diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc | 62 |
1 files changed, 8 insertions, 54 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index 712eeb26877..4234f146474 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc @@ -16,7 +16,8 @@ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" -#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/test_suite.h" #include "webrtc/test/testsupport/fileutils.h" @@ -42,12 +43,14 @@ class AcmReceiverTest : public AudioPacketizationCallback, public ::testing::Test { protected: AcmReceiverTest() - : receiver_(new AcmReceiver), - acm_(new AudioCodingModuleImpl(0)), - timestamp_(0), + : timestamp_(0), packet_sent_(false), last_packet_send_timestamp_(timestamp_), - last_frame_type_(kFrameEmpty) {} + last_frame_type_(kFrameEmpty) { + AudioCodingModule::Config config; + acm_.reset(new AudioCodingModuleImpl(config)); + receiver_.reset(new AcmReceiver(config)); + } ~AcmReceiverTest() {} @@ -302,55 +305,6 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) { EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_); } -TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(FlushBuffer)) { - const int id = ACMCodecDB::kISAC; - EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, - NULL)); - const int kNumPackets = 5; - const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); - for (int n = 0; n < kNumPackets; ++n) - InsertOnePacketOfSilence(id); - ACMNetworkStatistics statistics; - receiver_->NetworkStatistics(&statistics); - ASSERT_EQ(num_10ms_frames * kNumPackets * 10, statistics.currentBufferSize); - - receiver_->FlushBuffers(); - receiver_->NetworkStatistics(&statistics); - ASSERT_EQ(0, statistics.currentBufferSize); -} - -TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PlayoutTimestamp)) { - const int id = ACMCodecDB::kPCM16Bwb; - EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, - NULL)); - receiver_->SetPlayoutMode(fax); - const int kNumPackets = 5; - const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); - uint32_t expected_timestamp; - AudioFrame frame; - int ts_offset = 0; - bool first_audio_frame = true; - for (int n = 0; n < kNumPackets; ++n) { - packet_sent_ = false; - InsertOnePacketOfSilence(id); - ASSERT_TRUE(packet_sent_); - expected_timestamp = last_packet_send_timestamp_; - for (int k = 0; k < num_10ms_frames; ++k) { - ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); - if (first_audio_frame) { - // There is an offset in playout timestamps. Perhaps, it is related to - // initial delay that NetEq applies - ts_offset = receiver_->PlayoutTimestamp() - expected_timestamp; - first_audio_frame = false; - } else { - EXPECT_EQ(expected_timestamp + ts_offset, - receiver_->PlayoutTimestamp()); - } - expected_timestamp += codecs_[id].plfreq / 100; // Increment by 10 ms. - } - } -} - TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { const int kCodecId[] = { ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB, |