| /* |
| * Copyright (c) 2012 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_TESTSTEREO_H_ |
| #define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ |
| |
| #include <math.h> |
| |
| #include <memory> |
| |
| #include "modules/audio_coding/test/ACMTest.h" |
| #include "modules/audio_coding/test/Channel.h" |
| #include "modules/audio_coding/test/PCMFile.h" |
| |
| #define PCMA_AND_PCMU |
| |
| namespace webrtc { |
| |
| enum StereoMonoMode { |
| kNotSet, |
| kMono, |
| kStereo |
| }; |
| |
| class TestPackStereo : public AudioPacketizationCallback { |
| public: |
| TestPackStereo(); |
| ~TestPackStereo(); |
| |
| void RegisterReceiverACM(AudioCodingModule* acm); |
| |
| int32_t SendData(const FrameType frame_type, |
| const uint8_t payload_type, |
| const uint32_t timestamp, |
| const uint8_t* payload_data, |
| const size_t payload_size, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| uint16_t payload_size(); |
| uint32_t timestamp_diff(); |
| void reset_payload_size(); |
| void set_codec_mode(StereoMonoMode mode); |
| void set_lost_packet(bool lost); |
| |
| private: |
| AudioCodingModule* receiver_acm_; |
| int16_t seq_no_; |
| uint32_t timestamp_diff_; |
| uint32_t last_in_timestamp_; |
| uint64_t total_bytes_; |
| int payload_size_; |
| StereoMonoMode codec_mode_; |
| // Simulate packet losses |
| bool lost_packet_; |
| }; |
| |
| class TestStereo : public ACMTest { |
| public: |
| explicit TestStereo(int test_mode); |
| ~TestStereo(); |
| |
| void Perform() override; |
| |
| private: |
| // The default value of '-1' indicates that the registration is based only on |
| // codec name and a sampling frequncy matching is not required. This is useful |
| // for codecs which support several sampling frequency. |
| void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz, |
| int rate, int pack_size, int channels, |
| int payload_type); |
| |
| void Run(TestPackStereo* channel, int in_channels, int out_channels, |
| int percent_loss = 0); |
| void OpenOutFile(int16_t test_number); |
| void DisplaySendReceiveCodec(); |
| |
| int test_mode_; |
| |
| std::unique_ptr<AudioCodingModule> acm_a_; |
| std::unique_ptr<AudioCodingModule> acm_b_; |
| |
| TestPackStereo* channel_a2b_; |
| |
| PCMFile* in_file_stereo_; |
| PCMFile* in_file_mono_; |
| PCMFile out_file_; |
| int16_t test_cntr_; |
| uint16_t pack_size_samp_; |
| uint16_t pack_size_bytes_; |
| int counter_; |
| char* send_codec_name_; |
| |
| // Payload types for stereo codecs and CNG |
| int g722_pltype_; |
| int l16_8khz_pltype_; |
| int l16_16khz_pltype_; |
| int l16_32khz_pltype_; |
| #ifdef PCMA_AND_PCMU |
| int pcma_pltype_; |
| int pcmu_pltype_; |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| int opus_pltype_; |
| #endif |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ |