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1 /* | |
2 * Copyright (c) 2014 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_MAIN_ACM2_ACM_SEND_TEST_H_ | |
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ | |
13 | |
14 #include <vector> | |
15 | |
16 #include "webrtc/base/constructormagic.h" | |
17 #include "webrtc/base/scoped_ptr.h" | |
18 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" | |
19 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" | |
20 #include "webrtc/system_wrappers/include/clock.h" | |
21 | |
22 namespace webrtc { | |
23 class AudioEncoder; | |
24 | |
25 namespace test { | |
26 class InputAudioFile; | |
27 class Packet; | |
28 | |
29 class AcmSendTestOldApi : public AudioPacketizationCallback, | |
30 public PacketSource { | |
31 public: | |
32 AcmSendTestOldApi(InputAudioFile* audio_source, | |
33 int source_rate_hz, | |
34 int test_duration_ms); | |
35 virtual ~AcmSendTestOldApi() {} | |
36 | |
37 // Registers the send codec. Returns true on success, false otherwise. | |
38 bool RegisterCodec(const char* payload_name, | |
39 int sampling_freq_hz, | |
40 int channels, | |
41 int payload_type, | |
42 int frame_size_samples); | |
43 | |
44 // Registers an external send codec. Returns true on success, false otherwise. | |
45 bool RegisterExternalCodec(AudioEncoder* external_speech_encoder); | |
46 | |
47 // Returns the next encoded packet. Returns NULL if the test duration was | |
48 // exceeded. Ownership of the packet is handed over to the caller. | |
49 // Inherited from PacketSource. | |
50 Packet* NextPacket(); | |
51 | |
52 // Inherited from AudioPacketizationCallback. | |
53 int32_t SendData(FrameType frame_type, | |
54 uint8_t payload_type, | |
55 uint32_t timestamp, | |
56 const uint8_t* payload_data, | |
57 size_t payload_len_bytes, | |
58 const RTPFragmentationHeader* fragmentation) override; | |
59 | |
60 AudioCodingModule* acm() { return acm_.get(); } | |
61 | |
62 private: | |
63 static const int kBlockSizeMs = 10; | |
64 | |
65 // Creates a Packet object from the last packet produced by ACM (and received | |
66 // through the SendData method as a callback). Ownership of the new Packet | |
67 // object is transferred to the caller. | |
68 Packet* CreatePacket(); | |
69 | |
70 SimulatedClock clock_; | |
71 rtc::scoped_ptr<AudioCodingModule> acm_; | |
72 InputAudioFile* audio_source_; | |
73 int source_rate_hz_; | |
74 const size_t input_block_size_samples_; | |
75 AudioFrame input_frame_; | |
76 bool codec_registered_; | |
77 int test_duration_ms_; | |
78 // The following member variables are set whenever SendData() is called. | |
79 FrameType frame_type_; | |
80 int payload_type_; | |
81 uint32_t timestamp_; | |
82 uint16_t sequence_number_; | |
83 std::vector<uint8_t> last_payload_vec_; | |
84 bool data_to_send_; | |
85 | |
86 RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi); | |
87 }; | |
88 | |
89 } // namespace test | |
90 } // namespace webrtc | |
91 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ | |
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