acm_send_test.cc (5956B)
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 "modules/audio_coding/acm2/acm_send_test.h" 12 13 #include <cstdint> 14 #include <cstdio> 15 #include <cstring> 16 #include <memory> 17 #include <utility> 18 19 #include "absl/strings/match.h" 20 #include "absl/strings/str_cat.h" 21 #include "absl/strings/string_view.h" 22 #include "api/audio_codecs/audio_encoder.h" 23 #include "api/audio_codecs/audio_format.h" 24 #include "api/audio_codecs/builtin_audio_encoder_factory.h" 25 #include "api/environment/environment_factory.h" 26 #include "modules/audio_coding/include/audio_coding_module.h" 27 #include "modules/audio_coding/include/audio_coding_module_typedefs.h" 28 #include "modules/audio_coding/neteq/tools/input_audio_file.h" 29 #include "modules/rtp_rtcp/source/rtp_packet_received.h" 30 #include "rtc_base/checks.h" 31 32 namespace webrtc { 33 namespace test { 34 35 AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source, 36 int source_rate_hz, 37 int test_duration_ms) 38 : clock_(0), 39 env_(CreateEnvironment(&clock_)), 40 acm_(AudioCodingModule::Create()), 41 audio_source_(audio_source), 42 source_rate_hz_(source_rate_hz), 43 input_block_size_samples_( 44 static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)), 45 codec_registered_(false), 46 test_duration_ms_(test_duration_ms), 47 frame_type_(AudioFrameType::kAudioFrameSpeech), 48 payload_type_(0), 49 timestamp_(0), 50 sequence_number_(0) { 51 input_frame_.sample_rate_hz_ = source_rate_hz_; 52 input_frame_.num_channels_ = 1; 53 input_frame_.samples_per_channel_ = input_block_size_samples_; 54 RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_, 55 AudioFrame::kMaxDataSizeSamples); 56 acm_->RegisterTransportCallback(this); 57 } 58 59 AcmSendTestOldApi::~AcmSendTestOldApi() = default; 60 61 bool AcmSendTestOldApi::RegisterCodec(absl::string_view payload_name, 62 int clockrate_hz, 63 int num_channels, 64 int payload_type, 65 int frame_size_samples) { 66 SdpAudioFormat format(payload_name, clockrate_hz, num_channels); 67 if (absl::EqualsIgnoreCase(payload_name, "g722")) { 68 RTC_CHECK_EQ(16000, clockrate_hz); 69 format.clockrate_hz = 8000; 70 } else if (absl::EqualsIgnoreCase(payload_name, "opus")) { 71 RTC_CHECK(num_channels == 1 || num_channels == 2); 72 if (num_channels == 2) { 73 format.parameters["stereo"] = "1"; 74 } 75 format.num_channels = 2; 76 } 77 format.parameters["ptime"] = absl::StrCat( 78 CheckedDivExact(frame_size_samples, CheckedDivExact(clockrate_hz, 1000))); 79 auto factory = CreateBuiltinAudioEncoderFactory(); 80 acm_->SetEncoder( 81 factory->Create(env_, format, {.payload_type = payload_type})); 82 codec_registered_ = true; 83 input_frame_.num_channels_ = num_channels; 84 RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_, 85 AudioFrame::kMaxDataSizeSamples); 86 return codec_registered_; 87 } 88 89 void AcmSendTestOldApi::RegisterExternalCodec( 90 std::unique_ptr<AudioEncoder> external_speech_encoder) { 91 input_frame_.num_channels_ = external_speech_encoder->NumChannels(); 92 acm_->SetEncoder(std::move(external_speech_encoder)); 93 RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_, 94 AudioFrame::kMaxDataSizeSamples); 95 codec_registered_ = true; 96 } 97 98 std::unique_ptr<RtpPacketReceived> AcmSendTestOldApi::NextPacket() { 99 RTC_DCHECK(codec_registered_); 100 if (filter_.test(static_cast<size_t>(payload_type_))) { 101 // This payload type should be filtered out. Since the payload type is the 102 // same throughout the whole test run, no packet at all will be delivered. 103 // We can just as well signal that the test is over by returning NULL. 104 return nullptr; 105 } 106 // Insert audio and process until one packet is produced. 107 while (clock_.TimeInMilliseconds() < test_duration_ms_) { 108 clock_.AdvanceTimeMilliseconds(kBlockSizeMs); 109 RTC_CHECK(audio_source_->Read( 110 input_block_size_samples_ * input_frame_.num_channels_, 111 input_frame_.mutable_data())); 112 data_to_send_ = false; 113 RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0); 114 input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_); 115 if (data_to_send_) { 116 // Encoded packet received. 117 return CreatePacket(); 118 } 119 } 120 // Test ended. 121 return nullptr; 122 } 123 124 // This method receives the callback from ACM when a new packet is produced. 125 int32_t AcmSendTestOldApi::SendData( 126 AudioFrameType frame_type, 127 uint8_t payload_type, 128 uint32_t timestamp, 129 const uint8_t* payload_data, 130 size_t payload_len_bytes, 131 int64_t /* absolute_capture_timestamp_ms */) { 132 // Store the packet locally. 133 frame_type_ = frame_type; 134 payload_type_ = payload_type; 135 timestamp_ = timestamp; 136 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 137 RTC_DCHECK_EQ(last_payload_vec_.size(), payload_len_bytes); 138 data_to_send_ = true; 139 return 0; 140 } 141 142 std::unique_ptr<RtpPacketReceived> AcmSendTestOldApi::CreatePacket() { 143 auto rtp_packet = std::make_unique<RtpPacketReceived>(); 144 145 // Populate the header. 146 rtp_packet->SetPayloadType(payload_type_); 147 rtp_packet->SetSequenceNumber(sequence_number_); 148 rtp_packet->SetTimestamp(timestamp_); 149 rtp_packet->SetSsrc(0x12345678); 150 ++sequence_number_; 151 152 rtp_packet->SetPayload(last_payload_vec_); 153 rtp_packet->set_arrival_time(clock_.CurrentTime()); 154 return rtp_packet; 155 } 156 157 } // namespace test 158 } // namespace webrtc