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 #include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h" 12 13 #include <assert.h> 14 #include <stdio.h> 15 #include <string.h> 16 17 #include "testing/gtest/include/gtest/gtest.h" 18 #include "webrtc/base/checks.h" 19 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" 20 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" 21 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" 22 23 namespace webrtc { 24 namespace test { 25 26 AcmSendTest::AcmSendTest(InputAudioFile* audio_source, 27 int source_rate_hz, 28 int test_duration_ms) 29 : clock_(0), 30 audio_source_(audio_source), 31 source_rate_hz_(source_rate_hz), 32 input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000), 33 codec_registered_(false), 34 test_duration_ms_(test_duration_ms), 35 frame_type_(kAudioFrameSpeech), 36 payload_type_(0), 37 timestamp_(0), 38 sequence_number_(0) { 39 webrtc::AudioCoding::Config config; 40 config.clock = &clock_; 41 config.transport = this; 42 acm_.reset(webrtc::AudioCoding::Create(config)); 43 input_frame_.sample_rate_hz_ = source_rate_hz_; 44 input_frame_.num_channels_ = 1; 45 input_frame_.samples_per_channel_ = input_block_size_samples_; 46 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 47 AudioFrame::kMaxDataSizeSamples); 48 } 49 50 bool AcmSendTest::RegisterCodec(int codec_type, 51 int channels, 52 int payload_type, 53 int frame_size_samples) { 54 codec_registered_ = 55 acm_->RegisterSendCodec(codec_type, payload_type, frame_size_samples); 56 input_frame_.num_channels_ = channels; 57 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 58 AudioFrame::kMaxDataSizeSamples); 59 return codec_registered_; 60 } 61 62 Packet* AcmSendTest::NextPacket() { 63 assert(codec_registered_); 64 if (filter_.test(payload_type_)) { 65 // This payload type should be filtered out. Since the payload type is the 66 // same throughout the whole test run, no packet at all will be delivered. 67 // We can just as well signal that the test is over by returning NULL. 68 return NULL; 69 } 70 // Insert audio and process until one packet is produced. 71 while (clock_.TimeInMilliseconds() < test_duration_ms_) { 72 clock_.AdvanceTimeMilliseconds(kBlockSizeMs); 73 CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_)); 74 if (input_frame_.num_channels_ > 1) { 75 InputAudioFile::DuplicateInterleaved(input_frame_.data_, 76 input_block_size_samples_, 77 input_frame_.num_channels_, 78 input_frame_.data_); 79 } 80 int32_t encoded_bytes = acm_->Add10MsAudio(input_frame_); 81 EXPECT_GE(encoded_bytes, 0); 82 input_frame_.timestamp_ += input_block_size_samples_; 83 if (encoded_bytes > 0) { 84 // Encoded packet received. 85 return CreatePacket(); 86 } 87 } 88 // Test ended. 89 return NULL; 90 } 91 92 // This method receives the callback from ACM when a new packet is produced. 93 int32_t AcmSendTest::SendData(FrameType frame_type, 94 uint8_t payload_type, 95 uint32_t timestamp, 96 const uint8_t* payload_data, 97 uint16_t payload_len_bytes, 98 const RTPFragmentationHeader* fragmentation) { 99 // Store the packet locally. 100 frame_type_ = frame_type; 101 payload_type_ = payload_type; 102 timestamp_ = timestamp; 103 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 104 assert(last_payload_vec_.size() == payload_len_bytes); 105 return 0; 106 } 107 108 Packet* AcmSendTest::CreatePacket() { 109 const size_t kRtpHeaderSize = 12; 110 size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize; 111 uint8_t* packet_memory = new uint8_t[allocated_bytes]; 112 // Populate the header bytes. 113 packet_memory[0] = 0x80; 114 packet_memory[1] = payload_type_; 115 packet_memory[2] = (sequence_number_ >> 8) & 0xFF; 116 packet_memory[3] = (sequence_number_) & 0xFF; 117 packet_memory[4] = (timestamp_ >> 24) & 0xFF; 118 packet_memory[5] = (timestamp_ >> 16) & 0xFF; 119 packet_memory[6] = (timestamp_ >> 8) & 0xFF; 120 packet_memory[7] = timestamp_ & 0xFF; 121 // Set SSRC to 0x12345678. 122 packet_memory[8] = 0x12; 123 packet_memory[9] = 0x34; 124 packet_memory[10] = 0x56; 125 packet_memory[11] = 0x78; 126 127 ++sequence_number_; 128 129 // Copy the payload data. 130 memcpy(packet_memory + kRtpHeaderSize, 131 &last_payload_vec_[0], 132 last_payload_vec_.size()); 133 Packet* packet = 134 new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()); 135 assert(packet); 136 assert(packet->valid_header()); 137 return packet; 138 } 139 140 } // namespace test 141 } // namespace webrtc 142
