| /* |
| * 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_TESTALLCODECS_H_ |
| #define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ |
| |
| #include <memory> |
| |
| #include "modules/audio_coding/acm2/acm_receiver.h" |
| #include "modules/audio_coding/include/audio_coding_module.h" |
| #include "modules/audio_coding/test/PCMFile.h" |
| |
| namespace webrtc { |
| |
| class TestPack : public AudioPacketizationCallback { |
| public: |
| TestPack(); |
| ~TestPack(); |
| |
| void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver); |
| |
| int32_t SendData(AudioFrameType frame_type, |
| uint8_t payload_type, |
| uint32_t timestamp, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| int64_t absolute_capture_timestamp_ms) override; |
| |
| size_t payload_size(); |
| uint32_t timestamp_diff(); |
| void reset_payload_size(); |
| |
| private: |
| acm2::AcmReceiver* receiver_acm_; |
| uint16_t sequence_number_; |
| uint8_t payload_data_[60 * 32 * 2 * 2]; |
| uint32_t timestamp_diff_; |
| uint32_t last_in_timestamp_; |
| uint64_t total_bytes_; |
| size_t payload_size_; |
| }; |
| |
| class TestAllCodecs { |
| public: |
| TestAllCodecs(); |
| ~TestAllCodecs(); |
| |
| void Perform(); |
| |
| private: |
| // The default value of '-1' indicates that the registration is based only on |
| // codec name, and a sampling frequency matching is not required. |
| // This is useful for codecs which support several sampling frequency. |
| // Note! Only mono mode is tested in this test. |
| void RegisterSendCodec(char* codec_name, |
| int32_t sampling_freq_hz, |
| int rate, |
| int packet_size, |
| size_t extra_byte); |
| |
| void Run(TestPack* channel); |
| void OpenOutFile(int test_number); |
| |
| std::unique_ptr<AudioCodingModule> acm_a_; |
| std::unique_ptr<acm2::AcmReceiver> acm_b_; |
| TestPack* channel_a_to_b_; |
| PCMFile infile_a_; |
| PCMFile outfile_b_; |
| int test_count_; |
| int packet_size_samples_; |
| size_t packet_size_bytes_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ |