Adding decoded_fec_rate to NetEQ Network Statistics.

A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data.

BUG=3867
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34969004

Cr-Commit-Position: refs/heads/master@{#8384}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc

@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqExternalDecoderTest::NetEqExternalDecoderTest(NetEqDecoder codec,
+                                                   AudioDecoder* decoder)
+    : codec_(codec),
+      decoder_(decoder),
+      sample_rate_hz_(CodecSampleRateHz(codec_)),
+      channels_(static_cast<int>(decoder_->channels())) {
+  NetEq::Config config;
+  config.sample_rate_hz = sample_rate_hz_;
+  neteq_.reset(NetEq::Create(config));
+  printf("%d\n", channels_);
+}
+
+void NetEqExternalDecoderTest::Init() {
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->RegisterExternalDecoder(decoder_, codec_, kPayloadType));
+}
+
+void NetEqExternalDecoderTest::InsertPacket(WebRtcRTPHeader rtp_header,
+                                            const uint8_t* payload,
+                                            size_t payload_size_bytes,
+                                            uint32_t receive_timestamp) {
+  ASSERT_EQ(
+      NetEq::kOK,
+      neteq_->InsertPacket(
+          rtp_header, payload, payload_size_bytes, receive_timestamp));
+}
+
+int NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
+                                             int16_t* output,
+                                             NetEqOutputType* output_type) {
+  // Get audio from regular instance.
+  int samples_per_channel;
+  int num_channels;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(max_length,
+                             output,
+                             &samples_per_channel,
+                             &num_channels,
+                             output_type));
+  EXPECT_EQ(channels_, num_channels);
+  EXPECT_EQ(kOutputLengthMs * sample_rate_hz_ / 1000, samples_per_channel);
+  return samples_per_channel;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+namespace test {
+// This test class provides a way run NetEQ with an external decoder.
+class NetEqExternalDecoderTest {
+ protected:
+  static const uint8_t kPayloadType = 95;
+  static const int kOutputLengthMs = 10;
+
+  // The external decoder |decoder| is suppose to be of type |codec|.
+  NetEqExternalDecoderTest(NetEqDecoder codec, AudioDecoder* decoder);
+
+  virtual ~NetEqExternalDecoderTest() { }
+
+  // In Init(), we register the external decoder.
+  void Init();
+
+  // Inserts a new packet with |rtp_header| and |payload| of
+  // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
+  // of the time when the packet was received, and should be measured with
+  // the same tick rate as the RTP timestamp of the current payload.
+  virtual void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
+                            size_t payload_size_bytes,
+                            uint32_t receive_timestamp);
+
+  // Get 10 ms of audio data. The data is written to |output|, which can hold
+  // (at least) |max_length| elements. Returns number of samples.
+  int GetOutputAudio(size_t max_length, int16_t* output,
+                     NetEqOutputType* output_type);
+
+  NetEq* neteq() { return neteq_.get(); }
+
+ private:
+  NetEqDecoder codec_;
+  AudioDecoder* decoder_;
+  int sample_rate_hz_;
+  int channels_;
+  scoped_ptr<NetEq> neteq_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_