Dedicated speed test for NetEq3

This is the same test as was aleready implemented for NetEq3 in r4763.

BUG=1363
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4782 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/neteq.gypi

@@ -157,6 +157,20 @@
           ],
         },
 
+        {
+          'target_name': 'neteq3_speed_test',
+          'type': 'executable',
+          'dependencies': [
+            'NetEq',
+            'PCM16B',
+            'neteq_unittest_tools',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+            'test/neteq_speed_test.cc',
+          ],
+        },
+
         {
          'target_name': 'NetEqTestTools',
           # Collection of useful functions used in other tests

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

@@ -0,0 +1,233 @@
+/*
+ *  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/neteq/interface/webrtc_neteq.h"
+#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
+#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.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::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 WebRtcNetEQDecoder kDecoderType = 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.
+  int error;
+  int inst_size_bytes;
+  error = WebRtcNetEQ_AssignSize(&inst_size_bytes);
+  if (error) {
+    std::cerr << "Error returned from WebRtcNetEQ_AssignSize." << std::endl;
+    exit(1);
+  }
+  char* inst_mem = new char[inst_size_bytes];
+  void* neteq_inst;
+  error = WebRtcNetEQ_Assign(&neteq_inst, inst_mem);
+  if (error) {
+    std::cerr << "Error returned from WebRtcNetEQ_Assign." << std::endl;
+    exit(1);
+  }
+  // Select decoders.
+  WebRtcNetEQDecoder decoder_list[] = {kDecoderType};
+  int max_number_of_packets;
+  int buffer_size_bytes;
+  int overhead_bytes_dummy;
+  error = WebRtcNetEQ_GetRecommendedBufferSize(
+      neteq_inst, decoder_list, sizeof(decoder_list) / sizeof(decoder_list[1]),
+      kTCPLargeJitter, &max_number_of_packets, &buffer_size_bytes,
+      &overhead_bytes_dummy);
+  if (error) {
+    std::cerr << "Error returned from WebRtcNetEQ_GetRecommendedBufferSize."
+              << std::endl;
+    exit(1);
+  }
+  char* buffer_mem = new char[buffer_size_bytes];
+  error = WebRtcNetEQ_AssignBuffer(neteq_inst, max_number_of_packets,
+                                   buffer_mem, buffer_size_bytes);
+  if (error) {
+      std::cerr << "Error returned from WebRtcNetEQ_AssignBuffer." << std::endl;
+      exit(1);
+    }
+  error = WebRtcNetEQ_Init(neteq_inst, kSampRateHz);
+  if (error) {
+      std::cerr << "Error returned from WebRtcNetEQ_Init." << std::endl;
+      exit(1);
+    }
+
+  // Register decoder.
+  WebRtcNetEQ_CodecDef codec_definition;
+  SET_CODEC_PAR(codec_definition, kDecoderType, kPayloadType, NULL,
+                kSampRateHz);
+  SET_PCM16B_SWB32_FUNCTIONS(codec_definition);
+  error = WebRtcNetEQ_CodecDbAdd(neteq_inst, &codec_definition);
+  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) {
+        WebRtcNetEQ_RTPInfo rtp_info;
+        rtp_info.payloadType = rtp_header.header.payloadType;
+        rtp_info.sequenceNumber = rtp_header.header.sequenceNumber;
+        rtp_info.timeStamp = rtp_header.header.timestamp;
+        rtp_info.SSRC = rtp_header.header.ssrc;
+        rtp_info.markerBit = rtp_header.header.markerBit;
+        // Insert packet.
+        error = WebRtcNetEQ_RecInRTPStruct(
+            neteq_inst, &rtp_info, input_payload, payload_len,
+            packet_input_time_ms * kSampRateHz / 1000);
+        if (error != 0) {
+          std::cerr << "WebRtcNetEQ_RecInRTPStruct returned error code " <<
+              WebRtcNetEQ_GetErrorCode(neteq_inst) << 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];
+    int16_t samples_per_channel;
+    error = WebRtcNetEQ_RecOut(neteq_inst, out_data, &samples_per_channel);
+    if (error != 0) {
+      std::cerr << "WebRtcNetEQ_RecOut returned error code " <<
+          WebRtcNetEQ_GetErrorCode(neteq_inst) << 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 [] buffer_mem;
+  delete [] inst_mem;
+  return 0;
+}

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

@@ -104,7 +104,6 @@ int main(int argc, char* argv[]) {
     exit(1);
   }
 
-
   int32_t time_now_ms = 0;
 
   // Get first input packet.