Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
new file mode 100644
index 0000000..c0a0fd3
--- /dev/null
+++ b/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
@@ -0,0 +1,208 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Test to verify correct operation for externally created decoders.
+
+#include <string>
+#include <list>
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+using ::testing::_;
+
+// This test encodes a few packets of PCM16b 32 kHz data and inserts it into two
+// different NetEq instances. The first instance uses the internal version of
+// the decoder object, while the second one uses an externally created decoder
+// object (ExternalPcm16B wrapped in MockExternalPcm16B, both defined above).
+// The test verifies that the output from both instances match.
+class NetEqExternalDecoderTest : public ::testing::Test {
+ protected:
+ static const int kTimeStepMs = 10;
+ static const int kMaxBlockSize = 480; // 10 ms @ 48 kHz.
+ static const uint8_t kPayloadType = 95;
+ static const int kSampleRateHz = 32000;
+
+ NetEqExternalDecoderTest()
+ : sample_rate_hz_(kSampleRateHz),
+ samples_per_ms_(sample_rate_hz_ / 1000),
+ frame_size_ms_(10),
+ frame_size_samples_(frame_size_ms_ * samples_per_ms_),
+ output_size_samples_(frame_size_ms_ * samples_per_ms_),
+ neteq_external_(NetEq::Create(sample_rate_hz_)),
+ neteq_(NetEq::Create(sample_rate_hz_)),
+ external_decoder_(new MockExternalPcm16B(kDecoderPCM16Bswb32kHz)),
+ rtp_generator_(samples_per_ms_),
+ payload_size_bytes_(0),
+ last_send_time_(0),
+ last_arrival_time_(0) {
+ input_ = new int16_t[frame_size_samples_];
+ encoded_ = new uint8_t[2 * frame_size_samples_];
+ }
+
+ ~NetEqExternalDecoderTest() {
+ delete neteq_external_;
+ delete neteq_;
+ // We will now delete the decoder ourselves, so expecting Die to be called.
+ EXPECT_CALL(*external_decoder_, Die()).Times(1);
+ delete external_decoder_;
+ delete [] input_;
+ delete [] encoded_;
+ }
+
+ virtual void SetUp() {
+ const std::string file_name =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ input_file_.reset(new test::InputAudioFile(file_name));
+ assert(sample_rate_hz_ == 32000);
+ NetEqDecoder decoder = kDecoderPCM16Bswb32kHz;
+ EXPECT_CALL(*external_decoder_, Init());
+ // NetEq is not allowed to delete the external decoder (hence Times(0)).
+ EXPECT_CALL(*external_decoder_, Die()).Times(0);
+ ASSERT_EQ(NetEq::kOK,
+ neteq_external_->RegisterExternalDecoder(external_decoder_,
+ decoder,
+ sample_rate_hz_,
+ kPayloadType));
+ ASSERT_EQ(NetEq::kOK,
+ neteq_->RegisterPayloadType(decoder, kPayloadType));
+ }
+
+ virtual void TearDown() {}
+
+ int GetNewPackets() {
+ if (!input_file_->Read(frame_size_samples_, input_)) {
+ return -1;
+ }
+ payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
+ encoded_);
+ if (frame_size_samples_ * 2 != payload_size_bytes_) {
+ return -1;
+ }
+ int next_send_time = rtp_generator_.GetRtpHeader(kPayloadType,
+ frame_size_samples_,
+ &rtp_header_);
+ return next_send_time;
+ }
+
+ void VerifyOutput(size_t num_samples) {
+ for (size_t i = 0; i < num_samples; ++i) {
+ ASSERT_EQ(output_[i], output_external_[i]) <<
+ "Diff in sample " << i << ".";
+ }
+ }
+
+ virtual int GetArrivalTime(int send_time) {
+ int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
+ last_send_time_ = send_time;
+ last_arrival_time_ = arrival_time;
+ return arrival_time;
+ }
+
+ virtual bool Lost() { return false; }
+
+ void RunTest(int num_loops) {
+ // Get next input packets (mono and multi-channel).
+ int next_send_time;
+ int next_arrival_time;
+ do {
+ next_send_time = GetNewPackets();
+ ASSERT_NE(-1, next_send_time);
+ next_arrival_time = GetArrivalTime(next_send_time);
+ } while (Lost()); // If lost, immediately read the next packet.
+
+ EXPECT_CALL(*external_decoder_, Decode(_, payload_size_bytes_, _, _))
+ .Times(num_loops);
+
+ int time_now = 0;
+ for (int k = 0; k < num_loops; ++k) {
+ while (time_now >= next_arrival_time) {
+ // Insert packet in regular instance.
+ ASSERT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header_, encoded_,
+ payload_size_bytes_,
+ next_arrival_time));
+ // Insert packet in external decoder instance.
+ EXPECT_CALL(*external_decoder_,
+ IncomingPacket(_, payload_size_bytes_,
+ rtp_header_.header.sequenceNumber,
+ rtp_header_.header.timestamp,
+ next_arrival_time));
+ ASSERT_EQ(NetEq::kOK,
+ neteq_external_->InsertPacket(rtp_header_, encoded_,
+ payload_size_bytes_,
+ next_arrival_time));
+ // Get next input packet.
+ do {
+ next_send_time = GetNewPackets();
+ ASSERT_NE(-1, next_send_time);
+ next_arrival_time = GetArrivalTime(next_send_time);
+ } while (Lost()); // If lost, immediately read the next packet.
+ }
+ NetEqOutputType output_type;
+ // Get audio from regular instance.
+ int samples_per_channel;
+ int num_channels;
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->GetAudio(kMaxBlockSize, output_,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(output_size_samples_, samples_per_channel);
+ // Get audio from external decoder instance.
+ ASSERT_EQ(NetEq::kOK,
+ neteq_external_->GetAudio(kMaxBlockSize, output_external_,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ EXPECT_EQ(1, num_channels);
+ EXPECT_EQ(output_size_samples_, samples_per_channel);
+ std::ostringstream ss;
+ ss << "Lap number " << k << ".";
+ SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
+ // Compare mono and multi-channel.
+ ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
+
+ time_now += kTimeStepMs;
+ }
+ }
+
+ const int sample_rate_hz_;
+ const int samples_per_ms_;
+ const int frame_size_ms_;
+ const int frame_size_samples_;
+ const int output_size_samples_;
+ NetEq* neteq_external_;
+ NetEq* neteq_;
+ MockExternalPcm16B* external_decoder_;
+ test::RtpGenerator rtp_generator_;
+ int16_t* input_;
+ uint8_t* encoded_;
+ int16_t output_[kMaxBlockSize];
+ int16_t output_external_[kMaxBlockSize];
+ WebRtcRTPHeader rtp_header_;
+ int payload_size_bytes_;
+ int last_send_time_;
+ int last_arrival_time_;
+ scoped_ptr<test::InputAudioFile> input_file_;
+};
+
+TEST_F(NetEqExternalDecoderTest, RunTest) {
+ RunTest(100); // Run 100 laps @ 10 ms each in the test loop.
+}
+
+} // namespace webrtc