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 WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ 13 14 #include <math.h> 15 16 #include "webrtc/base/scoped_ptr.h" 17 #include "webrtc/modules/audio_coding/test/ACMTest.h" 18 #include "webrtc/modules/audio_coding/test/Channel.h" 19 #include "webrtc/modules/audio_coding/test/PCMFile.h" 20 21 #define PCMA_AND_PCMU 22 23 namespace webrtc { 24 25 enum StereoMonoMode { 26 kNotSet, 27 kMono, 28 kStereo 29 }; 30 31 class TestPackStereo : public AudioPacketizationCallback { 32 public: 33 TestPackStereo(); 34 ~TestPackStereo(); 35 36 void RegisterReceiverACM(AudioCodingModule* acm); 37 38 int32_t SendData(const FrameType frame_type, 39 const uint8_t payload_type, 40 const uint32_t timestamp, 41 const uint8_t* payload_data, 42 const size_t payload_size, 43 const RTPFragmentationHeader* fragmentation) override; 44 45 uint16_t payload_size(); 46 uint32_t timestamp_diff(); 47 void reset_payload_size(); 48 void set_codec_mode(StereoMonoMode mode); 49 void set_lost_packet(bool lost); 50 51 private: 52 AudioCodingModule* receiver_acm_; 53 int16_t seq_no_; 54 uint32_t timestamp_diff_; 55 uint32_t last_in_timestamp_; 56 uint64_t total_bytes_; 57 int payload_size_; 58 StereoMonoMode codec_mode_; 59 // Simulate packet losses 60 bool lost_packet_; 61 }; 62 63 class TestStereo : public ACMTest { 64 public: 65 explicit TestStereo(int test_mode); 66 ~TestStereo(); 67 68 void Perform() override; 69 70 private: 71 // The default value of '-1' indicates that the registration is based only on 72 // codec name and a sampling frequncy matching is not required. This is useful 73 // for codecs which support several sampling frequency. 74 void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz, 75 int rate, int pack_size, int channels, 76 int payload_type); 77 78 void Run(TestPackStereo* channel, int in_channels, int out_channels, 79 int percent_loss = 0); 80 void OpenOutFile(int16_t test_number); 81 void DisplaySendReceiveCodec(); 82 83 int test_mode_; 84 85 rtc::scoped_ptr<AudioCodingModule> acm_a_; 86 rtc::scoped_ptr<AudioCodingModule> acm_b_; 87 88 TestPackStereo* channel_a2b_; 89 90 PCMFile* in_file_stereo_; 91 PCMFile* in_file_mono_; 92 PCMFile out_file_; 93 int16_t test_cntr_; 94 uint16_t pack_size_samp_; 95 uint16_t pack_size_bytes_; 96 int counter_; 97 char* send_codec_name_; 98 99 // Payload types for stereo codecs and CNG 100 #ifdef WEBRTC_CODEC_G722 101 int g722_pltype_; 102 #endif 103 int l16_8khz_pltype_; 104 int l16_16khz_pltype_; 105 int l16_32khz_pltype_; 106 #ifdef PCMA_AND_PCMU 107 int pcma_pltype_; 108 int pcmu_pltype_; 109 #endif 110 #ifdef WEBRTC_CODEC_OPUS 111 int opus_pltype_; 112 #endif 113 }; 114 115 } // namespace webrtc 116 117 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ 118
