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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.cc62
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,