src/third_party/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc

   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_receive_test.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/main/interface/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 AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
  26                                AudioSink* audio_sink,
  27                                int output_freq_hz,
  28                                NumOutputChannels exptected_output_channels)
  29     : clock_(0),
  30       packet_source_(packet_source),
  31       audio_sink_(audio_sink),
  32       output_freq_hz_(output_freq_hz),
  33       exptected_output_channels_(exptected_output_channels) {
  34   webrtc::AudioCoding::Config config;
  35   config.clock = &clock_;
  36   config.playout_frequency_hz = output_freq_hz_;
  37   acm_.reset(webrtc::AudioCoding::Create(config));
  38 }
  39
  40 void AcmReceiveTest::RegisterDefaultCodecs() {
  41   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
  42   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
  43 #ifndef WEBRTC_ANDROID
  44   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
  45   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105));
  46 #endif
  47   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107));
  48   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108));
  49   ASSERT_TRUE(
  50       acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109));
  51   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111));
  52   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112));
  53   ASSERT_TRUE(
  54       acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113));
  55   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
  56   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
  57   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
  58   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
  59   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
  60   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
  61   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
  62   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
  63   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
  64   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
  65   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
  66 }
  67
  68 void AcmReceiveTest::RegisterNetEqTestCodecs() {
  69   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
  70 #ifndef WEBRTC_ANDROID
  71   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
  72   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 124));
  73 #endif
  74   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93));
  75   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94));
  76   ASSERT_TRUE(
  77       acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95));
  78   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
  79   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
  80   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
  81   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
  82   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
  83   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
  84   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
  85   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117));
  86 }
  87
  88 void AcmReceiveTest::Run() {
  89   for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
  90        packet.reset(packet_source_->NextPacket())) {
  91     // Pull audio until time to insert packet.
  92     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
  93       AudioFrame output_frame;
  94       EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
  95       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
  96       const int samples_per_block = output_freq_hz_ * 10 / 1000;
  97       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
  98       if (exptected_output_channels_ != kArbitraryChannels) {
  99         if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
 100           // Don't check number of channels for PLC output, since each test run
 101           // usually starts with a short period of mono PLC before decoding the
 102           // first packet.
 103         } else {
 104           EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
 105         }
 106       }
 107       ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
 108       clock_.AdvanceTimeMilliseconds(10);
 109     }
 110
 111     // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
 112     WebRtcRTPHeader header;
 113     header.header = packet->header();
 114     header.frameType = kAudioFrameSpeech;
 115     memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
 116     EXPECT_TRUE(
 117         acm_->InsertPacket(packet->payload(),
 118                            static_cast<int32_t>(packet->payload_length_bytes()),
 119                            header))
 120         << "Failure when inserting packet:" << std::endl
 121         << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
 122         << "  TS = " << header.header.timestamp << std::endl
 123         << "  SN = " << header.header.sequenceNumber;
 124   }
 125 }
 126
 127 }  // namespace test
 128 }  // namespace webrtc