media/cast/audio_receiver/audio_decoder_unittest.cc

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "base/memory/ref_counted.h"
   6 #include "media/cast/audio_receiver/audio_decoder.h"
   7 #include "testing/gmock/include/gmock/gmock.h"
   8
   9 namespace media {
  10 namespace cast {
  11
  12 class AudioDecoderTest : public ::testing::Test {
  13  protected:
  14   AudioDecoderTest() {}
  15   virtual ~AudioDecoderTest() {}
  16
  17   void Configure(const AudioReceiverConfig& audio_config) {
  18     audio_decoder_ = new AudioDecoder(audio_config);
  19   }
  20
  21   scoped_refptr<AudioDecoder> audio_decoder_;
  22 };
  23
  24 TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
  25   AudioReceiverConfig audio_config;
  26   audio_config.rtp_payload_type = 127;
  27   audio_config.frequency = 16000;
  28   audio_config.channels = 1;
  29   audio_config.codec = kPcm16;
  30   audio_config.use_external_decoder = false;
  31   Configure(audio_config);
  32
  33   RtpCastHeader rtp_header;
  34   rtp_header.webrtc.header.payloadType = 127;
  35   rtp_header.webrtc.header.sequenceNumber = 1234;
  36   rtp_header.webrtc.header.timestamp = 0x87654321;
  37   rtp_header.webrtc.header.ssrc = 0x12345678;
  38   rtp_header.webrtc.header.paddingLength = 0;
  39   rtp_header.webrtc.header.headerLength = 12;
  40   rtp_header.webrtc.type.Audio.channel = 1;
  41   rtp_header.webrtc.type.Audio.isCNG = false;
  42
  43   std::vector<int16> payload(640, 0x1234);
  44
  45   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
  46   size_t payload_size = payload.size() * sizeof(int16);
  47
  48   audio_decoder_->IncomingParsedRtpPacket(payload_data,
  49       payload_size, rtp_header);
  50
  51   int number_of_10ms_blocks = 4;
  52   int desired_frequency = 16000;
  53   PcmAudioFrame audio_frame;
  54   uint32 rtp_timestamp;
  55
  56   EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
  57                                                desired_frequency,
  58                                                &audio_frame,
  59                                                &rtp_timestamp));
  60
  61   EXPECT_EQ(1, audio_frame.channels);
  62   EXPECT_EQ(16000, audio_frame.frequency);
  63   EXPECT_EQ(640ul, audio_frame.samples.size());
  64   // First 10 samples per channel are 0 from NetEq.
  65   for (size_t i = 10; i < audio_frame.samples.size(); ++i) {
  66     EXPECT_EQ(0x3412, audio_frame.samples[i]);
  67   }
  68 }
  69
  70 TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
  71   AudioReceiverConfig audio_config;
  72   audio_config.rtp_payload_type = 127;
  73   audio_config.frequency = 16000;
  74   audio_config.channels = 2;
  75   audio_config.codec = kPcm16;
  76   audio_config.use_external_decoder = false;
  77   Configure(audio_config);
  78
  79   RtpCastHeader rtp_header;
  80   rtp_header.webrtc.header.payloadType = 127;
  81   rtp_header.webrtc.header.sequenceNumber = 1234;
  82   rtp_header.webrtc.header.timestamp = 0x87654321;
  83   rtp_header.webrtc.header.ssrc = 0x12345678;
  84   rtp_header.webrtc.header.paddingLength = 0;
  85   rtp_header.webrtc.header.headerLength = 12;
  86
  87   rtp_header.webrtc.type.Audio.isCNG = false;
  88   rtp_header.webrtc.type.Audio.channel = 2;
  89
  90   std::vector<int16> payload(640, 0x1234);
  91
  92   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
  93   size_t payload_size = payload.size() * sizeof(int16);
  94
  95   audio_decoder_->IncomingParsedRtpPacket(payload_data,
  96       payload_size, rtp_header);
  97
  98   int number_of_10ms_blocks = 2;
  99   int desired_frequency = 16000;
 100   PcmAudioFrame audio_frame;
 101   uint32 rtp_timestamp;
 102
 103   EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
 104                                                desired_frequency,
 105                                                &audio_frame,
 106                                                &rtp_timestamp));
 107
 108   EXPECT_EQ(2, audio_frame.channels);
 109   EXPECT_EQ(16000, audio_frame.frequency);
 110   EXPECT_EQ(640ul, audio_frame.samples.size());
 111   // First 10 samples per channel are 0 from NetEq.
 112   for (size_t i = 10 * audio_config.channels; i < audio_frame.samples.size();
 113       ++i) {
 114     EXPECT_EQ(0x3412, audio_frame.samples[i]);
 115   }
 116
 117   rtp_header.webrtc.header.sequenceNumber++;
 118   rtp_header.webrtc.header.timestamp += (audio_config.frequency / 100) * 2 * 2;
 119
 120   audio_decoder_->IncomingParsedRtpPacket(payload_data,
 121       payload_size, rtp_header);
 122
 123   EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
 124                                                desired_frequency,
 125                                                &audio_frame,
 126                                                &rtp_timestamp));
 127   EXPECT_EQ(2, audio_frame.channels);
 128   EXPECT_EQ(16000, audio_frame.frequency);
 129   EXPECT_EQ(640ul, audio_frame.samples.size());
 130   for (size_t i = 0; i < audio_frame.samples.size(); ++i) {
 131     EXPECT_NEAR(0x3412, audio_frame.samples[i], 1000);
 132   }
 133 }
 134
 135 TEST_F(AudioDecoderTest, Pcm16Resample) {
 136   AudioReceiverConfig audio_config;
 137   audio_config.rtp_payload_type = 127;
 138   audio_config.frequency = 16000;
 139   audio_config.channels = 2;
 140   audio_config.codec = kPcm16;
 141   audio_config.use_external_decoder = false;
 142   Configure(audio_config);
 143
 144   RtpCastHeader rtp_header;
 145   rtp_header.webrtc.header.payloadType = 127;
 146   rtp_header.webrtc.header.sequenceNumber = 1234;
 147   rtp_header.webrtc.header.timestamp = 0x87654321;
 148   rtp_header.webrtc.header.ssrc = 0x12345678;
 149   rtp_header.webrtc.header.paddingLength = 0;
 150   rtp_header.webrtc.header.headerLength = 12;
 151
 152   rtp_header.webrtc.type.Audio.isCNG = false;
 153   rtp_header.webrtc.type.Audio.channel = 2;
 154
 155   std::vector<int16> payload(640, 0x1234);
 156
 157   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
 158   size_t payload_size = payload.size() * sizeof(int16);
 159
 160   audio_decoder_->IncomingParsedRtpPacket(payload_data,
 161       payload_size, rtp_header);
 162
 163   int number_of_10ms_blocks = 2;
 164   int desired_frequency = 48000;
 165   PcmAudioFrame audio_frame;
 166   uint32 rtp_timestamp;
 167
 168   EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
 169                                                desired_frequency,
 170                                                &audio_frame,
 171                                                &rtp_timestamp));
 172
 173   EXPECT_EQ(2, audio_frame.channels);
 174   EXPECT_EQ(48000, audio_frame.frequency);
 175   EXPECT_EQ(1920ul, audio_frame.samples.size());  // Upsampled to 48 KHz.
 176   int count = 0;
 177   // Resampling makes the variance worse.
 178   for (size_t i = 100 * audio_config.channels; i < audio_frame.samples.size();
 179       ++i) {
 180     EXPECT_NEAR(0x3412, audio_frame.samples[i], 400);
 181     if (0x3412 == audio_frame.samples[i])  count++;
 182   }
 183 }
 184
 185 }  // namespace cast
 186 }  // namespace media