Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 196 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc
new file mode 100644
index 0000000000..8fdabeb16f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2019 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 "test/pc/e2e/echo/echo_emulation.h"
+
+#include <limits>
+#include <utility>
+
+#include "api/test/pclf/media_configuration.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr int kSingleBufferDurationMs = 10;
+
+}  // namespace
+
+EchoEmulatingCapturer::EchoEmulatingCapturer(
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    EchoEmulationConfig config)
+    : delegate_(std::move(capturer)),
+      config_(config),
+      renderer_queue_(2 * config_.echo_delay.ms() / kSingleBufferDurationMs),
+      queue_input_(TestAudioDeviceModule::SamplesPerFrame(
+                       delegate_->SamplingFrequency()) *
+                   delegate_->NumChannels()),
+      queue_output_(TestAudioDeviceModule::SamplesPerFrame(
+                        delegate_->SamplingFrequency()) *
+                    delegate_->NumChannels()) {
+  renderer_thread_.Detach();
+  capturer_thread_.Detach();
+}
+
+void EchoEmulatingCapturer::OnAudioRendered(
+    rtc::ArrayView<const int16_t> data) {
+  RTC_DCHECK_RUN_ON(&renderer_thread_);
+  if (!recording_started_) {
+    // Because rendering can start before capturing in the beginning we can have
+    // a set of empty audio data frames. So we will skip them and will start
+    // fill the queue only after 1st non-empty audio data frame will arrive.
+    bool is_empty = true;
+    for (auto d : data) {
+      if (d != 0) {
+        is_empty = false;
+        break;
+      }
+    }
+    if (is_empty) {
+      return;
+    }
+    recording_started_ = true;
+  }
+  queue_input_.assign(data.begin(), data.end());
+  if (!renderer_queue_.Insert(&queue_input_)) {
+    RTC_LOG(LS_WARNING) << "Echo queue is full";
+  }
+}
+
+bool EchoEmulatingCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
+  RTC_DCHECK_RUN_ON(&capturer_thread_);
+  bool result = delegate_->Capture(buffer);
+  // Now we have to reduce input signal to avoid saturation when mixing in the
+  // fake echo.
+  for (size_t i = 0; i < buffer->size(); ++i) {
+    (*buffer)[i] /= 2;
+  }
+
+  // When we accumulated enough delay in the echo buffer we will pop from
+  // that buffer on each ::Capture(...) call. If the buffer become empty it
+  // will mean some bug, so we will crash during removing item from the queue.
+  if (!delay_accumulated_) {
+    delay_accumulated_ =
+        renderer_queue_.SizeAtLeast() >=
+        static_cast<size_t>(config_.echo_delay.ms() / kSingleBufferDurationMs);
+  }
+
+  if (delay_accumulated_) {
+    RTC_CHECK(renderer_queue_.Remove(&queue_output_));
+    for (size_t i = 0; i < buffer->size() && i < queue_output_.size(); ++i) {
+      int32_t res = (*buffer)[i] + queue_output_[i];
+      if (res < std::numeric_limits<int16_t>::min()) {
+        res = std::numeric_limits<int16_t>::min();
+      }
+      if (res > std::numeric_limits<int16_t>::max()) {
+        res = std::numeric_limits<int16_t>::max();
+      }
+      (*buffer)[i] = static_cast<int16_t>(res);
+    }
+  }
+
+  return result;
+}
+
+EchoEmulatingRenderer::EchoEmulatingRenderer(
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+    EchoEmulatingCapturer* echo_emulating_capturer)
+    : delegate_(std::move(renderer)),
+      echo_emulating_capturer_(echo_emulating_capturer) {
+  RTC_DCHECK(echo_emulating_capturer_);
+}
+
+bool EchoEmulatingRenderer::Render(rtc::ArrayView<const int16_t> data) {
+  if (data.size() > 0) {
+    echo_emulating_capturer_->OnAudioRendered(data);
+  }
+  return delegate_->Render(data);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h
new file mode 100644
index 0000000000..359a481e46
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 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 TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
+#define TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
+
+#include <atomic>
+#include <deque>
+#include <memory>
+#include <vector>
+
+#include "api/test/pclf/media_configuration.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/swap_queue.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Reduces audio input strength from provided capturer twice and adds input
+// provided into EchoEmulatingCapturer::OnAudioRendered(...).
+class EchoEmulatingCapturer : public TestAudioDeviceModule::Capturer {
+ public:
+  EchoEmulatingCapturer(
+      std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+      EchoEmulationConfig config);
+
+  void OnAudioRendered(rtc::ArrayView<const int16_t> data);
+
+  int SamplingFrequency() const override {
+    return delegate_->SamplingFrequency();
+  }
+  int NumChannels() const override { return delegate_->NumChannels(); }
+  bool Capture(rtc::BufferT<int16_t>* buffer) override;
+
+ private:
+  std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_;
+  const EchoEmulationConfig config_;
+
+  SwapQueue<std::vector<int16_t>> renderer_queue_;
+
+  SequenceChecker renderer_thread_;
+  std::vector<int16_t> queue_input_ RTC_GUARDED_BY(renderer_thread_);
+  bool recording_started_ RTC_GUARDED_BY(renderer_thread_) = false;
+
+  SequenceChecker capturer_thread_;
+  std::vector<int16_t> queue_output_ RTC_GUARDED_BY(capturer_thread_);
+  bool delay_accumulated_ RTC_GUARDED_BY(capturer_thread_) = false;
+};
+
+// Renders output into provided renderer and also copy output into provided
+// EchoEmulationCapturer.
+class EchoEmulatingRenderer : public TestAudioDeviceModule::Renderer {
+ public:
+  EchoEmulatingRenderer(
+      std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+      EchoEmulatingCapturer* echo_emulating_capturer);
+
+  int SamplingFrequency() const override {
+    return delegate_->SamplingFrequency();
+  }
+  int NumChannels() const override { return delegate_->NumChannels(); }
+  bool Render(rtc::ArrayView<const int16_t> data) override;
+
+ private:
+  std::unique_ptr<TestAudioDeviceModule::Renderer> delegate_;
+  EchoEmulatingCapturer* echo_emulating_capturer_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ECHO_ECHO_EMULATION_H_