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1 /*
2  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
12 #define MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
13 
14 #include <memory>
15 #include <vector>
16 
17 #include "api/audio/audio_frame.h"
18 #include "modules/audio_processing/agc2/limiter.h"
19 
20 namespace webrtc {
21 class ApmDataDumper;
22 
23 class FrameCombiner {
24  public:
25   enum class LimiterType { kNoLimiter, kApmAgcLimiter, kApmAgc2Limiter };
26   explicit FrameCombiner(bool use_limiter);
27   ~FrameCombiner();
28 
29   // Combine several frames into one. Assumes sample_rate,
30   // samples_per_channel of the input frames match the parameters. The
31   // parameters 'number_of_channels' and 'sample_rate' are needed
32   // because 'mix_list' can be empty. The parameter
33   // 'number_of_streams' is used for determining whether to pass the
34   // data through a limiter.
35   void Combine(const std::vector<AudioFrame*>& mix_list,
36                size_t number_of_channels,
37                int sample_rate,
38                size_t number_of_streams,
39                AudioFrame* audio_frame_for_mixing);
40 
41   // Stereo, 48 kHz, 10 ms.
42   static constexpr size_t kMaximumNumberOfChannels = 8;
43   static constexpr size_t kMaximumChannelSize = 48 * 10;
44 
45   using MixingBuffer = std::array<std::array<float, kMaximumChannelSize>,
46                                   kMaximumNumberOfChannels>;
47 
48  private:
49   void LogMixingStats(const std::vector<AudioFrame*>& mix_list,
50                       int sample_rate,
51                       size_t number_of_streams) const;
52 
53   std::unique_ptr<ApmDataDumper> data_dumper_;
54   std::unique_ptr<MixingBuffer> mixing_buffer_;
55   Limiter limiter_;
56   const bool use_limiter_;
57   mutable int uma_logging_counter_ = 0;
58 };
59 }  // namespace webrtc
60 
61 #endif  // MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
62