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, 151 insertions, 0 deletions

diff --git a/webrtc/test/fake_audio_device.cc b/webrtc/test/fake_audio_device.cc
new file mode 100644
index 0000000000..e307dd7664
--- /dev/null
+++ b/webrtc/test/fake_audio_device.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2013 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/test/fake_audio_device.h"
+
+#include <algorithm>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/thread_wrapper.h"
+
+namespace webrtc {
+namespace test {
+
+FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
+    : audio_callback_(NULL),
+      capturing_(false),
+      captured_audio_(),
+      playout_buffer_(),
+      last_playout_ms_(-1),
+      clock_(clock),
+      tick_(EventTimerWrapper::Create()),
+      file_utility_(new ModuleFileUtility(0)),
+      input_stream_(FileWrapper::Create()) {
+  memset(captured_audio_, 0, sizeof(captured_audio_));
+  memset(playout_buffer_, 0, sizeof(playout_buffer_));
+  // Open audio input file as read-only and looping.
+  EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
+      << filename;
+}
+
+FakeAudioDevice::~FakeAudioDevice() {
+  Stop();
+
+  if (thread_.get() != NULL)
+    thread_->Stop();
+}
+
+int32_t FakeAudioDevice::Init() {
+  rtc::CritScope cs(&lock_);
+  if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
+    return -1;
+
+  if (!tick_->StartTimer(true, 10))
+    return -1;
+  thread_ = ThreadWrapper::CreateThread(FakeAudioDevice::Run, this,
+                                        "FakeAudioDevice");
+  if (thread_.get() == NULL)
+    return -1;
+  if (!thread_->Start()) {
+    thread_.reset();
+    return -1;
+  }
+  thread_->SetPriority(webrtc::kHighPriority);
+  return 0;
+}
+
+int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
+  rtc::CritScope cs(&lock_);
+  audio_callback_ = callback;
+  return 0;
+}
+
+bool FakeAudioDevice::Playing() const {
+  rtc::CritScope cs(&lock_);
+  return capturing_;
+}
+
+int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
+  *delay_ms = 0;
+  return 0;
+}
+
+bool FakeAudioDevice::Recording() const {
+  rtc::CritScope cs(&lock_);
+  return capturing_;
+}
+
+bool FakeAudioDevice::Run(void* obj) {
+  static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
+  return true;
+}
+
+void FakeAudioDevice::CaptureAudio() {
+  {
+    rtc::CritScope cs(&lock_);
+    if (capturing_) {
+      int bytes_read = file_utility_->ReadPCMData(
+          *input_stream_.get(), captured_audio_, kBufferSizeBytes);
+      if (bytes_read <= 0)
+        return;
+      // 2 bytes per sample.
+      size_t num_samples = static_cast<size_t>(bytes_read / 2);
+      uint32_t new_mic_level;
+      EXPECT_EQ(0,
+                audio_callback_->RecordedDataIsAvailable(captured_audio_,
+                                                         num_samples,
+                                                         2,
+                                                         1,
+                                                         kFrequencyHz,
+                                                         0,
+                                                         0,
+                                                         0,
+                                                         false,
+                                                         new_mic_level));
+      size_t samples_needed = kFrequencyHz / 100;
+      int64_t now_ms = clock_->TimeInMilliseconds();
+      uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
+      if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
+        samples_needed = std::min(
+            static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
+            kBufferSizeBytes / 2);
+      }
+      size_t samples_out = 0;
+      int64_t elapsed_time_ms = -1;
+      int64_t ntp_time_ms = -1;
+      EXPECT_EQ(0,
+                audio_callback_->NeedMorePlayData(samples_needed,
+                                                  2,
+                                                  1,
+                                                  kFrequencyHz,
+                                                  playout_buffer_,
+                                                  samples_out,
+                                                  &elapsed_time_ms,
+                                                  &ntp_time_ms));
+    }
+  }
+  tick_->Wait(WEBRTC_EVENT_INFINITE);
+}
+
+void FakeAudioDevice::Start() {
+  rtc::CritScope cs(&lock_);
+  capturing_ = true;
+}
+
+void FakeAudioDevice::Stop() {
+  rtc::CritScope cs(&lock_);
+  capturing_ = false;
+}
+}  // namespace test
+}  // namespace webrtc