| /* | |
| * Copyright (c) 2014 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_ACM2_ACM_SEND_TEST_H_ | |
| #define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_ | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <memory> | |
| #include <vector> | |
| #include "absl/strings/string_view.h" | |
| #include "api/audio/audio_frame.h" | |
| #include "api/environment/environment.h" | |
| #include "modules/audio_coding/include/audio_coding_module.h" | |
| #include "modules/audio_coding/include/audio_coding_module_typedefs.h" | |
| #include "modules/audio_coding/neteq/tools/packet_source.h" | |
| #include "system_wrappers/include/clock.h" | |
| namespace webrtc { | |
| class AudioEncoder; | |
| namespace test { | |
| class InputAudioFile; | |
| class Packet; | |
| class AcmSendTestOldApi : public AudioPacketizationCallback, | |
| public PacketSource { | |
| public: | |
| AcmSendTestOldApi(InputAudioFile* audio_source, | |
| int source_rate_hz, | |
| int test_duration_ms); | |
| ~AcmSendTestOldApi() override; | |
| AcmSendTestOldApi(const AcmSendTestOldApi&) = delete; | |
| AcmSendTestOldApi& operator=(const AcmSendTestOldApi&) = delete; | |
| // Registers the send codec. Returns true on success, false otherwise. | |
| bool RegisterCodec(absl::string_view payload_name, | |
| int sampling_freq_hz, | |
| int channels, | |
| int payload_type, | |
| int frame_size_samples); | |
| // Registers an external send codec. | |
| void RegisterExternalCodec( | |
| std::unique_ptr<AudioEncoder> external_speech_encoder); | |
| // Inherited from PacketSource. | |
| std::unique_ptr<Packet> NextPacket() override; | |
| // Inherited from AudioPacketizationCallback. | |
| 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; | |
| AudioCodingModule* acm() { return acm_.get(); } | |
| private: | |
| static const int kBlockSizeMs = 10; | |
| // Creates a Packet object from the last packet produced by ACM (and received | |
| // through the SendData method as a callback). | |
| std::unique_ptr<Packet> CreatePacket(); | |
| SimulatedClock clock_; | |
| const Environment env_; | |
| std::unique_ptr<AudioCodingModule> acm_; | |
| InputAudioFile* audio_source_; | |
| int source_rate_hz_; | |
| const size_t input_block_size_samples_; | |
| AudioFrame input_frame_; | |
| bool codec_registered_; | |
| int test_duration_ms_; | |
| // The following member variables are set whenever SendData() is called. | |
| AudioFrameType frame_type_; | |
| int payload_type_; | |
| uint32_t timestamp_; | |
| uint16_t sequence_number_; | |
| std::vector<uint8_t> last_payload_vec_; | |
| bool data_to_send_; | |
| }; | |
| } // namespace test | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_ |