Simple Jitter Test for AudioConverter.

media/audio/fake_audio_consumer_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include <cmath>
+
 #include "base/bind.h"
+#include "base/command_line.h"
 #include "base/message_loop.h"
+#include "base/rand_util.h"
+#include "base/run_loop.h"
+#include "base/string_number_conversions.h"
+#include "base/string_util.h"
+#include "base/stringprintf.h"
 #include "base/time.h"
 #include "media/audio/audio_buffers_state.h"
 #include "media/audio/fake_audio_consumer.h"
 #include "media/audio/simple_sources.h"
+#include "media/base/audio_converter.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 
 static const int kTestCallbacks = 5;
 
+// Switch to choose the runtime for the jitter test.
+static const char kJitterRunTime[] = "jitter-runtime";
+static const int kDefaultJitterRunTimeSeconds = 2;
+
+// Switch to choose the buffer size or to emulate an OS for output buffer size.
+// Valid values are <size_t>, WIN, MAC, LINUX.
+static const char kJitterOutputSize[] = "jitter-output-size";
+
+// Input buffer size.
+static const char kJitterInputSize[] = "jitter-input-size";
+static const int kDefaultJitterInputBufferSize = 2048;
+
 class FakeAudioConsumerTest : public testing::Test {
  public:
   FakeAudioConsumerTest()
       : params_(
             AudioParameters::AUDIO_FAKE, CHANNEL_LAYOUT_STEREO, 44100, 8, 128),
         fake_consumer_(message_loop_.message_loop_proxy(), params_),
         source_(params_.channels(), 200.0, params_.sample_rate()) {
     time_between_callbacks_ = base::TimeDelta::FromMicroseconds(
         params_.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
         static_cast<float>(params_.sample_rate()));
@@ skipping 105 matching lines @@
   // Issue a Stop() / Start() in between expected callbacks to maximize the
   // chance of catching the FakeAudioOutputStream doing the wrong thing.
   message_loop_.PostDelayedTask(FROM_HERE, base::Bind(
       &FakeAudioConsumerTest::StopStartOnAudioThread,
       base::Unretained(this)), time_between_callbacks_ / 2);
 
   // EndTest() will ensure the proper number of callbacks have occurred.
   message_loop_.Run();
 }
 
+class JitterTest : public testing::Test, public AudioConverter::InputCallback {
+ public:
+  enum BufferSizeChoice { kCustom, kMac, kWin, kLinux };
+
+  JitterTest() {
+    std::string output_size(
+        CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
+            kJitterOutputSize));
+    if (LowerCaseEqualsASCII(output_size, "win")) {
+      buffer_size_choice_ = kWin;
+    } else if (LowerCaseEqualsASCII(output_size, "linux")) {
+      buffer_size_choice_ = kLinux;
+    } else if (LowerCaseEqualsASCII(output_size, "mac")) {
+      buffer_size_choice_ = kMac;
+    } else if (base::StringToInt(output_size, &custom_output_buffer_size_)) {
+      buffer_size_choice_ = kCustom;
+    } else {
+      buffer_size_choice_ = kCustom;
+      custom_output_buffer_size_ = kDefaultJitterInputBufferSize;
+    }
+
+    std::string input_size(
+        CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
+            kJitterInputSize));
+    if (!base::StringToInt(input_size, &input_buffer_size_))
+      input_buffer_size_ = kDefaultJitterInputBufferSize;
+
+    std::string jitter_run_time(
+        CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
+            kJitterRunTime));
+    if (!base::StringToInt(jitter_run_time, &jitter_run_time_secs_))
+      jitter_run_time_secs_ = kDefaultJitterRunTimeSeconds;
+  }
+
+  virtual ~JitterTest() {}
+
+  base::TimeDelta ExpectedTimeBetweenCallbacks(const AudioParameters& params) {
+    return base::TimeDelta::FromMicroseconds(
+        params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
+        static_cast<float>(params.sample_rate()));
+  }
+
+  int ChooseOutputBufferSize(int output_sample_rate) {
+    switch (buffer_size_choice_) {
+      case kCustom:
+        return custom_output_buffer_size_;
+      case kMac: {
+        int buffer_size = 128;
+        if (output_sample_rate > 48000) {
+          // The default buffer size is too small for higher sample rates and
+          // may lead to glitching.  Adjust upwards by multiples of the default
+          // size.
+          if (output_sample_rate <= 96000)
+            buffer_size *= 2;
+          else if (output_sample_rate <= 192000)
+            buffer_size *= 4;
+        }
+        return buffer_size;
+      }
+      case kWin:
+        return output_sample_rate / 100;
+      case kLinux:
+        return 512;
+    }
+  }
+
+  int ChooseInputBufferSize(float input_sample_rate, int output_buffer_size,
+                            float output_sample_rate) {
+    return input_buffer_size_;
+  }
+
+  virtual double ProvideInput(AudioBus* audio_bus,
+                              base::TimeDelta buffer_delay) {
+    if (first_run_) {
+      first_run_ = false;
+      last_call_ = base::Time::Now();
+    } else {
+      base::Time now = base::Time::Now();
+      time_between_callbacks_.push_back(now - last_call_);
+      last_call_ = now;
+    }
+
+    if (base::Time::Now() - start_ >
+        base::TimeDelta::FromSeconds(jitter_run_time_secs_)) {
+      consumer_->Stop();
+      run_loop_->Quit();
+    }
+
+    return 0;
+  }
+
+ protected:
+  MessageLoop message_loop_;
+  base::Time last_call_;
+  base::Time start_;
+  std::vector<base::TimeDelta> time_between_callbacks_;
+  scoped_ptr<FakeAudioConsumer> consumer_;
+  scoped_ptr<base::RunLoop> run_loop_;
+  bool first_run_;
+  BufferSizeChoice buffer_size_choice_;
+  int input_buffer_size_;
+  int custom_output_buffer_size_;
+  int jitter_run_time_secs_;
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(JitterTest);
+};
+
+TEST_F(JitterTest, Jitter) {
+  static const int kInputRates[] =
+      { 44100, 48000 };
+  static const int kOutputRates[] =
+      { 16000, 44100, 48000, 96000, 192000 };
+
+  MessageLoop message_loop;
+  for (size_t i = 0; i < arraysize(kInputRates); ++i) {
+    for (size_t j = 0; j < arraysize(kOutputRates); ++j) {
+      const int output_buffer_size = ChooseOutputBufferSize(kOutputRates[j]);
+      const int input_buffer_size = ChooseInputBufferSize(
+          kInputRates[i], output_buffer_size, kOutputRates[j]);
+
+      const AudioParameters input_params(
+          AudioParameters::AUDIO_FAKE, CHANNEL_LAYOUT_STEREO,
+          kInputRates[i], 16, input_buffer_size);
+      const AudioParameters output_params(
+          AudioParameters::AUDIO_FAKE, CHANNEL_LAYOUT_STEREO,
+          kOutputRates[j], 16, output_buffer_size);
+
+      consumer_.reset(new FakeAudioConsumer(
+          message_loop.message_loop_proxy(), output_params));
+      AudioConverter converter(input_params, output_params, false);
+      converter.AddInput(this);
+      consumer_->Start(
+          base::Bind(&AudioConverter::Convert, base::Unretained(&converter)));
+      run_loop_.reset(new base::RunLoop());
+      start_ = base::Time::Now();
+      first_run_ = true;
+      run_loop_->Run();
+
+      base::TimeDelta min = base::TimeDelta::FromSeconds(10000);
+      base::TimeDelta max = base::TimeDelta::FromSeconds(-10000);
+      base::TimeDelta avg;
+      int bad = 0;
+      for (size_t k = 0; k < time_between_callbacks_.size(); ++k) {
+        base::TimeDelta sample = time_between_callbacks_[k];
+        if (sample < min)
+          min = sample;
+        else if (sample > max)
+          max = sample;
+        avg += sample;
+        if (sample < base::TimeDelta::FromMilliseconds(5))
+          bad++;
+      }
+      avg /= time_between_callbacks_.size();
+
+      printf(
+          "rate=%d -> %d, size=%d -> %d, %d callbacks, %d bad, min=%f, "
+          "max=%f, avg=%f, expected=%f\n",
+          input_params.sample_rate(), output_params.sample_rate(),
+          input_params.frames_per_buffer(), output_params.frames_per_buffer(),
+          static_cast<int>(time_between_callbacks_.size()), bad,
+          min.InMillisecondsF(), max.InMillisecondsF(), avg.InMillisecondsF(),
+          ExpectedTimeBetweenCallbacks(input_params).InMillisecondsF());
+
+      time_between_callbacks_.clear();
+
+    }
+  }
+}
+
 }  // namespace media