1 /* 2 * Copyright (c) 2012 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_TEST_TESTSTEREO_H_ 12 #define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ 13 14 #include <math.h> 15 16 #include <memory> 17 18 #include "modules/audio_coding/include/audio_coding_module.h" 19 #include "modules/audio_coding/test/PCMFile.h" 20 21 #define PCMA_AND_PCMU 22 23 namespace webrtc { 24 25 enum StereoMonoMode { kNotSet, kMono, kStereo }; 26 27 class TestPackStereo : public AudioPacketizationCallback { 28 public: 29 TestPackStereo(); 30 ~TestPackStereo(); 31 32 void RegisterReceiverACM(AudioCodingModule* acm); 33 34 int32_t SendData(const AudioFrameType frame_type, 35 const uint8_t payload_type, 36 const uint32_t timestamp, 37 const uint8_t* payload_data, 38 const size_t payload_size, 39 int64_t absolute_capture_timestamp_ms) override; 40 41 uint16_t payload_size(); 42 uint32_t timestamp_diff(); 43 void reset_payload_size(); 44 void set_codec_mode(StereoMonoMode mode); 45 void set_lost_packet(bool lost); 46 47 private: 48 AudioCodingModule* receiver_acm_; 49 int16_t seq_no_; 50 uint32_t timestamp_diff_; 51 uint32_t last_in_timestamp_; 52 uint64_t total_bytes_; 53 int payload_size_; 54 StereoMonoMode codec_mode_; 55 // Simulate packet losses 56 bool lost_packet_; 57 }; 58 59 class TestStereo { 60 public: 61 TestStereo(); 62 ~TestStereo(); 63 64 void Perform(); 65 66 private: 67 // The default value of '-1' indicates that the registration is based only on 68 // codec name and a sampling frequncy matching is not required. This is useful 69 // for codecs which support several sampling frequency. 70 void RegisterSendCodec(char side, 71 char* codec_name, 72 int32_t samp_freq_hz, 73 int rate, 74 int pack_size, 75 int channels); 76 77 void Run(TestPackStereo* channel, 78 int in_channels, 79 int out_channels, 80 int percent_loss = 0); 81 void OpenOutFile(int16_t test_number); 82 83 std::unique_ptr<AudioCodingModule> acm_a_; 84 std::unique_ptr<AudioCodingModule> acm_b_; 85 86 TestPackStereo* channel_a2b_; 87 88 PCMFile* in_file_stereo_; 89 PCMFile* in_file_mono_; 90 PCMFile out_file_; 91 int16_t test_cntr_; 92 uint16_t pack_size_samp_; 93 uint16_t pack_size_bytes_; 94 int counter_; 95 char* send_codec_name_; 96 }; 97 98 } // namespace webrtc 99 100 #endif // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ 101