Make audio example up-to-date

BUG=none
TEST=none

Review URL: http://codereview.chromium.org/6135007

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@71095 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 31 insertions, 18 deletions

diff --git a/ppapi/examples/audio/audio.cc b/ppapi/examples/audio/audio.cc
index fcef2b1..a0ba067 100644
--- a/ppapi/examples/audio/audio.cc
+++ b/ppapi/examples/audio/audio.cc
@@ -12,7 +12,7 @@
 
 // Separate left and right frequency to make sure we didn't swap L & R.
 // Sounds pretty horrible, though...
-const double frequency_l = 200;
+const double frequency_l = 400;
 const double frequency_r = 1000;
 
 // This sample frequency is guaranteed to work.
@@ -20,46 +20,59 @@ const PP_AudioSampleRate_Dev sample_frequency = PP_AUDIOSAMPLERATE_44100;
 const uint32_t sample_count = 4096;
 uint32_t obtained_sample_count = 0;
 
+const double kPi = 3.141592653589;
+const double kTwoPi = 2.0 * kPi;
+
 class MyInstance : public pp::Instance {
  public:
   explicit MyInstance(PP_Instance instance)
       : pp::Instance(instance),
-        audio_time_(0) {
+        audio_wave_l_(0.0),
+        audio_wave_r_(0.0) {
   }
 
   virtual bool Init(uint32_t argc, const char* argn[], const char* argv[]) {
     pp::AudioConfig_Dev config;
     obtained_sample_count = pp::AudioConfig_Dev::RecommendSampleFrameCount(
         sample_count);
-    config = pp::AudioConfig_Dev(sample_frequency, obtained_sample_count);
-    audio_ = pp::Audio_Dev(*this, config, SineWaveCallback, this);
+    config = pp::AudioConfig_Dev(this, sample_frequency, obtained_sample_count);
+    audio_ = pp::Audio_Dev(this, config, SineWaveCallback, this);
     return audio_.StartPlayback();
   }
 
  private:
   static void SineWaveCallback(void* samples, size_t num_bytes, void* thiz) {
-    const double th_l = 2 * 3.141592653589 * frequency_l / sample_frequency;
-    const double th_r = 2 * 3.141592653589 * frequency_r / sample_frequency;
-
-    // Store time value to avoid clicks on buffer boundries.
-    size_t t = reinterpret_cast<MyInstance*>(thiz)->audio_time_;
+    const double delta_l = kTwoPi * frequency_l / sample_frequency;
+    const double delta_r = kTwoPi * frequency_r / sample_frequency;
 
-    uint16_t* buf = reinterpret_cast<uint16_t*>(samples);
+    // 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_;
+    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) {
-      *buf++ = static_cast<uint16_t>(sin(th_l * t)
-          * std::numeric_limits<uint16_t>::max());
-      *buf++ = static_cast<uint16_t>(sin(th_r * t++)
-          * std::numeric_limits<uint16_t>::max());
+      *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.
+      wave_l += delta_l;
+      if (wave_l > kTwoPi)
+        wave_l -= kTwoPi * 2.0;
+      wave_r += delta_r;
+      if (wave_r > kTwoPi)
+        wave_r -= kTwoPi * 2.0;
     }
-    reinterpret_cast<MyInstance*>(thiz)->audio_time_ = t;
+    // 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 resource. Allocated in Init(), freed on destruction.
   pp::Audio_Dev audio_;
 
-  // Audio buffer time. Used to make prevent sine wave skips on buffer
-  // boundaries.
-  size_t audio_time_;
+  // Current audio wave position, used to prevent sine wave skips
+  // on buffer boundaries.
+  double audio_wave_l_;
+  double audio_wave_r_;
 };
 
 class MyModule : public pp::Module {