Pepper: Make the sample rate settable in the audio (output) example plugin.

ppapi/examples/audio/audio.cc

Index: ppapi/examples/audio/audio.cc
diff --git a/ppapi/examples/audio/audio.cc b/ppapi/examples/audio/audio.cc
index 91042fa11fb3fda3d0eb7fa961fbddeecd58102d..8b813b327ffb03b4c8f2148cb3d398e271f86651 100644
--- a/ppapi/examples/audio/audio.cc
+++ b/ppapi/examples/audio/audio.cc
@@ -2,7 +2,15 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include <cmath>
+// Needed on Windows to get |M_PI| from math.h on Windows.

[viettrungluu, 2011/11/30 02:26:31]

From the Department of Redundancy Department, obviously.

+#ifdef _WIN32
+#define _USE_MATH_DEFINES
+#endif
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
 #include <limits>
 
 #include "ppapi/c/pp_errors.h"
@@ -14,61 +22,79 @@
 
 // Separate left and right frequency to make sure we didn't swap L & R.
 // Sounds pretty horrible, though...
-const double frequency_l = 400;
-const double frequency_r = 1000;
+const double kLeftFrequency = 400;
+const double kRightFrequency = 1000;
 
 // This sample frequency is guaranteed to work.
-const PP_AudioSampleRate sample_frequency = PP_AUDIOSAMPLERATE_44100;
-const uint32_t sample_count = 4096;
-uint32_t obtained_sample_count = 0;
+const PP_AudioSampleRate kDefaultSampleRate = PP_AUDIOSAMPLERATE_44100;
+const uint32_t kDefaultSampleCount = 4096;
 
-const double kPi = 3.141592653589;
-const double kTwoPi = 2.0 * kPi;
+const char kSampleRateAttributeName[] = "samplerate";
 
 class MyInstance : public pp::Instance {
  public:
   explicit MyInstance(PP_Instance instance)
       : pp::Instance(instance),
+        sample_rate_(kDefaultSampleRate),
+        sample_count_(0),
         audio_wave_l_(0.0),
         audio_wave_r_(0.0) {
   }
 
   virtual bool Init(uint32_t argc, const char* argn[], const char* argv[]) {
+    for (uint32_t i = 0; i < argc; i++) {
+      if (strcmp(kSampleRateAttributeName, argn[i]) == 0) {
+        int value = atoi(argv[i]);
+        if (value > 0 && value <= 1000000)
+          sample_rate_ = static_cast<PP_AudioSampleRate>(value);
+        else
+          return false;
+      }
+    }
+
     pp::AudioConfig config;
-    obtained_sample_count = pp::AudioConfig::RecommendSampleFrameCount(
-        sample_frequency, sample_count);
-    config = pp::AudioConfig(this, sample_frequency, obtained_sample_count);
-    audio_ = pp::Audio(this, config, SineWaveCallback, this);
+    sample_count_ = pp::AudioConfig::RecommendSampleFrameCount(
+        sample_rate_, kDefaultSampleCount);
+    config = pp::AudioConfig(this, sample_rate_, sample_count_);
+    audio_ = pp::Audio(this, config, SineWaveCallbackTrampoline, this);
     return audio_.StartPlayback();
   }
 
  private:
-  static void SineWaveCallback(void* samples, uint32_t num_bytes, void* thiz) {
-    const double delta_l = kTwoPi * frequency_l / sample_frequency;
-    const double delta_r = kTwoPi * frequency_r / sample_frequency;
+  static void SineWaveCallbackTrampoline(void* samples,
+                                         uint32_t num_bytes,
+                                         void* thiz) {
+    static_cast<MyInstance*>(thiz)->SineWaveCallback(samples, num_bytes);
+  }
+
+  void SineWaveCallback(void* samples, uint32_t num_bytes) {
+    double delta_l = 2.0 * M_PI * kLeftFrequency / sample_rate_;
+    double delta_r = 2.0 * M_PI * kRightFrequency / sample_rate_;
 
     // Use per channel audio wave value to avoid clicks on buffer boundries.
-    double wave_l = reinterpret_cast<MyInstance*>(thiz)->audio_wave_l_;
-    double wave_r = reinterpret_cast<MyInstance*>(thiz)->audio_wave_r_;
+    double wave_l = audio_wave_l_;
+    double wave_r = audio_wave_r_;
     const int16_t max_int16 = std::numeric_limits<int16_t>::max();
     int16_t* buf = reinterpret_cast<int16_t*>(samples);
-    for (size_t sample = 0; sample < obtained_sample_count; ++sample) {
+    for (size_t sample = 0; sample < sample_count_; ++sample) {
       *buf++ = static_cast<int16_t>(sin(wave_l) * max_int16);
       *buf++ = static_cast<int16_t>(sin(wave_r) * max_int16);
-      // Add delta, keep within -kTwoPi..kTwoPi to preserve precision.
+      // Add delta, keep within -2 * M_PI .. 2 * M_PI to preserve precision.
       wave_l += delta_l;
-      if (wave_l > kTwoPi)
-        wave_l -= kTwoPi * 2.0;
+      if (wave_l > 2.0 * M_PI)
+        wave_l -= 2.0 * M_PI;
       wave_r += delta_r;
-      if (wave_r > kTwoPi)
-        wave_r -= kTwoPi * 2.0;
+      if (wave_r > 2.0 * M_PI)
+        wave_r -= 2.0 * M_PI;
     }
     // Store current value to use as starting point for next callback.
-    reinterpret_cast<MyInstance*>(thiz)->audio_wave_l_ = wave_l;
-    reinterpret_cast<MyInstance*>(thiz)->audio_wave_r_ = wave_r;
+    audio_wave_l_ = wave_l;
+    audio_wave_r_ = wave_r;
   }
 
-  // Audio resource. Allocated in Init(), freed on destruction.
+  PP_AudioSampleRate sample_rate_;
+  uint32_t sample_count_;
+
   pp::Audio audio_;
 
   // Current audio wave position, used to prevent sine wave skips