| /* | 
 |  *  Copyright (c) 2011 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_CODING_TEST_TESTVADDTX_H_ | 
 | #define MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "absl/strings/string_view.h" | 
 | #include "api/audio_codecs/audio_decoder_factory.h" | 
 | #include "api/audio_codecs/audio_encoder_factory.h" | 
 | #include "common_audio/vad/include/vad.h" | 
 | #include "modules/audio_coding/acm2/acm_receiver.h" | 
 | #include "modules/audio_coding/include/audio_coding_module.h" | 
 | #include "modules/audio_coding/include/audio_coding_module_typedefs.h" | 
 | #include "modules/audio_coding/test/Channel.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | // This class records the frame type, and delegates actual sending to the | 
 | // `next_` AudioPacketizationCallback. | 
 | class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback { | 
 |  public: | 
 |   explicit MonitoringAudioPacketizationCallback( | 
 |       AudioPacketizationCallback* next); | 
 |  | 
 |   int32_t SendData(AudioFrameType frame_type, | 
 |                    uint8_t payload_type, | 
 |                    uint32_t timestamp, | 
 |                    const uint8_t* payload_data, | 
 |                    size_t payload_len_bytes, | 
 |                    int64_t absolute_capture_timestamp_ms) override; | 
 |  | 
 |   void PrintStatistics(); | 
 |   void ResetStatistics(); | 
 |   void GetStatistics(uint32_t* stats); | 
 |  | 
 |  private: | 
 |   // 0 - kEmptyFrame | 
 |   // 1 - kAudioFrameSpeech | 
 |   // 2 - kAudioFrameCN | 
 |   uint32_t counter_[3]; | 
 |   AudioPacketizationCallback* const next_; | 
 | }; | 
 |  | 
 | // TestVadDtx is to verify that VAD/DTX perform as they should. It runs through | 
 | // an audio file and check if the occurrence of various packet types follows | 
 | // expectation. TestVadDtx needs its derived class to implement the Perform() | 
 | // to put the test together. | 
 | class TestVadDtx { | 
 |  public: | 
 |   static const int kOutputFreqHz = 16000; | 
 |  | 
 |   TestVadDtx(); | 
 |  | 
 |  protected: | 
 |   // Returns true iff CN was added. | 
 |   bool RegisterCodec(const SdpAudioFormat& codec_format, | 
 |                      absl::optional<Vad::Aggressiveness> vad_mode); | 
 |  | 
 |   // Encoding a file and see if the numbers that various packets occur follow | 
 |   // the expectation. Saves result to a file. | 
 |   // expects[x] means | 
 |   // -1 : do not care, | 
 |   // 0  : there have been no packets of type `x`, | 
 |   // 1  : there have been packets of type `x`, | 
 |   // with `x` indicates the following packet types | 
 |   // 0 - kEmptyFrame | 
 |   // 1 - kAudioFrameSpeech | 
 |   // 2 - kAudioFrameCN | 
 |   void Run(absl::string_view in_filename, | 
 |            int frequency, | 
 |            int channels, | 
 |            absl::string_view out_filename, | 
 |            bool append, | 
 |            const int* expects); | 
 |  | 
 |   const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_; | 
 |   const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 
 |   std::unique_ptr<AudioCodingModule> acm_send_; | 
 |   std::unique_ptr<acm2::AcmReceiver> acm_receive_; | 
 |   std::unique_ptr<Channel> channel_; | 
 |   std::unique_ptr<MonitoringAudioPacketizationCallback> packetization_callback_; | 
 |   uint32_t time_stamp_ = 0x12345678; | 
 | }; | 
 |  | 
 | // TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should. | 
 | class TestWebRtcVadDtx final : public TestVadDtx { | 
 |  public: | 
 |   TestWebRtcVadDtx(); | 
 |  | 
 |   void Perform(); | 
 |  | 
 |  private: | 
 |   void RunTestCases(const SdpAudioFormat& codec_format); | 
 |   void Test(bool new_outfile, bool expect_dtx_enabled); | 
 |  | 
 |   int output_file_num_; | 
 | }; | 
 |  | 
 | // TestOpusDtx is to verify that the Opus DTX performs as it should. | 
 | class TestOpusDtx final : public TestVadDtx { | 
 |  public: | 
 |   void Perform(); | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ |