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