diff --git a/DEPS b/DEPS
index 99b48af..e95ff42 100644
--- a/DEPS
+++ b/DEPS
@@ -34,6 +34,7 @@
   # WebRTC production code.
   '-base',
   '-chromium',
+  '+external/webrtc/webrtc',  # Android platform build.
   '+gflags',
   '+libyuv',
   '+net',
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 7b7acd3..50438f9 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -7,6 +7,7 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/arm.gni")
+import("//third_party/protobuf/proto_library.gni")
 import("../../build/webrtc.gni")
 
 config("audio_coding_config") {
@@ -79,6 +80,35 @@
   }
 }
 
+proto_library("acm_dump_proto") {
+  sources = [
+    "main/acm2/dump.proto",
+  ]
+  proto_out_dir = "webrtc/audio_coding"
+}
+
+source_set("acm_dump") {
+  sources = [
+    "main/acm2/acm_dump.cc",
+    "main/acm2/acm_dump.h",
+  ]
+
+  defines = []
+
+  configs += [ "../..:common_config" ]
+
+  public_configs = [ "../..:common_inherited_config" ]
+
+  deps = [
+    ":acm_dump_proto",
+    "../..:webrtc_common",
+  ]
+
+  if (rtc_enable_protobuf) {
+    defines += [ "RTC_AUDIOCODING_DEBUG_DUMP" ]
+  }
+}
+
 source_set("audio_decoder_interface") {
   sources = [
     "codecs/audio_decoder.cc",
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
new file mode 100644
index 0000000..4454c25
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc
@@ -0,0 +1,220 @@
+/*
+ *  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/modules/audio_coding/main/acm2/acm_dump.h"
+
+#include <sstream>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/file_wrapper.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
+#else
+#include "webrtc/audio_coding/dump.pb.h"
+#endif
+
+namespace webrtc {
+
+// Noop implementation if flag is not set
+#ifndef RTC_AUDIOCODING_DEBUG_DUMP
+class AcmDumpImpl final : public AcmDump {
+ public:
+  void StartLogging(const std::string& file_name, int duration_ms) override{};
+  void LogRtpPacket(bool incoming,
+                    const uint8_t* packet,
+                    size_t length) override{};
+  void LogDebugEvent(DebugEvent event_type,
+                     const std::string& event_message) override{};
+  void LogDebugEvent(DebugEvent event_type) override{};
+};
+#else
+
+class AcmDumpImpl final : public AcmDump {
+ public:
+  AcmDumpImpl();
+
+  void StartLogging(const std::string& file_name, int duration_ms) override;
+  void LogRtpPacket(bool incoming,
+                    const uint8_t* packet,
+                    size_t length) override;
+  void LogDebugEvent(DebugEvent event_type,
+                     const std::string& event_message) override;
+  void LogDebugEvent(DebugEvent event_type) override;
+
+ private:
+  // Checks if the logging time has expired, and if so stops the logging.
+  void StopIfNecessary() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Stops logging and clears the stored data and buffers.
+  void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Returns true if the logging is currently active.
+  bool CurrentlyLogging() const EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+    return active_ &&
+           (clock_->TimeInMicroseconds() <= start_time_us_ + duration_us_);
+  }
+  // This function is identical to LogDebugEvent, but requires holding the lock.
+  void LogDebugEventLocked(DebugEvent event_type,
+                           const std::string& event_message)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
+  rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
+  rtc::scoped_ptr<ACMDumpEventStream> stream_ GUARDED_BY(crit_);
+  bool active_ GUARDED_BY(crit_);
+  int64_t start_time_us_ GUARDED_BY(crit_);
+  int64_t duration_us_ GUARDED_BY(crit_);
+  const webrtc::Clock* clock_ GUARDED_BY(crit_);
+};
+
+namespace {
+
+// Convert from AcmDump's debug event enum (runtime format) to the corresponding
+// protobuf enum (serialized format).
+ACMDumpDebugEvent_EventType convertDebugEvent(AcmDump::DebugEvent event_type) {
+  switch (event_type) {
+    case AcmDump::DebugEvent::kLogStart:
+      return ACMDumpDebugEvent::LOG_START;
+    case AcmDump::DebugEvent::kLogEnd:
+      return ACMDumpDebugEvent::LOG_END;
+    case AcmDump::DebugEvent::kAudioPlayout:
+      return ACMDumpDebugEvent::AUDIO_PLAYOUT;
+  }
+  return ACMDumpDebugEvent::UNKNOWN_EVENT;
+}
+
+}  // Anonymous namespace.
+
+// AcmDumpImpl member functions.
+AcmDumpImpl::AcmDumpImpl()
+    : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+      file_(webrtc::FileWrapper::Create()),
+      stream_(new webrtc::ACMDumpEventStream()),
+      active_(false),
+      start_time_us_(0),
+      duration_us_(0),
+      clock_(webrtc::Clock::GetRealTimeClock()) {
+}
+
+void AcmDumpImpl::StartLogging(const std::string& file_name, int duration_ms) {
+  CriticalSectionScoped lock(crit_.get());
+  Clear();
+  if (file_->OpenFile(file_name.c_str(), false) != 0) {
+    return;
+  }
+  // Add a single object to the stream that is reused at every log event.
+  stream_->add_stream();
+  active_ = true;
+  start_time_us_ = clock_->TimeInMicroseconds();
+  duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
+  // Log the start event.
+  std::stringstream log_msg;
+  log_msg << "Initial timestamp: " << start_time_us_;
+  LogDebugEventLocked(DebugEvent::kLogStart, log_msg.str());
+}
+
+void AcmDumpImpl::LogRtpPacket(bool incoming,
+                               const uint8_t* packet,
+                               size_t length) {
+  CriticalSectionScoped lock(crit_.get());
+  if (!CurrentlyLogging()) {
+    StopIfNecessary();
+    return;
+  }
+  // Reuse the same object at every log event.
+  auto rtp_event = stream_->mutable_stream(0);
+  rtp_event->clear_debug_event();
+  const int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
+  rtp_event->set_timestamp_us(timestamp);
+  rtp_event->set_type(webrtc::ACMDumpEvent::RTP_EVENT);
+  rtp_event->mutable_packet()->set_direction(
+      incoming ? ACMDumpRTPPacket::INCOMING : ACMDumpRTPPacket::OUTGOING);
+  rtp_event->mutable_packet()->set_rtp_data(packet, length);
+  std::string dump_buffer;
+  stream_->SerializeToString(&dump_buffer);
+  file_->Write(dump_buffer.data(), dump_buffer.size());
+  file_->Flush();
+}
+
+void AcmDumpImpl::LogDebugEvent(DebugEvent event_type,
+                                const std::string& event_message) {
+  CriticalSectionScoped lock(crit_.get());
+  LogDebugEventLocked(event_type, event_message);
+}
+
+void AcmDumpImpl::LogDebugEvent(DebugEvent event_type) {
+  CriticalSectionScoped lock(crit_.get());
+  LogDebugEventLocked(event_type, "");
+}
+
+void AcmDumpImpl::StopIfNecessary() {
+  if (active_) {
+    DCHECK_GT(clock_->TimeInMicroseconds(), start_time_us_ + duration_us_);
+    LogDebugEventLocked(DebugEvent::kLogEnd, "");
+    Clear();
+  }
+}
+
+void AcmDumpImpl::Clear() {
+  if (active_ || file_->Open()) {
+    file_->CloseFile();
+  }
+  active_ = false;
+  stream_->Clear();
+}
+
+void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
+                                      const std::string& event_message) {
+  if (!CurrentlyLogging()) {
+    StopIfNecessary();
+    return;
+  }
+
+  // Reuse the same object at every log event.
+  auto event = stream_->mutable_stream(0);
+  int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
+  event->set_timestamp_us(timestamp);
+  event->set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
+  event->clear_packet();
+  auto debug_event = event->mutable_debug_event();
+  debug_event->set_type(convertDebugEvent(event_type));
+  debug_event->set_message(event_message);
+  std::string dump_buffer;
+  stream_->SerializeToString(&dump_buffer);
+  file_->Write(dump_buffer.data(), dump_buffer.size());
+}
+
+#endif  // RTC_AUDIOCODING_DEBUG_DUMP
+
+// AcmDump member functions.
+rtc::scoped_ptr<AcmDump> AcmDump::Create() {
+  return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl());
+}
+
+bool AcmDump::ParseAcmDump(const std::string& file_name,
+                           ACMDumpEventStream* result) {
+  char tmp_buffer[1024];
+  int bytes_read = 0;
+  rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create());
+  if (dump_file->OpenFile(file_name.c_str(), true) != 0) {
+    return false;
+  }
+  std::string dump_buffer;
+  while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
+    dump_buffer.append(tmp_buffer, bytes_read);
+  }
+  dump_file->CloseFile();
+  return result->ParseFromString(dump_buffer);
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.h b/webrtc/modules/audio_coding/main/acm2/acm_dump.h
new file mode 100644
index 0000000..c72c387
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump.h
@@ -0,0 +1,59 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
+
+#include <string>
+
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+// Forward declaration of storage class that is automatically generated from
+// the protobuf file.
+class ACMDumpEventStream;
+
+class AcmDumpImpl;
+
+class AcmDump {
+ public:
+  // The types of debug events that are currently supported for logging.
+  enum class DebugEvent { kLogStart, kLogEnd, kAudioPlayout };
+
+  virtual ~AcmDump() {}
+
+  static rtc::scoped_ptr<AcmDump> Create();
+
+  // Starts logging for the specified duration to the specified file.
+  // The logging will stop automatically after the specified duration.
+  // If the file already exists it will be overwritten.
+  // The function will return false on failure.
+  virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
+
+  // Logs an incoming or outgoing RTP packet.
+  virtual void LogRtpPacket(bool incoming,
+                            const uint8_t* packet,
+                            size_t length) = 0;
+
+  // Logs a debug event, with optional message.
+  virtual void LogDebugEvent(DebugEvent event_type,
+                             const std::string& event_message) = 0;
+  virtual void LogDebugEvent(DebugEvent event_type) = 0;
+
+  // Reads an AcmDump file and returns true when reading was successful.
+  // The result is stored in the given ACMDumpEventStream object.
+  static bool ParseAcmDump(const std::string& file_name,
+                           ACMDumpEventStream* result);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
new file mode 100644
index 0000000..55c948e
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc
@@ -0,0 +1,117 @@
+/*
+ *  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.
+ */
+
+#ifdef RTC_AUDIOCODING_DEBUG_DUMP
+
+#include <stdio.h>
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_dump.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
+#else
+#include "webrtc/audio_coding/dump.pb.h"
+#endif
+
+namespace webrtc {
+
+// Test for the acm dump class. Dumps some RTP packets to disk, then reads them
+// back to see if they match.
+class AcmDumpTest : public ::testing::Test {
+ public:
+  AcmDumpTest() : log_dumper_(AcmDump::Create()) {}
+  void VerifyResults(const ACMDumpEventStream& parsed_stream,
+                     size_t packet_size) {
+    // Verify the result.
+    EXPECT_EQ(3, parsed_stream.stream_size());
+    const ACMDumpEvent& start_event = parsed_stream.stream(0);
+    ASSERT_TRUE(start_event.has_type());
+    EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type());
+    EXPECT_TRUE(start_event.has_timestamp_us());
+    EXPECT_FALSE(start_event.has_packet());
+    ASSERT_TRUE(start_event.has_debug_event());
+    auto start_debug_event = start_event.debug_event();
+    ASSERT_TRUE(start_debug_event.has_type());
+    EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type());
+    ASSERT_TRUE(start_debug_event.has_message());
+
+    for (int i = 1; i < parsed_stream.stream_size(); i++) {
+      const ACMDumpEvent& test_event = parsed_stream.stream(i);
+      ASSERT_TRUE(test_event.has_type());
+      EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type());
+      EXPECT_TRUE(test_event.has_timestamp_us());
+      EXPECT_FALSE(test_event.has_debug_event());
+      ASSERT_TRUE(test_event.has_packet());
+      const ACMDumpRTPPacket& test_packet = test_event.packet();
+      ASSERT_TRUE(test_packet.has_direction());
+      if (i == 1) {
+        EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction());
+      } else if (i == 2) {
+        EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction());
+      }
+      ASSERT_TRUE(test_packet.has_rtp_data());
+      ASSERT_EQ(packet_size, test_packet.rtp_data().size());
+      for (size_t i = 0; i < packet_size; i++) {
+        EXPECT_EQ(rtp_packet_[i],
+                  static_cast<uint8_t>(test_packet.rtp_data()[i]));
+      }
+    }
+  }
+
+  void Run(int packet_size, int random_seed) {
+    rtp_packet_.clear();
+    rtp_packet_.reserve(packet_size);
+    srand(random_seed);
+    // Fill the packet vector with random data.
+    for (int i = 0; i < packet_size; i++) {
+      rtp_packet_.push_back(rand());
+    }
+    // Find the name of the current test, in order to use it as a temporary
+    // filename.
+    auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+    const std::string temp_filename =
+        test::OutputPath() + test_info->test_case_name() + test_info->name();
+
+    log_dumper_->StartLogging(temp_filename, 10000000);
+    log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
+    log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
+
+    // Read the generated file from disk.
+    ACMDumpEventStream parsed_stream;
+
+    ASSERT_EQ(true, AcmDump::ParseAcmDump(temp_filename, &parsed_stream));
+
+    VerifyResults(parsed_stream, packet_size);
+
+    // Clean up temporary file - can be pretty slow.
+    remove(temp_filename.c_str());
+  }
+
+  std::vector<uint8_t> rtp_packet_;
+  rtc::scoped_ptr<AcmDump> log_dumper_;
+};
+
+TEST_F(AcmDumpTest, DumpAndRead) {
+  Run(256, 321);
+  Run(256, 123);
+}
+
+}  // namespace webrtc
+
+#endif  // RTC_AUDIOCODING_DEBUG_DUMP
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
index 9a38fac..c78bcd7 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -78,6 +78,34 @@
         'nack.h',
       ],
     },
+    {
+      'target_name': 'acm_dump_proto',
+      'type': 'static_library',
+      'sources': ['dump.proto',],
+      'variables': {
+        'proto_in_dir': '.',
+        # Workaround to protect against gyp's pathname relativization when
+        # this file is included by modules.gyp.
+        'proto_out_protected': 'webrtc/audio_coding',
+        'proto_out_dir': '<(proto_out_protected)',
+      },
+      'includes': ['../../../../build/protoc.gypi',],
+    },
+    {
+      'target_name': 'acm_dump',
+      'type': 'static_library',
+      'conditions': [
+        ['enable_protobuf==1', {
+          'defines': ['RTC_AUDIOCODING_DEBUG_DUMP'],
+          }
+        ],
+      ],
+      'sources': [
+        'acm_dump.h',
+        'acm_dump.cc'
+      ],
+      'dependencies': ['acm_dump_proto'],
+    },
   ],
   'conditions': [
     ['include_tests==1', {
diff --git a/webrtc/modules/audio_coding/main/acm2/dump.proto b/webrtc/modules/audio_coding/main/acm2/dump.proto
new file mode 100644
index 0000000..416bb7a
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/dump.proto
@@ -0,0 +1,78 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc;
+
+// This is the main message to dump to a file, it can contain multiple event
+// messages, but it is possible to append multiple EventStreams (each with a
+// single event) to a file.
+// This has the benefit that there's no need to keep all data in memory.
+message ACMDumpEventStream {
+  repeated ACMDumpEvent stream = 1;
+}
+
+message ACMDumpEvent {
+  // required - Elapsed wallclock time in us since the start of the log.
+  optional int64 timestamp_us = 1;
+
+  // The different types of events that can occur, the UNKNOWN_EVENT entry
+  // is added in case future EventTypes are added, in that case old code will
+  // receive the new events as UNKNOWN_EVENT.
+  enum EventType {
+    UNKNOWN_EVENT = 0;
+    RTP_EVENT = 1;
+    DEBUG_EVENT = 2;
+  }
+
+  // required - Indicates the type of this event
+  optional EventType type = 2;
+
+  // optional - but required if type == RTP_EVENT
+  optional ACMDumpRTPPacket packet = 3;
+
+  // optional - but required if type == DEBUG_EVENT
+  optional ACMDumpDebugEvent debug_event = 4;
+}
+
+message ACMDumpRTPPacket {
+  // Indicates if the packet is incoming or outgoing with respect to the user
+  // that is logging the data.
+  enum Direction {
+    UNKNOWN_DIRECTION = 0;
+    OUTGOING = 1;
+    INCOMING = 2;
+  }
+  enum PayloadType {
+    UNKNOWN_TYPE = 0;
+    AUDIO = 1;
+    VIDEO = 2;
+    RTX = 3;
+  }
+
+  // required
+  optional Direction direction = 1;
+
+  // required
+  optional PayloadType type = 2;
+
+  // required - Contains the whole RTP packet (header+payload).
+  optional bytes RTP_data = 3;
+}
+
+message ACMDumpDebugEvent {
+  // Indicates the type of the debug event.
+  // LOG_START and LOG_END indicate the start and end of the log respectively.
+  // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM.
+  enum EventType {
+    UNKNOWN_EVENT = 0;
+    LOG_START = 1;
+    LOG_END = 2;
+    AUDIO_PLAYOUT = 3;
+  }
+
+  // required
+  optional EventType type = 1;
+
+  // An optional message that can be used to store additional information about
+  // the debug event.
+  optional string message = 2;
+}
\ No newline at end of file
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index e29f683..150ee8e 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -310,12 +310,17 @@
               'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
             }],
             ['enable_protobuf==1', {
-              'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
+              'defines': [
+                'WEBRTC_AUDIOPROC_DEBUG_DUMP',
+                'RTC_AUDIOCODING_DEBUG_DUMP',
+              ],
               'dependencies': [
+                'acm_dump',
                 'audioproc_protobuf_utils',
                 'audioproc_unittest_proto',
               ],
               'sources': [
+                'audio_coding/main/acm2/acm_dump_unittest.cc',
                 'audio_processing/audio_processing_impl_unittest.cc',
                 'audio_processing/test/audio_processing_unittest.cc',
                 'audio_processing/test/test_utils.h',