Dedicated speed test for NetEq4

This CL implements a new speed test application for NetEq4.
The application runs a minimum of overhead in order to
highlight the performance of NetEq itself.

BUG=1363
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4763 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq4/neteq.gypi

@@ -179,6 +179,8 @@
             'tools',
           ],
           'sources': [
+            'tools/audio_loop.cc',
+            'tools/audio_loop.h',
             'tools/input_audio_file.cc',
             'tools/input_audio_file.h',
             'tools/rtp_generator.cc',

webrtc/modules/audio_coding/neteq4/neteq_tests.gypi

@@ -137,6 +137,20 @@
       ],
     },
 
+    {
+      'target_name': 'neteq4_speed_test',
+      'type': 'executable',
+      'dependencies': [
+        'NetEq4',
+        'neteq_unittest_tools',
+        'PCM16B',
+        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+      ],
+      'sources': [
+        'test/neteq_speed_test.cc',
+      ],
+    },
+
     {
      'target_name': 'NetEq4TestTools',
       # Collection of useful functions used in other tests.

webrtc/modules/audio_coding/neteq4/test/neteq_speed_test.cc

@@ -0,0 +1,185 @@
+/*
+ *  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.
+ */
+
+#include <stdio.h>
+
+#include <iostream>
+
+#include "gflags/gflags.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+using webrtc::NetEq;
+using webrtc::test::AudioLoop;
+using webrtc::test::RtpGenerator;
+using webrtc::WebRtcRTPHeader;
+
+// Flag validators.
+static bool ValidateRuntime(const char* flagname, int value) {
+  if (value > 0)  // Value is ok.
+    return true;
+  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  return false;
+}
+static bool ValidateLossrate(const char* flagname, int value) {
+  if (value >= 0)  // Value is ok.
+    return true;
+  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  return false;
+}
+static bool ValidateDriftfactor(const char* flagname, double value) {
+  if (value >= 0.0 && value < 1.0)  // Value is ok.
+    return true;
+  printf("Invalid value for --%s: %f\n", flagname, value);
+  return false;
+}
+
+// Define command line flags.
+DEFINE_int32(runtime_ms, 10000, "Simulated runtime in ms.");
+static const bool runtime_ms_dummy =
+    google::RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
+DEFINE_int32(lossrate, 10,
+             "Packet lossrate; drop every N packets.");
+static const bool lossrate_dummy =
+    google::RegisterFlagValidator(&FLAGS_lossrate, &ValidateLossrate);
+DEFINE_double(drift, 0.1,
+             "Clockdrift factor.");
+static const bool drift_dummy =
+    google::RegisterFlagValidator(&FLAGS_drift, &ValidateDriftfactor);
+
+int main(int argc, char* argv[]) {
+  static const int kMaxChannels = 1;
+  static const int kMaxSamplesPerMs = 48000 / 1000;
+  static const int kOutputBlockSizeMs = 10;
+  const std::string kInputFileName =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const int kSampRateHz = 32000;
+  const webrtc::NetEqDecoder kDecoderType = webrtc::kDecoderPCM16Bswb32kHz;
+  const int kPayloadType = 95;
+
+  std::string program_name = argv[0];
+  std::string usage = "Tool for measuring the speed of NetEq.\n"
+      "Usage: " + program_name + " [options]\n\n"
+      "  --runtime_ms=N         runtime in ms; default is 10000 ms\n"
+      "  --lossrate=N           drop every N packets; default is 10\n"
+      "  --drift=F              clockdrift factor between 0.0 and 1.0; "
+      "default is 0.1\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 1) {
+    // Print usage information.
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+
+  // Initialize NetEq instance.
+  NetEq* neteq = NetEq::Create(kSampRateHz);
+  // Register decoder in |neteq|.
+  int error;
+  error = neteq->RegisterPayloadType(kDecoderType, kPayloadType);
+  if (error) {
+    std::cerr << "Cannot register decoder." << std::endl;
+    exit(1);
+  }
+
+  // Set up AudioLoop object.
+  AudioLoop audio_loop;
+  const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
+  const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
+  if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                       kInputBlockSizeSamples)) {
+    std::cerr << "Cannot initialize AudioLoop object." << std::endl;
+    exit(1);
+  }
+
+
+  int32_t time_now_ms = 0;
+
+  // Get first input packet.
+  WebRtcRTPHeader rtp_header;
+  RtpGenerator rtp_gen(kSampRateHz / 1000);
+  // Start with positive drift first half of simulation.
+  double drift_factor = 0.1;
+  rtp_gen.set_drift_factor(drift_factor);
+  bool drift_flipped = false;
+  int32_t packet_input_time_ms =
+      rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
+  const int16_t* input_samples = audio_loop.GetNextBlock();
+  if (!input_samples) exit(1);
+  uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
+  int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+                                        kInputBlockSizeSamples,
+                                        input_payload);
+  assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+
+  // Main loop.
+  while (time_now_ms < FLAGS_runtime_ms) {
+    while (packet_input_time_ms <= time_now_ms) {
+      // Drop every N packets, where N = FLAGS_lossrate.
+      bool lost = false;
+      if (FLAGS_lossrate > 0) {
+        lost = ((rtp_header.header.sequenceNumber - 1) % FLAGS_lossrate) == 0;
+      }
+      if (!lost) {
+        // Insert packet.
+        int error = neteq->InsertPacket(
+            rtp_header, input_payload, payload_len,
+            packet_input_time_ms * kSampRateHz / 1000);
+        if (error != NetEq::kOK) {
+          std::cerr << "InsertPacket returned error code " <<
+              neteq->LastError() << std::endl;
+          exit(1);
+        }
+      }
+
+      // Get next packet.
+      packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
+                                                  kInputBlockSizeSamples,
+                                                  &rtp_header);
+      input_samples = audio_loop.GetNextBlock();
+      if (!input_samples) exit(1);
+      payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+                                        kInputBlockSizeSamples,
+                                        input_payload);
+      assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+    }
+
+    // Get output audio, but don't do anything with it.
+    static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
+        kMaxChannels;
+    int16_t out_data[kOutDataLen];
+    int num_channels;
+    int samples_per_channel;
+    int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
+                                &num_channels, NULL);
+    if (error != NetEq::kOK) {
+      std::cerr << "GetAudio returned error code " <<
+          neteq->LastError() << std::endl;
+      exit(1);
+    }
+    assert(samples_per_channel == kSampRateHz * 10 / 1000);
+
+    time_now_ms += kOutputBlockSizeMs;
+    if (time_now_ms >= FLAGS_runtime_ms / 2 && !drift_flipped) {
+      // Apply negative drift second half of simulation.
+      rtp_gen.set_drift_factor(-drift_factor);
+      drift_flipped = true;
+    }
+  }
+
+  std::cout << "Simulation done" << std::endl;
+  delete neteq;
+  return 0;
+}

webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc

@@ -0,0 +1,57 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <string.h>
+
+namespace webrtc {
+namespace test {
+
+bool AudioLoop::Init(const std::string file_name,
+                     size_t max_loop_length_samples,
+                     size_t block_length_samples) {
+  FILE* fp = fopen(file_name.c_str(), "rb");
+  if (!fp) return false;
+
+  audio_array_.reset(new int16_t[max_loop_length_samples +
+                                 block_length_samples]);
+  size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
+                              max_loop_length_samples, fp);
+  fclose(fp);
+
+  // Block length must be shorter than the loop length.
+  if (block_length_samples > samples_read) return false;
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&audio_array_[samples_read], audio_array_.get(),
+         block_length_samples * sizeof(int16_t));
+
+  loop_length_samples_ = samples_read;
+  block_length_samples_ = block_length_samples;
+  return true;
+}
+
+const int16_t* AudioLoop::GetNextBlock() {
+  // Check that the AudioLoop is initialized.
+  if (block_length_samples_ == 0) return NULL;
+
+  const int16_t* output_ptr = &audio_array_[next_index_];
+  next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
+  return output_ptr;
+}
+
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq4/tools/audio_loop.h

@@ -0,0 +1,60 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_
+
+#include <string>
+
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class serving as an infinite source of audio, realized by looping an audio
+// clip.
+class AudioLoop {
+ public:
+  AudioLoop()
+      : next_index_(0),
+        loop_length_samples_(0),
+        block_length_samples_(0),
+        audio_array_(NULL) {
+  }
+
+  virtual ~AudioLoop() {}
+
+  // Initializes the AudioLoop by reading from |file_name|. The loop will be no
+  // longer than |max_loop_length_samples|, if the length of the file is
+  // greater. Otherwise, the loop length is the same as the file length.
+  // The audio will be delivered in blocks of |block_length_samples|.
+  // Returns false if the initialization failed, otherwise true.
+  bool Init(const std::string file_name, size_t max_loop_length_samples,
+            size_t block_length_samples);
+
+  // Returns a pointer to the next block of audio. The number given as
+  // |block_length_samples| to the Init() function determines how many samples
+  // that can be safely read from the pointer.
+  const int16_t* GetNextBlock();
+
+ private:
+  size_t next_index_;
+  size_t loop_length_samples_;
+  size_t block_length_samples_;
+  scoped_array<int16_t> audio_array_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioLoop);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_AUDIO_LOOP_H_