Same as 14820 but moved into media folder

- Low level audio interface for 'raw' formats
- Upcomming windows implementation


Review URL: http://codereview.chromium.org/15047

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@7254 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 135 insertions, 0 deletions

diff --git a/media/audio/audio_output.h b/media/audio/audio_output.h
new file mode 100644
index 0000000..3a6420e
--- /dev/null
+++ b/media/audio/audio_output.h
@@ -0,0 +1,135 @@
+// Copyright (c) 2006-2008 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.
+
+#ifndef MEDIA_AUDIO_AUDIO_OUTPUT_H_
+#define MEDIA_AUDIO_AUDIO_OUTPUT_H_
+
+// Low-level audio output support. To make sound there are 3 objects involved:
+// - AudioSource : produces audio samples on a pull model. Implements
+//   the AudioSourceCallback interface.
+// - AudioOutputStream : uses the AudioSource to render audio on a given
+//   channel, format and sample frequency configuration. Data from the
+//   AudioSource is delivered in a 'pull' model.
+// - AudioManager : factory for the AudioOutputStream objects, manager
+//   of the hardware resources and mixer control.
+//
+// The number and configuration of AudioOutputStream does not need to match the
+// physically available hardware resources. For example you can have:
+//
+//  MonoPCMSource1 --> MonoPCMStream1 --> |       | --> audio left channel
+//  StereoPCMSource -> StereoPCMStream -> | mixer |
+//  MonoPCMSource2 --> MonoPCMStream2 --> |       | --> audio right channel
+//
+// This facility's objective is mix and render audio with low overhead using
+// the OS basic audio support, abstracting as much as possible the
+// idiosyncrasies of each platform. Non-goals:
+// - Positional, 3d audio
+// - Dependence on non-default libraries such as DirectX 9, 10, XAudio
+// - Digital signal processing or effects
+// - Extra features if a specific hardware is installed (EAX, X-fi)
+//
+// The primary client of this facility is audio coming from several tabs.
+// Specifically for this case we avoid supporting complex formats such as MP3
+// or WMA. Complex format decoding should be done by the renderers.
+
+// Models an audio stream that gets rendered to the audio hardware output.
+// Because we support more audio streams than physically available channels
+// a given AudioOutputStream might or might not talk directly to hardware.
+class AudioOutputStream {
+ public:
+  // Audio sources must implement AudioSourceCallback. This interface will be
+  // called in a random thread which very likely is a high priority thread. Do
+  // not rely on using this thread TLS or make calls that alter the thread
+  // itself such as creating Windows or initializing COM.
+  class AudioSourceCallback {
+   public:
+    virtual ~AudioSourceCallback() {}
+
+    // Provide more data by filling |dest| up to |max_size| bytes. The provided
+    // buffer size is usually what is specified in Open(). The source
+    // will return the number of bytes it filled. The expected structure of
+    // |dest| is platform and format specific.
+    virtual size_t OnMoreData(AudioOutputStream* stream,
+                              void* dest, size_t max_size) = 0;
+
+    // The stream is done with this callback. After this call the audio source
+    // can go away or be destroyed.
+    virtual void OnClose(AudioOutputStream* stream) = 0;
+
+    // There was an error while playing a buffer. Audio source cannot be
+    // destroyed yet. No direct action needed by the AudioStream, but it is
+    // a good place to stop accumulating sound data since is is likely that
+    // playback will not continue. |code| is an error code that is platform
+    // specific.
+    virtual void OnError(AudioOutputStream* stream, int code) = 0;
+  };
+
+  // Open the stream. |packet_size| is the requested buffer allocation which
+  // the audio source thinks it can usually fill without blocking. Internally
+  // two buffers of |packet_size| size are created, one will be locked for
+  // playback and one will be ready to be filled in the call to
+  // AudioSourceCallback::OnMoreData().
+  virtual bool Open(size_t packet_size) = 0;
+
+  // Starts playing audio and generating AudioSourceCallback::OnMoreData().
+  virtual void Start(AudioSourceCallback* callback) = 0;
+
+  // Stops playing audio. Effect might no be instantaneous as the hardware
+  // might have locked audio data that is processing.
+  virtual void Stop() = 0;
+
+  // Sets the relative volume, with range [0.0, 1.0] inclusive. For mono audio
+  // sources the volume must be the same in both channels.
+  virtual void SetVolume(double left_level, double right_level) = 0;
+
+  // Gets the relative volume, with range [0.0, 1.0] inclusive. For mono audio
+  // sources the level is returned in both channels.
+  virtual void GetVolume(double* left_level, double* right_level) = 0;
+
+  // Close the stream. This also generates AudioSourceCallback::OnClose().
+  // After calling this method, the object should not be used anymore.
+  virtual void Close() = 0;
+
+ protected:
+  virtual ~AudioOutputStream() {}
+};
+
+// Manages all audio resources. In particular it owns the AudioOutputStream
+// objects. Provides some convenience functions that avoid the need to provide
+// iterators over the existing streams.
+class AudioManager {
+ public:
+  enum Format {
+    AUDIO_PCM_LINEAR,  // Pulse code modulation means 'raw' amplitude samples.
+    AUDIO_PCM_DELTA,   // Delta-encoded pulse code modulation. 
+    AUDIO_MOCK         // Creates a dummy AudioOutputStream object.
+  };
+
+  // Factory for all the supported stream formats. At this moment |channels|
+  // can be 1 (mono) or 2 (stereo). The |sample_rate| is in hertz and can be
+  // any value supported by the underlying platform. For some future formats
+  // the |sample_rate| and |bits_per_sample| can take special values.
+  // Returns NULL if the combination of the parameters is not supported, or if
+  // we have reached some other platform specific limit.
+  //
+  // Do not free the returned AudioOutputStream. It is owned by AudioManager.
+  virtual AudioOutputStream* MakeAudioStream(Format format, int channels,
+                                             int sample_rate,
+                                             char bits_per_sample) = 0;
+
+  // Muting continues playback but effectively the volume is set to zero.
+  // Un-muting returns the volume to the previous level.
+  virtual void MuteAll() = 0;
+  virtual void UnMuteAll() = 0;
+
+ protected:
+  virtual ~AudioManager();
+};
+
+// Get AudioManager singleton.
+// TODO(cpu): Define threading requirements for interacting with AudioManager.
+AudioManager* GetAudioManager();
+
+#endif  // MEDIA_AUDIO_AUDIO_OUTPUT_H_
+