_send_stream_unittest.cc

@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "api/test/mock_frame_encryptor.h"
 #include "api/units/time_delta.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
@@ -196,7 +197,7 @@ struct ConfigHelper {
     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
     EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
-    EXPECT_CALL(*channel_proxy_, SetFrameEncryptor(nullptr)).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetFrameEncryptor(_)).Times(1);
     EXPECT_CALL(*channel_proxy_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
         .Times(1);
@@ -528,6 +529,32 @@ TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
   send_stream->Reconfigure(new_config);
 }
 
+// Validates that reconfiguring the AudioSendStream with a Frame encryptor
+// correctly reconfigures on the object without crashing.
+TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
+  ConfigHelper helper(false, true);
+  auto send_stream = helper.CreateAudioSendStream();
+  auto new_config = helper.config();
+
+  rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0(
+      new rtc::RefCountedObject<MockFrameEncryptor>());
+  new_config.frame_encryptor = mock_frame_encryptor_0;
+  EXPECT_CALL(*helper.channel_proxy(), SetFrameEncryptor(Ne(nullptr))).Times(1);
+  send_stream->Reconfigure(new_config);
+
+  // Not updating the frame encryptor shouldn't force it to reconfigure.
+  EXPECT_CALL(*helper.channel_proxy(), SetFrameEncryptor(_)).Times(0);
+  send_stream->Reconfigure(new_config);
+
+  // Updating frame encryptor to a new object should force a call to the proxy.
+  rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1(
+      new rtc::RefCountedObject<MockFrameEncryptor>());
+  new_config.frame_encryptor = mock_frame_encryptor_1;
+  new_config.crypto_options.sframe.require_frame_encryption = true;
+  EXPECT_CALL(*helper.channel_proxy(), SetFrameEncryptor(Ne(nullptr))).Times(1);
+  send_stream->Reconfigure(new_config);
+}
+
 // Checks that AudioSendStream logs the times at which RTP packets are sent
 // through its interface.
 TEST(AudioSendStreamTest, UpdateLifetime) {

audio/channel_receive.cc

@@ -207,7 +207,7 @@ ChannelReceive::ChannelReceive(
     bool jitter_buffer_fast_playout,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
     absl::optional<AudioCodecPairId> codec_pair_id,
-    FrameDecryptorInterface* frame_decryptor,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
     const webrtc::CryptoOptions& crypto_options)
     : event_log_(rtc_event_log),
       rtp_receive_statistics_(

audio/channel_receive.h

@@ -115,7 +115,7 @@ class ChannelReceive : public RtpData {
                  bool jitter_buffer_fast_playout,
                  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
                  absl::optional<AudioCodecPairId> codec_pair_id,
-                 FrameDecryptorInterface* frame_decryptor,
+                 rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
                  const webrtc::CryptoOptions& crypto_options);
   virtual ~ChannelReceive();
 
@@ -260,7 +260,7 @@ class ChannelReceive : public RtpData {
   rtc::ThreadChecker construction_thread_;
 
   // E2EE Audio Frame Decryption
-  FrameDecryptorInterface* frame_decryptor_ = nullptr;
+  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
   webrtc::CryptoOptions crypto_options_;
 };
 

audio/channel_send.cc

@@ -993,14 +993,15 @@ int64_t ChannelSend::GetRTT() const {
   return rtt;
 }
 
-void ChannelSend::SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor) {
+void ChannelSend::SetFrameEncryptor(
+    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
   rtc::CritScope cs(&encoder_queue_lock_);
   if (encoder_queue_is_active_) {
     encoder_queue_->PostTask([this, frame_encryptor]() {
-      this->frame_encryptor_ = frame_encryptor;
+      this->frame_encryptor_ = std::move(frame_encryptor);
     });
   } else {
-    frame_encryptor_ = frame_encryptor;
+    frame_encryptor_ = std::move(frame_encryptor);
   }
 }
 

audio/channel_send.h

@@ -228,7 +228,8 @@ class ChannelSend
   int64_t GetRTT() const;
 
   // E2EE Custom Audio Frame Encryption
-  void SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor);
+  void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
 
  private:
   class ProcessAndEncodeAudioTask;
@@ -300,7 +301,7 @@ class ChannelSend
   rtc::TaskQueue* encoder_queue_ = nullptr;
 
   // E2EE Audio Frame Encryption
-  FrameEncryptorInterface* frame_encryptor_ = nullptr;
+  rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
   // E2EE Frame Encryption Options
   webrtc::CryptoOptions crypto_options_;
   int configured_bitrate_bps_ = 0;

audio/channel_send_proxy.cc

@@ -12,6 +12,7 @@
 
 #include <utility>
 
+#include "api/crypto/frameencryptorinterface.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "rtc_base/checks.h"
 
@@ -199,7 +200,7 @@ ChannelSend* ChannelSendProxy::GetChannel() const {
 }
 
 void ChannelSendProxy::SetFrameEncryptor(
-    FrameEncryptorInterface* frame_encryptor) {
+    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel_->SetFrameEncryptor(frame_encryptor);
 }

audio/channel_send_proxy.h

@@ -87,7 +87,8 @@ class ChannelSendProxy {
   virtual ChannelSend* GetChannel() const;
 
   // E2EE Custom Audio Frame Encryption (Optional)
-  virtual void SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor);
+  virtual void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
 
  private:
   // Thread checkers document and lock usage of some methods on voe::Channel to

audio/mock_voe_channel_proxy.h

@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "api/test/mock_frame_encryptor.h"
 #include "audio/channel_receive_proxy.h"
 #include "audio/channel_send_proxy.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -105,8 +106,9 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
                void(float recoverable_packet_loss_rate));
   MOCK_METHOD0(StartSend, void());
   MOCK_METHOD0(StopSend, void());
-  MOCK_METHOD1(SetFrameEncryptor,
-               void(FrameEncryptorInterface* frame_encryptor));
+  MOCK_METHOD1(
+      SetFrameEncryptor,
+      void(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor));
 };
 }  // namespace test
 }  // namespace webrtc

pc/BUILD.gn

@@ -497,7 +497,8 @@ if (rtc_include_tests) {
     deps = [
       ":peerconnection",
       ":rtc_pc_base",
-      "../api:fake_frame_crypto",
+      "../api:fake_frame_decryptor",
+      "../api:fake_frame_encryptor",
       "../api:libjingle_peerconnection_api",
       "../api:mock_rtp",
       "../api/units:time_delta",