Port Pepper Audio API from the devsite.

native_client_sdk/src/doc/devguide/coding/audio.rst

 .. _devguide-coding-audio:
 
 #####
 Audio
 #####
 
-fooooooooooooooooooooo
-
+.. contents::
+  :local:
+  :backlinks: none
+  :depth: 2
+
+This chapter describes how to use the Pepper audio API to play an audio
+stream. The Pepper audio API provides a low-level means of playing a stream of
+audio samples generated by a Native Client module. The API generally works as
+follows: A Native Client module creates an audio resource that represents an
+audio stream, and tells the browser to start or stop playing the audio
+resource. The browser calls a function in the Native Client module to fill a
+buffer with audio samples every time it needs data to play from the audio
+stream.
+
+The code examples in this chapter describe a simple Native Client module that
+generates audio samples using a sine wave with a frequency of 440 Hz. The module
+starts playing the audio samples as soon as it is loaded into the browser. For a
+slightly more sophisticated example, see the ``audio`` example (source code in
+the SDK directory ``examples/api/audio``), which lets users specify a frequency
+for the sine wave and click buttons to start and stop audio playback.
+
+Reference information
+=====================
+
+For reference information related to the Pepper audio API, see the following
+documentation:
+
+* `pp::AudioConfig class
+  <https://developers.google.com/native-client/peppercpp/classpp_1_1_audio_config>`_
+
+* `pp::Audio class
+  <https://developers.google.com/native-client/peppercpp/classpp_1_1_audio>`_
+
+* `audio_config.h
+  <https://developers.google.com/native-client/peppercpp/audio__config_8h>`_
+
+* `audio.h <https://developers.google.com/native-client/peppercpp/audio_8h>`_
+
+* `PP_AudioSampleRate
+  <https://developers.google.com/native-client/pepperc/group___enums.html#gaee750c350655f2fb0fe04c04029e0ff8>`_
+
+About the Pepper audio API
+==========================
+
+The Pepper audio API lets Native Client modules play audio streams in a
+browser. To play an audio stream, a module generates audio samples and writes
+them into a buffer. The browser reads the audio samples from the buffer and
+plays them using an audio device on the client computer.
+
+.. image:: /images/pepper-audio-buffer.png
+
+This mechanism is simple but low-level. If you want to play plain sound files in
+a web application, you may want to consider higher-level alternatives such as
+using the HTML ``<audio>`` tag, JavaScript, or the new `Web Audio API
+<http://chromium.googlecode.com/svn/trunk/samples/audio/index.html>`_.

[binji, 2013/10/02 07:00:42]

We also include ogg/vorbis/openAL in the SDK now, but we don't really have an
example of it. And there are other audio libraries in naclports. Worth
referencing?

[Jim Stichnoth, 2013/10/17 21:57:36]

I don't really have enough knowledge to add something accurate and intelligent
here, but I'll be happy to paste in some text. :)

[binji, 2013/10/17 21:59:54]

OK, we can leave it out for now.

+
+The Pepper audio API is a good option for playing audio data if you want to do
+audio processing in your web application. You might use the audio API, for
+example, if you want to apply audio effects to sounds, synthesize your own
+sounds, or do any other type of CPU-intensive processing of audio
+samples. Another likely use case is gaming applications: you might use a gaming
+library to process audio data, and then simply use the audio API to output the
+processed data.
+
+The Pepper audio API is straightforward to use:
+
+#. Your module creates an audio configuration resource and an audio resource.
+
+#. Your module implements a callback function that fills an audio buffer with
+   data.
+
+#. Your module invokes the StartPlayback and StopPlayback methods of the audio
+   resource (e.g., when certain events occur).
+
+#. The browser invokes your callback function whenever it needs audio data to
+   play. Your callback function can generate the audio data in a number of
+   ways---e.g., it can generate new data, or it can copy pre-mixed data into the
+   audio buffer.
+
+This basic interaction is illustrated below, and described in detail in the
+sections that follow.
+
+.. image:: /images/pepper-audio-api.png
+
+Digital audio concepts
+======================
+
+Before you use the Pepper audio API, it's helpful to understand a few concepts
+that are fundamental to how digital audio is recorded and played back:
+
+sample rate
+  the number of times an input sound source is sampled per second;
+  correspondingly, the number of samples that are played back per second
+
+bit depth
+  the number of bits used to represent a sample
+
+channels
+  the number of input sources recorded in each sampling interval;
+  correspondingly, the number of outputs that are played back simultaneously
+  (typically using different speakers)
+
+The higher the sample rate and bit depth used to record a sound wave, the more
+accurately the sound wave can be reproduced, since it will have been sampled
+more frequently and stored using a higher level of quantization. Common sampling
+rates include 44,100 Hz (44,100 samples/second, the sample rate used on CDs),
+and 48,000 Hz (the sample rate used on DVDs and Digital Audio Tapes). A common
+bit depth is 16 bits per sample, and a common number of channels is 2 (left and
+right channels for stereo sound).
+
+The Pepper audio API currently lets Native Client modules play audio streams
+with the following configurations:
+
+* **sample rate**: 44,100 Hz or 48,000 Hz
+* **bit depth**: 16
+* **channels**: 2 (stereo)
+
+Setting up the module
+=====================
+
+The code examples below describe a simple Native Client module that generates
+audio samples using a sine wave with a frequency of 440 Hz. The module starts
+playing the audio samples as soon as it is loaded into the browser.
+
+The Native Client module is set up by implementing subclasses of the
+``pp::Module`` and ``pp::Instance`` classes, as normal.

[binji, 2013/10/02 07:00:42]

None of the other documents (aside from the tutorial) show how to set up the
instance/module, do they? I think this may be overkill.

Maybe we should remove this, or just keep the SineWaveCallback and pp::Audio,
which is different than normal.

[Jim Stichnoth, 2013/10/17 21:57:36]

I trimmed the example back a bunch.

+
+.. naclcode::
+
+  #include <cassert>
+  #include <cmath>
+  #include <limits>
+  #include "ppapi/cpp/audio.h"
+  #include "ppapi/cpp/instance.h"
+  #include "ppapi/cpp/module.h"
+
+  namespace {
+  // Constants used to generate a sine wave.
+  const double kFrequency = 440.0;
+  const double kTwoPi = 2.0 * 3.141592653589;
+
+  // Constants used to create an audio configuration resource.
+  // The sample count we will request from the browser.
+  const uint32_t kSampleFrameCount = 4096u;
+  // The number of channels in the audio stream (only supporting stereo audio
+  // for now).
+  const uint32_t kChannels = 2u;
+  }  // namespace
+
+  namespace sine_synth {
+  // The Instance class.  One of these exists for each instance of your NaCl
+  // module on the web page.  The browser will ask the Module object to create
+  // a new Instance for each occurrence of the <embed> tag that has these
+  // attributes:
+  //     type="application/x-nacl"
+  //     src="sine_synth.nmf"
+  class SineSynthInstance : public pp::Instance {
+   public:
+    explicit SineSynthInstance(PP_Instance instance)
+        : pp::Instance(instance),
+          theta_(0),
+          sample_frame_count_(kSampleFrameCount) {}
+    virtual ~SineSynthInstance() {}
+
+    // Called by the browser once the NaCl module is loaded and ready to
+    // initialize.  Creates a Pepper audio context and initializes it. Returns
+    // true on success.  Returning false causes the NaCl module to be deleted
+    // and no other functions to be called.
+    virtual bool Init(uint32_t argc, const char* argn[], const char* argv[]);
+
+   private:
+    // Function called by the browser when it needs more audio samples.
+    static void SineWaveCallback(void* samples,
+                                 uint32_t buffer_size,
+                                 void* data) {
+    ...
+    }
+
+    // Audio resource.
+    pp::Audio audio_;
+
+    // The last parameter sent to the sin function.  Used to prevent sine wave
+    // skips on buffer boundaries.
+    double theta_;
+
+    // The count of sample frames per channel in an audio buffer.
+    uint32_t sample_frame_count_;
+
+  };
+
+  // The Module class.  The browser calls the CreateInstance() method to create
+  // an instance of your NaCl module on the web page.  The browser creates a new
+  // instance for each <embed> tag with type="application/x-nacl".
+  class SineSynthModule : public pp::Module {
+   public:
+    SineSynthModule() : pp::Module() {}
+    ~SineSynthModule() {}
+
+    // Create and return a SineSynthInstance object.
+    virtual pp::Instance* CreateInstance(PP_Instance instance) {
+      return new SineSynthInstance(instance);
+    }
+  };
+
+  }  // namespace sine_synth
+
+  // Factory function called by the browser when the module is first loaded.
+  // The browser keeps a singleton of this module.  It calls the
+  // CreateInstance() method on the object you return to make instances.  There
+  // is one instance per <embed> tag on the page.  This is the main binding
+  // point for your NaCl module with the browser.
+  namespace pp {
+  Module* CreateModule() {
+    return new sine_synth::SineSynthModule();
+  }
+
+Creating an audio configuration resource
+========================================
+
+Resources
+---------
+
+Before the module can play an audio stream, it must create two resources: an
+audio configuration resource and an audio resource. Resources are handles to
+objects that the browser provides to module instances. An audio resource is an
+object that represents the state of an audio stream, including whether the
+stream is paused or being played back, and which callback function to invoke
+when the samples in the stream's buffer run out. An audio configuration resource
+is an object that stores configuration data for an audio resource, including the
+sampling frequency of the audio samples, and the number of samples that the
+callback function must provide when the browser invokes it.
+
+Sample frame count
+------------------
+
+Prior to creating an audio configuration resource, the module should call
+``RecommendSampleFrameCount`` to obtain a *sample frame count* from the
+browser. The sample frame count is the number of samples that the callback
+function must provide per channel each time the browser invokes the callback
+function. For example, if the sample frame count is 4096 for a stereo audio
+stream, the callback function must provide a 8192 samples (4096 for the left
+channel and 4096 for the right channel).
+
+The module can request a specific sample frame count, but the browser may return
+a different sample frame count depending on the capabilities of the client
+device. At present, ``RecommendSampleFrameCount`` simply bound-checks the
+requested sample frame count (see
+``toolchain/platform/x86_64-nacl/include/ppapi/c/ppb_audio_config.h`` for the

[binji, 2013/10/02 07:00:42]

this location is ``include/ppapi/c/ppb_audio_config.h`` now

[Jim Stichnoth, 2013/10/17 21:57:36]

Done.

+minimum and maximum sample frame counts, currently 64 and 32768). In the future,
+``RecommendSampleFrameCount`` may perform a more sophisticated calculation,
+particularly if there is an intrinsic buffer size for the client device.
+
+Selecting a sample frame count for an audio stream involves a tradeoff between
+latency and CPU usage. If you want your module to have short audio latency so
+that it can rapidly change what's playing in the audio stream, you should
+request a small sample frame count. That could be useful in gaming applications,
+for example, where sounds have to change frequently in response to game
+action. However, a small sample frame count results in higher CPU usage, since
+the browser must invoke the callback function frequently to refill the audio
+buffer. Conversely, a large sample frame count results in higher latency but
+lower CPU usage. You should request a large sample frame count if your module
+will play long, uninterrupted audio segments.
+
+Supported audio configurations
+------------------------------
+
+After the module obtains a sample frame count, it can create an audio
+configuration resource. Currently the Pepper audio API supports audio streams
+with the following configuration settings:
+
+* sample rate: 44,100 Hz or 48,000 Hz

[binji, 2013/10/02 07:00:42]

There is a duplicated table above. Should we remove one copy?

[Jim Stichnoth, 2013/10/17 21:57:36]

Done.

+* bit depth: 16
+* channels: 2
+
+C++ modules can create a configuration resource by instantiating a
+``pp::AudioConfig`` object. Check ``audio_config.h`` for the latest
+configurations that are supported.
+
+.. naclcode::
+
+  bool SineSynthInstance::Init(uint32_t argc,

[binji, 2013/10/02 07:00:42]

This has changed slightly, but maybe it's better to keep this version because it
is clearer?
https://code.google.com/p/chromium/codesearch#chromium/src/native_client_sdk/...

+                               const char* argn[],
+                               const char* argv[]) {
+
+    // Ask the browser/device for an appropriate sample frame count size.
+    sample_frame_count_ =
+        pp::AudioConfig::RecommendSampleFrameCount(PP_AUDIOSAMPLERATE_44100,
+                                                   kSampleFrameCount);
+
+    // Create an audio configuration resource.
+    pp::AudioConfig audio_config = pp::AudioConfig(this,
+                                                   PP_AUDIOSAMPLERATE_44100,
+                                                   sample_frame_count_);
+
+    // Create an audio resource.
+    audio_ = pp::Audio(this,
+                       audio_config,
+                       SineWaveCallback,
+                       this);
+
+    // Start playback when the module instance is initialized.
+    return audio_.StartPlayback();
+  }
+
+Creating an audio resource
+==========================
+
+Once the module has created an audio configuration resource, it can create an
+audio resource. To do so, it instantiates a ``pp::Audio`` object, passing in a
+pointer to the module instance, the audio configuration resource, a callback
+function, and a pointer to user data (data that is used in the callback
+function).
+
+.. naclcode::

[binji, 2013/10/02 07:00:42]

This is a duplicate of the above code. Remove one?

[Jim Stichnoth, 2013/10/17 21:57:36]

Done.

+
+  bool SineSynthInstance::Init(uint32_t argc,
+                               const char* argn[],
+                               const char* argv[]) {
+
+    // Ask the browser/device for an appropriate sample frame count size.
+    sample_frame_count_ =
+        pp::AudioConfig::RecommendSampleFrameCount(PP_AUDIOSAMPLERATE_44100,
+                                                   kSampleFrameCount);
+
+    // Create an audio configuration resource.
+    pp::AudioConfig audio_config = pp::AudioConfig(this,
+                                       PP_AUDIOSAMPLERATE_44100,
+                                       sample_frame_count_);
+
+    // Create an audio resource.
+    audio_ = pp::Audio(this,
+                       audio_config,
+                       SineWaveCallback,
+                       this);
+
+    // Start playback when the module instance is initialized.
+    return audio_.StartPlayback();
+  }
+
+Implementing a callback function
+================================
+
+The browser calls the callback function associated with an audio resource every
+time it needs more samples to play. The callback function can generate new
+samples (e.g., by applying sound effects), or copy pre-mixed samples into the
+audio buffer. The example below generates new samples by computing values of a
+sine wave.
+
+The last parameter passed to the callback function is generic user data that the
+function can use in processing samples. In the example below, the user data is a
+pointer to the module instance, which includes member variables
+``sample_frame_count_`` (the sample frame count obtained from the browser) and
+``theta_`` (the last angle that was used to compute a sine value in the previous
+callback; this lets the function generate a smooth sine wave by starting at that
+angle plus a small delta).
+
+.. naclcode::
+
+  class SineSynthInstance : public pp::Instance {
+   public:
+    ...
+
+   private:
+    static void SineWaveCallback(void* samples,
+                                 uint32_t buffer_size,
+                                 void* data) {
+
+      // The user data in this example is a pointer to the module instance.
+      SineSynthInstance* sine_synth_instance =
+          reinterpret_cast<SineSynthInstance*>(data);

[binji, 2013/10/02 07:00:42]

hmm, these reinterpret_casts should really be static_casts. But they are
reinterpret_casts in the code as well. :-/

+
+      // Delta by which to increase theta_ for each sample.
+      const double delta = kTwoPi * kFrequency / PP_AUDIOSAMPLERATE_44100;
+      // Amount by which to scale up the computed sine value.
+      const int16_t max_int16 = std::numeric_limits<int16_t>::max();
+
+      int16_t* buff = reinterpret_cast<int16_t*>(samples);
+
+      // Make sure we can't write outside the buffer.
+      assert(buffer_size >= (sizeof(*buff) * kChannels *
+                             sine_synth_instance->sample_frame_count_));
+
+      for (size_t sample_i = 0;
+           sample_i < sine_synth_instance->sample_frame_count_;
+           ++sample_i, sine_synth_instance->theta_ += delta) {
+
+        // Keep theta_ from going beyond 2*Pi.
+        if (sine_synth_instance->theta_ > kTwoPi) {
+          sine_synth_instance->theta_ -= kTwoPi;
+        }
+
+        // Compute the sine value for the current theta_, scale it up,
+        // and write it into the buffer once for each channel.
+        double sin_value(std::sin(sine_synth_instance->theta_));
+        int16_t scaled_value = static_cast<int16_t>(sin_value * max_int16);
+        for (size_t channel = 0; channel < kChannels; ++channel) {
+          *buff++ = scaled_value;
+        }

[binji, 2013/10/02 07:00:42]

This code is basically the same as the example. I thought it might be too
complex, but it it's probably OK.

+      }
+    }
+
+    ...
+  };
+
+Application threads and real-time requirements
+----------------------------------------------
+
+The callback function runs in a background application thread. This allows audio
+processing to continue even when the application is busy doing something
+else. If the main application thread and the callback thread access the same
+data, you may be tempted to use a lock to control access to that data. You
+should avoid the use of locks in the callback thread, however, as attempting to
+acquire a lock may cause the thread to get swapped out, resulting in audio
+dropouts.
+
+In general, you must program the callback thread carefully, as the Pepper audio
+API is a very low level API that needs to meet hard real-time requirements. If
+the callback thread spends too much time processing, it can easily miss the
+real-time deadline, resulting in audio dropouts. One way the callback thread can
+miss the deadline is by taking too much time doing computation. Another way the
+callback thread can miss the deadline is by executing a function call that swaps
+out the callback thread. Unfortunately, such function calls include just about
+all C Run-Time (CRT) library calls and Pepper API calls. The callback thread
+should therefore avoid calls to malloc, gettimeofday, mutex, condvars, critical
+sections, and so forth; any such calls could attempt to take a lock and swap out
+the callback thread, which would be disastrous for audio playback. Similarly,
+the callback thread should avoid Pepper API calls. Audio dropouts due to thread
+swapping can be very rare and very hard to track down and debug---it's best to
+avoid making system/Pepper calls in the first place. In short, the audio
+(callback) thread should use "lock-free" techniques and avoid making CRT library
+calls.
+
+One other issue to be aware of is that the ``StartPlayback`` function (discussed
+below) is an asynchronous RPC; i.e., it does not block. That means that the
+callback function may not be called immediately after the call to
+``StartPlayback``. If it's important to synchronize the callback thread with
+another thread so that the audio stream starts playing simultaneously with
+another action in your application, you must handle such synchronization
+manually.
+
+Starting and stopping playback
+==============================
+
+To start and stop audio playback, the module simply calls the ``StartPlayback``
+and ``StopPlayback`` methods of the ``Audio`` object. In the example below, the
+module calls ``StartPlayback`` when the module instance is created; a more

[binji, 2013/10/02 07:00:42]

This is no longer true, the StartPlayback/StopPlayback calls happen as a result
of JavaScript messages:
https://code.google.com/p/chromium/codesearch#chromium/src/native_client_sdk/...

[Jim Stichnoth, 2013/10/17 21:57:36]

Done.

+realistic use would be for a module to start playback in response to user input.
+
+.. naclcode::
+
+  bool SineSynthInstance::Init(uint32_t argc,
+                               const char* argn[],
+                               const char* argv[]) {
+
+    // Ask the browser/device for an appropriate sample frame count size.
+    sample_frame_count_ =
+        pp::AudioConfig::RecommendSampleFrameCount(PP_AUDIOSAMPLERATE_44100,
+                                                   kSampleFrameCount);
+
+    // Create an audio configuration resource.
+    pp::AudioConfig audio_config = pp::AudioConfig(this,
+                                       PP_AUDIOSAMPLERATE_44100,
+                                       sample_frame_count_);
+
+    // Create an audio resource.
+    audio_ = pp::Audio(this,
+                       audio_config,
+                       SineWaveCallback,
+                       this);
+
+    // Start playback when the module instance is initialized.
+    return audio_.StartPlayback();
+  }