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_oldapi.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 AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source, 27 int source_rate_hz, 28 int test_duration_ms) 29 : clock_(0), 30 acm_(webrtc::AudioCodingModule::Create(0, &clock_)), 31 audio_source_(audio_source), 32 source_rate_hz_(source_rate_hz), 33 input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000), 34 codec_registered_(false), 35 test_duration_ms_(test_duration_ms), 36 frame_type_(kAudioFrameSpeech), 37 payload_type_(0), 38 timestamp_(0), 39 sequence_number_(0) { 40 input_frame_.sample_rate_hz_ = source_rate_hz_; 41 input_frame_.num_channels_ = 1; 42 input_frame_.samples_per_channel_ = input_block_size_samples_; 43 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 44 AudioFrame::kMaxDataSizeSamples); 45 acm_->RegisterTransportCallback(this); 46 } 47 48 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name, 49 int sampling_freq_hz, 50 int channels, 51 int payload_type, 52 int frame_size_samples) { 53 CHECK_EQ(0, 54 AudioCodingModule::Codec( 55 payload_name, &codec_, sampling_freq_hz, channels)); 56 codec_.pltype = payload_type; 57 codec_.pacsize = frame_size_samples; 58 codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0); 59 input_frame_.num_channels_ = channels; 60 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 61 AudioFrame::kMaxDataSizeSamples); 62 return codec_registered_; 63 } 64 65 Packet* AcmSendTestOldApi::NextPacket() { 66 assert(codec_registered_); 67 if (filter_.test(payload_type_)) { 68 // This payload type should be filtered out. Since the payload type is the 69 // same throughout the whole test run, no packet at all will be delivered. 70 // We can just as well signal that the test is over by returning NULL. 71 return NULL; 72 } 73 // Insert audio and process until one packet is produced. 74 while (clock_.TimeInMilliseconds() < test_duration_ms_) { 75 clock_.AdvanceTimeMilliseconds(kBlockSizeMs); 76 CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_)); 77 if (input_frame_.num_channels_ > 1) { 78 InputAudioFile::DuplicateInterleaved(input_frame_.data_, 79 input_block_size_samples_, 80 input_frame_.num_channels_, 81 input_frame_.data_); 82 } 83 CHECK_EQ(0, acm_->Add10MsData(input_frame_)); 84 input_frame_.timestamp_ += input_block_size_samples_; 85 int32_t encoded_bytes = acm_->Process(); 86 if (encoded_bytes > 0) { 87 // Encoded packet received. 88 return CreatePacket(); 89 } 90 } 91 // Test ended. 92 return NULL; 93 } 94 95 // This method receives the callback from ACM when a new packet is produced. 96 int32_t AcmSendTestOldApi::SendData( 97 FrameType frame_type, 98 uint8_t payload_type, 99 uint32_t timestamp, 100 const uint8_t* payload_data, 101 uint16_t payload_len_bytes, 102 const RTPFragmentationHeader* fragmentation) { 103 // Store the packet locally. 104 frame_type_ = frame_type; 105 payload_type_ = payload_type; 106 timestamp_ = timestamp; 107 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 108 assert(last_payload_vec_.size() == payload_len_bytes); 109 return 0; 110 } 111 112 Packet* AcmSendTestOldApi::CreatePacket() { 113 const size_t kRtpHeaderSize = 12; 114 size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize; 115 uint8_t* packet_memory = new uint8_t[allocated_bytes]; 116 // Populate the header bytes. 117 packet_memory[0] = 0x80; 118 packet_memory[1] = payload_type_; 119 packet_memory[2] = (sequence_number_ >> 8) & 0xFF; 120 packet_memory[3] = (sequence_number_) & 0xFF; 121 packet_memory[4] = (timestamp_ >> 24) & 0xFF; 122 packet_memory[5] = (timestamp_ >> 16) & 0xFF; 123 packet_memory[6] = (timestamp_ >> 8) & 0xFF; 124 packet_memory[7] = timestamp_ & 0xFF; 125 // Set SSRC to 0x12345678. 126 packet_memory[8] = 0x12; 127 packet_memory[9] = 0x34; 128 packet_memory[10] = 0x56; 129 packet_memory[11] = 0x78; 130 131 ++sequence_number_; 132 133 // Copy the payload data. 134 memcpy(packet_memory + kRtpHeaderSize, 135 &last_payload_vec_[0], 136 last_payload_vec_.size()); 137 Packet* packet = 138 new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()); 139 assert(packet); 140 assert(packet->valid_header()); 141 return packet; 142 } 143 144 } // namespace test 145 } // namespace webrtc 146
