Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 91 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h
new file mode 100644
index 0000000000..8aea8f2e95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 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 MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc/gain_control.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class GainControlImpl : public GainControl {
+ public:
+  GainControlImpl();
+  GainControlImpl(const GainControlImpl&) = delete;
+  GainControlImpl& operator=(const GainControlImpl&) = delete;
+
+  ~GainControlImpl() override;
+
+  void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
+  int AnalyzeCaptureAudio(const AudioBuffer& audio);
+  int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
+
+  void Initialize(size_t num_proc_channels, int sample_rate_hz);
+
+  static void PackRenderAudioBuffer(const AudioBuffer& audio,
+                                    std::vector<int16_t>* packed_buffer);
+
+  // GainControl implementation.
+  int stream_analog_level() const override;
+  bool is_limiter_enabled() const override { return limiter_enabled_; }
+  Mode mode() const override { return mode_; }
+  int set_mode(Mode mode) override;
+  int compression_gain_db() const override { return compression_gain_db_; }
+  int set_analog_level_limits(int minimum, int maximum) override;
+  int set_compression_gain_db(int gain) override;
+  int set_target_level_dbfs(int level) override;
+  int enable_limiter(bool enable) override;
+  int set_stream_analog_level(int level) override;
+
+ private:
+  struct MonoAgcState;
+
+  // GainControl implementation.
+  int target_level_dbfs() const override { return target_level_dbfs_; }
+  int analog_level_minimum() const override { return minimum_capture_level_; }
+  int analog_level_maximum() const override { return maximum_capture_level_; }
+  bool stream_is_saturated() const override { return stream_is_saturated_; }
+
+  int Configure();
+
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+
+  Mode mode_;
+  int minimum_capture_level_;
+  int maximum_capture_level_;
+  bool limiter_enabled_;
+  int target_level_dbfs_;
+  int compression_gain_db_;
+  int analog_capture_level_ = 0;
+  bool was_analog_level_set_;
+  bool stream_is_saturated_;
+
+  std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
+  std::vector<int> capture_levels_;
+
+  absl::optional<size_t> num_proc_channels_;
+  absl::optional<int> sample_rate_hz_;
+
+  static int instance_counter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_