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 <stdio.h> 12 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h" 13 14 namespace webrtc { 15 namespace test { 16 17 const uint8_t kPayloadType = 95; 18 const int kOutputSizeMs = 10; 19 20 NetEqQualityTest::NetEqQualityTest(int block_duration_ms, 21 int in_sampling_khz, 22 int out_sampling_khz, 23 enum NetEqDecoder decoder_type, 24 int channels, 25 double drift_factor, 26 std::string in_filename, 27 std::string out_filename) 28 : decoded_time_ms_(0), 29 decodable_time_ms_(0), 30 drift_factor_(drift_factor), 31 block_duration_ms_(block_duration_ms), 32 in_sampling_khz_(in_sampling_khz), 33 out_sampling_khz_(out_sampling_khz), 34 decoder_type_(decoder_type), 35 channels_(channels), 36 in_filename_(in_filename), 37 out_filename_(out_filename), 38 in_size_samples_(in_sampling_khz_ * block_duration_ms_), 39 out_size_samples_(out_sampling_khz_ * kOutputSizeMs), 40 payload_size_bytes_(0), 41 max_payload_bytes_(0), 42 in_file_(new InputAudioFile(in_filename_)), 43 out_file_(NULL), 44 rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0, 45 decodable_time_ms_)) { 46 NetEq::Config config; 47 config.sample_rate_hz = out_sampling_khz_ * 1000; 48 neteq_.reset(NetEq::Create(config)); 49 max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t); 50 in_data_.reset(new int16_t[in_size_samples_ * channels_]); 51 payload_.reset(new uint8_t[max_payload_bytes_]); 52 out_data_.reset(new int16_t[out_size_samples_ * channels_]); 53 } 54 55 void NetEqQualityTest::SetUp() { 56 out_file_ = fopen(out_filename_.c_str(), "wb"); 57 ASSERT_TRUE(out_file_ != NULL); 58 ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType)); 59 rtp_generator_->set_drift_factor(drift_factor_); 60 } 61 62 void NetEqQualityTest::TearDown() { 63 fclose(out_file_); 64 } 65 66 int NetEqQualityTest::Transmit() { 67 int packet_input_time_ms = 68 rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_, 69 &rtp_header_); 70 if (!PacketLost(packet_input_time_ms) && payload_size_bytes_ > 0) { 71 int ret = neteq_->InsertPacket(rtp_header_, &payload_[0], 72 payload_size_bytes_, 73 packet_input_time_ms * in_sampling_khz_); 74 if (ret != NetEq::kOK) 75 return -1; 76 } 77 return packet_input_time_ms; 78 } 79 80 int NetEqQualityTest::DecodeBlock() { 81 int channels; 82 int samples; 83 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0], 84 &samples, &channels, NULL); 85 86 if (ret != NetEq::kOK) { 87 return -1; 88 } else { 89 assert(channels == channels_); 90 assert(samples == kOutputSizeMs * out_sampling_khz_); 91 fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_); 92 return samples; 93 } 94 } 95 96 void NetEqQualityTest::Simulate(int end_time_ms) { 97 int audio_size_samples; 98 99 while (decoded_time_ms_ < end_time_ms) { 100 while (decodable_time_ms_ - kOutputSizeMs < decoded_time_ms_) { 101 ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0])); 102 payload_size_bytes_ = EncodeBlock(&in_data_[0], 103 in_size_samples_, &payload_[0], 104 max_payload_bytes_); 105 decodable_time_ms_ = Transmit() + block_duration_ms_; 106 } 107 audio_size_samples = DecodeBlock(); 108 if (audio_size_samples > 0) { 109 decoded_time_ms_ += audio_size_samples / out_sampling_khz_; 110 } 111 } 112 } 113 114 } // namespace test 115 } // namespace webrtc 116
