Repo created

TMessagesProj/jni/voip/webrtc/modules/audio_mixer/frame_combiner.h

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+/*
+ *  Copyright (c) 2017 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_MIXER_FRAME_COMBINER_H_
+#define MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/audio_frame.h"
+#include "modules/audio_processing/agc2/limiter.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+class FrameCombiner {
+ public:
+  explicit FrameCombiner(bool use_limiter);
+  ~FrameCombiner();
+
+  // Combine several frames into one. Assumes sample_rate,
+  // samples_per_channel of the input frames match the parameters. The
+  // parameters 'number_of_channels' and 'sample_rate' are needed
+  // because 'mix_list' can be empty. The parameter
+  // 'number_of_streams' is used for determining whether to pass the
+  // data through a limiter.
+  void Combine(rtc::ArrayView<AudioFrame* const> mix_list,
+               size_t number_of_channels,
+               int sample_rate,
+               size_t number_of_streams,
+               AudioFrame* audio_frame_for_mixing);
+
+  // Stereo, 48 kHz, 10 ms.
+  static constexpr size_t kMaximumNumberOfChannels = 8;
+  static constexpr size_t kMaximumChannelSize = 48 * 10;
+
+  using MixingBuffer = std::array<std::array<float, kMaximumChannelSize>,
+                                  kMaximumNumberOfChannels>;
+
+ private:
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  std::unique_ptr<MixingBuffer> mixing_buffer_;
+  Limiter limiter_;
+  const bool use_limiter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_FRAME_COMBINER_H_