tor-browser

The Tor Browser
git clone https://git.dasho.dev/tor-browser.git
Log | Files | Refs | README | LICENSE

encode_neteq_input.h (2531B)

      1 /*
      2 *  Copyright (c) 2016 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_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
     12 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
     13 
     14 #include <cstddef>
     15 #include <cstdint>
     16 #include <memory>
     17 #include <optional>
     18 
     19 #include "api/array_view.h"
     20 #include "api/audio_codecs/audio_encoder.h"
     21 #include "modules/audio_coding/neteq/tools/neteq_input.h"
     22 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
     23 
     24 namespace webrtc {
     25 namespace test {
     26 
     27 // This class provides a NetEqInput that takes audio from a generator object and
     28 // encodes it using a given audio encoder.
     29 class EncodeNetEqInput : public NetEqInput {
     30 public:
     31  // Generator class, to be provided to the EncodeNetEqInput constructor.
     32  class Generator {
     33   public:
     34    virtual ~Generator() = default;
     35    // Returns the next num_samples values from the signal generator.
     36    virtual ArrayView<const int16_t> Generate(size_t num_samples) = 0;
     37  };
     38 
     39  // The source will end after the given input duration.
     40  EncodeNetEqInput(std::unique_ptr<Generator> generator,
     41                   std::unique_ptr<AudioEncoder> encoder,
     42                   int64_t input_duration_ms);
     43  ~EncodeNetEqInput() override;
     44 
     45  std::optional<int64_t> NextPacketTime() const override;
     46 
     47  std::optional<int64_t> NextOutputEventTime() const override;
     48 
     49  std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
     50    return std::nullopt;
     51  }
     52 
     53  std::unique_ptr<RtpPacketReceived> PopPacket() override;
     54 
     55  void AdvanceOutputEvent() override;
     56 
     57  void AdvanceSetMinimumDelay() override {}
     58 
     59  bool ended() const override;
     60 
     61  const RtpPacketReceived* NextPacket() const override;
     62 
     63 private:
     64  static constexpr int64_t kOutputPeriodMs = 10;
     65 
     66  void CreatePacket();
     67 
     68  std::unique_ptr<Generator> generator_;
     69  std::unique_ptr<AudioEncoder> encoder_;
     70  std::unique_ptr<RtpPacketReceived> packet_data_;
     71  uint32_t rtp_timestamp_ = 0;
     72  int16_t sequence_number_ = 0;
     73  int64_t next_packet_time_ms_ = 0;
     74  int64_t next_output_event_ms_ = 0;
     75  const int64_t input_duration_ms_;
     76 };
     77 
     78 }  // namespace test
     79 }  // namespace webrtc
     80 #endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_