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/acm2/acm_receive_test_oldapi.h"
12
13 #include <assert.h>
14 #include <stdio.h>
15
16 #include "testing/gtest/include/gtest/gtest.h"
17 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
18 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
19 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
20 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
21
22 namespace webrtc {
23 namespace test {
24
25 namespace {
26 // Returns true if the codec should be registered, otherwise false. Changes
27 // the number of channels for the Opus codec to always be 1.
ModifyAndUseThisCodec(CodecInst * codec_param)28 bool ModifyAndUseThisCodec(CodecInst* codec_param) {
29 if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
30 codec_param->plfreq == 48000)
31 return false; // Skip 48 kHz comfort noise.
32
33 if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
34 return false; // Skip DTFM.
35
36 return true;
37 }
38
39 // Remaps payload types from ACM's default to those used in the resource file
40 // neteq_universal_new.rtp. Returns true if the codec should be registered,
41 // otherwise false. The payload types are set as follows (all are mono codecs):
42 // PCMu = 0;
43 // PCMa = 8;
44 // Comfort noise 8 kHz = 13
45 // Comfort noise 16 kHz = 98
46 // Comfort noise 32 kHz = 99
47 // iLBC = 102
48 // iSAC wideband = 103
49 // iSAC super-wideband = 104
50 // AVT/DTMF = 106
51 // RED = 117
52 // PCM16b 8 kHz = 93
53 // PCM16b 16 kHz = 94
54 // PCM16b 32 kHz = 95
55 // G.722 = 94
RemapPltypeAndUseThisCodec(const char * plname,int plfreq,size_t channels,int * pltype)56 bool RemapPltypeAndUseThisCodec(const char* plname,
57 int plfreq,
58 size_t channels,
59 int* pltype) {
60 if (channels != 1)
61 return false; // Don't use non-mono codecs.
62
63 // Re-map pltypes to those used in the NetEq test files.
64 if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
65 *pltype = 0;
66 } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
67 *pltype = 8;
68 } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
69 *pltype = 13;
70 } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
71 *pltype = 98;
72 } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
73 *pltype = 99;
74 } else if (STR_CASE_CMP(plname, "ILBC") == 0) {
75 *pltype = 102;
76 } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
77 *pltype = 103;
78 } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
79 *pltype = 104;
80 } else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
81 *pltype = 106;
82 } else if (STR_CASE_CMP(plname, "red") == 0) {
83 *pltype = 117;
84 } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
85 *pltype = 93;
86 } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
87 *pltype = 94;
88 } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
89 *pltype = 95;
90 } else if (STR_CASE_CMP(plname, "G722") == 0) {
91 *pltype = 9;
92 } else {
93 // Don't use any other codecs.
94 return false;
95 }
96 return true;
97 }
98 } // namespace
99
AcmReceiveTestOldApi(PacketSource * packet_source,AudioSink * audio_sink,int output_freq_hz,NumOutputChannels exptected_output_channels)100 AcmReceiveTestOldApi::AcmReceiveTestOldApi(
101 PacketSource* packet_source,
102 AudioSink* audio_sink,
103 int output_freq_hz,
104 NumOutputChannels exptected_output_channels)
105 : clock_(0),
106 acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
107 packet_source_(packet_source),
108 audio_sink_(audio_sink),
109 output_freq_hz_(output_freq_hz),
110 exptected_output_channels_(exptected_output_channels) {
111 }
112
RegisterDefaultCodecs()113 void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
114 CodecInst my_codec_param;
115 for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
116 ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
117 if (ModifyAndUseThisCodec(&my_codec_param)) {
118 ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
119 << "Couldn't register receive codec.\n";
120 }
121 }
122 }
123
RegisterNetEqTestCodecs()124 void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
125 CodecInst my_codec_param;
126 for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
127 ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
128 if (!ModifyAndUseThisCodec(&my_codec_param)) {
129 // Skip this codec.
130 continue;
131 }
132
133 if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
134 my_codec_param.plfreq,
135 my_codec_param.channels,
136 &my_codec_param.pltype)) {
137 ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
138 << "Couldn't register receive codec.\n";
139 }
140 }
141 }
142
RegisterExternalReceiveCodec(int rtp_payload_type,AudioDecoder * external_decoder,int sample_rate_hz,int num_channels,const std::string & name)143 int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
144 int rtp_payload_type,
145 AudioDecoder* external_decoder,
146 int sample_rate_hz,
147 int num_channels,
148 const std::string& name) {
149 return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
150 sample_rate_hz, num_channels, name);
151 }
152
Run()153 void AcmReceiveTestOldApi::Run() {
154 for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
155 packet.reset(packet_source_->NextPacket())) {
156 // Pull audio until time to insert packet.
157 while (clock_.TimeInMilliseconds() < packet->time_ms()) {
158 AudioFrame output_frame;
159 EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
160 EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
161 const size_t samples_per_block =
162 static_cast<size_t>(output_freq_hz_ * 10 / 1000);
163 EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
164 if (exptected_output_channels_ != kArbitraryChannels) {
165 if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
166 // Don't check number of channels for PLC output, since each test run
167 // usually starts with a short period of mono PLC before decoding the
168 // first packet.
169 } else {
170 EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
171 }
172 }
173 ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
174 clock_.AdvanceTimeMilliseconds(10);
175 AfterGetAudio();
176 }
177
178 // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
179 WebRtcRTPHeader header;
180 header.header = packet->header();
181 header.frameType = kAudioFrameSpeech;
182 memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
183 EXPECT_EQ(0,
184 acm_->IncomingPacket(
185 packet->payload(),
186 static_cast<int32_t>(packet->payload_length_bytes()),
187 header))
188 << "Failure when inserting packet:" << std::endl
189 << " PT = " << static_cast<int>(header.header.payloadType) << std::endl
190 << " TS = " << header.header.timestamp << std::endl
191 << " SN = " << header.header.sequenceNumber;
192 }
193 }
194
AcmReceiveTestToggleOutputFreqOldApi(PacketSource * packet_source,AudioSink * audio_sink,int output_freq_hz_1,int output_freq_hz_2,int toggle_period_ms,NumOutputChannels exptected_output_channels)195 AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
196 PacketSource* packet_source,
197 AudioSink* audio_sink,
198 int output_freq_hz_1,
199 int output_freq_hz_2,
200 int toggle_period_ms,
201 NumOutputChannels exptected_output_channels)
202 : AcmReceiveTestOldApi(packet_source,
203 audio_sink,
204 output_freq_hz_1,
205 exptected_output_channels),
206 output_freq_hz_1_(output_freq_hz_1),
207 output_freq_hz_2_(output_freq_hz_2),
208 toggle_period_ms_(toggle_period_ms),
209 last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
210 }
211
AfterGetAudio()212 void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
213 if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
214 output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
215 ? output_freq_hz_2_
216 : output_freq_hz_1_;
217 last_toggle_time_ms_ = clock_.TimeInMilliseconds();
218 }
219 }
220
221 } // namespace test
222 } // namespace webrtc
223