Chromium Code Reviews| Index: ppapi/examples/audio/audio.cc |
| =================================================================== |
| --- ppapi/examples/audio/audio.cc (revision 70800) |
| +++ ppapi/examples/audio/audio.cc (working copy) |
| @@ -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 +20,58 @@ |
| 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; |
| + const double delta_l = kTwoPi * frequency_l / sample_frequency; |
| + const double delta_r = kTwoPi * frequency_r / sample_frequency; |
| - // Store time value to avoid clicks on buffer boundries. |
| - size_t t = reinterpret_cast<MyInstance*>(thiz)->audio_time_; |
| - |
| - 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 |
|
brettw
2011/01/11 04:13:41
Style nit: need cap & period.
|
| + 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_; |
| + // Audio waveform. Used to prevent sine wave skips on buffer boundaries. |
|
brettw
2011/01/11 04:13:41
"Audio waveform" -> can you say something like "cu
|
| + double audio_wave_l_; |
| + double audio_wave_r_; |
| }; |
| class MyModule : public pp::Module { |