| // Copyright (c) 2012 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_IO_H_ |
| #define MEDIA_AUDIO_AUDIO_IO_H_ |
| |
| #include <stdint.h> |
| |
| #include "base/time/time.h" |
| #include "media/base/audio_bus.h" |
| #include "media/base/media_export.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. |
| // An audio stream allocates several buffers for audio data and calls |
| // AudioSourceCallback::OnMoreData() periodically to fill these buffers, |
| // as the data is written to the audio device. Size of each packet is determined |
| // by |samples_per_packet| specified in AudioParameters when the stream is |
| // created. |
| |
| namespace media { |
| |
| class MEDIA_EXPORT 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 MEDIA_EXPORT AudioSourceCallback { |
| public: |
| virtual ~AudioSourceCallback() {} |
| |
| // Provide more data by fully filling |dest|. The source will return the |
| // number of frames it filled. |delay| is the duration of audio written to |
| // |dest| in prior calls to OnMoreData() that has not yet been played out, |
| // and |delay_timestamp| is the time when |delay| was measured. The time |
| // when the first sample added to |dest| is expected to be played out can be |
| // calculated by adding |delay| to |delay_timestamp|. The accuracy of |
| // |delay| and |delay_timestamp| may vary depending on the platform and |
| // implementation. |prior_frames_skipped| is the number of frames skipped by |
| // the consumer. |
| virtual int OnMoreData(base::TimeDelta delay, |
| base::TimeTicks delay_timestamp, |
| int prior_frames_skipped, |
| AudioBus* dest) = 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. |
| // |
| // An ErrorType may be provided with more information on what went wrong. An |
| // unhandled kDeviceChange type error is likely to result in further errors; |
| // so it's recommended that sources close their existing output stream and |
| // request a new one when this error is sent. |
| enum class ErrorType { kUnknown, kDeviceChange }; |
| virtual void OnError(ErrorType type) = 0; |
| }; |
| |
| virtual ~AudioOutputStream() {} |
| |
| // Open the stream. false is returned if the stream cannot be opened. Open() |
| // must always be followed by a call to Close() even if Open() fails. |
| virtual bool Open() = 0; |
| |
| // Starts playing audio and generating AudioSourceCallback::OnMoreData(). |
| // Since implementor of AudioOutputStream may have internal buffers, right |
| // after calling this method initial buffers are fetched. |
| // |
| // The output stream does not take ownership of this callback. |
| virtual void Start(AudioSourceCallback* callback) = 0; |
| |
| // Stops playing audio. The operation completes synchronously meaning that |
| // once Stop() has completed executing, no further callbacks will be made to |
| // the callback object that was supplied to Start() and it can be safely |
| // deleted. Stop() may be called in any state, e.g. before Start() or after |
| // Stop(). |
| virtual void Stop() = 0; |
| |
| // Sets the relative volume, with range [0.0, 1.0] inclusive. |
| virtual void SetVolume(double volume) = 0; |
| |
| // Gets the relative volume, with range [0.0, 1.0] inclusive. |
| virtual void GetVolume(double* volume) = 0; |
| |
| // Close the stream. |
| // After calling this method, the object should not be used anymore. |
| // After calling this method, no further AudioSourceCallback methods |
| // should be called on the callback object that was supplied to Start() |
| // by the AudioOutputStream implementation. |
| virtual void Close() = 0; |
| |
| // Flushes the stream. This should only be called if the stream is not |
| // playing. (i.e. called after Stop or Open) |
| virtual void Flush() = 0; |
| }; |
| |
| // Models an audio sink receiving recorded audio from the audio driver. |
| class MEDIA_EXPORT AudioInputStream { |
| public: |
| class MEDIA_EXPORT AudioInputCallback { |
| public: |
| // Called by the audio recorder when a full packet of audio data is |
| // available. This is called from a special audio thread and the |
| // implementation should return as soon as possible. |
| // |
| // |capture_time| is the time at which the first sample in |source| was |
| // received. The age of the audio data may be calculated by subtracting |
| // |capture_time| from base::TimeTicks::Now(). |capture_time| is always |
| // monotonically increasing. |
| virtual void OnData(const AudioBus* source, |
| base::TimeTicks capture_time, |
| double volume) = 0; |
| |
| // There was an error while recording audio. The audio sink cannot be |
| // destroyed yet. No direct action needed by the AudioInputStream, but it |
| // is a good place to stop accumulating sound data since is is likely that |
| // recording will not continue. |
| virtual void OnError() = 0; |
| |
| protected: |
| virtual ~AudioInputCallback() {} |
| }; |
| |
| virtual ~AudioInputStream() {} |
| |
| enum class OpenOutcome { |
| kSuccess, |
| kAlreadyOpen, |
| // Failed due to an unknown or unspecified reason. |
| kFailed, |
| // Failed to open due to OS-level System permissions. |
| kFailedSystemPermissions, |
| // Failed to open as the device is exclusively opened by another app. |
| kFailedInUse, |
| }; |
| |
| // Open the stream and prepares it for recording. Call Start() to actually |
| // begin recording. |
| virtual OpenOutcome Open() = 0; |
| |
| // Starts recording audio and generating AudioInputCallback::OnData(). |
| // The input stream does not take ownership of this callback. |
| virtual void Start(AudioInputCallback* callback) = 0; |
| |
| // Stops recording audio. Effect might not be instantaneous as there could be |
| // pending audio callbacks in the queue which will be issued first before |
| // recording stops. |
| virtual void Stop() = 0; |
| |
| // Close the stream. This also generates AudioInputCallback::OnClose(). This |
| // should be the last call made on this object. |
| virtual void Close() = 0; |
| |
| // Returns the maximum microphone analog volume or 0.0 if device does not |
| // have volume control. |
| virtual double GetMaxVolume() = 0; |
| |
| // Sets the microphone analog volume, with range [0, max_volume] inclusive. |
| virtual void SetVolume(double volume) = 0; |
| |
| // Returns the microphone analog volume, with range [0, max_volume] inclusive. |
| virtual double GetVolume() = 0; |
| |
| // Sets the Automatic Gain Control (AGC) state. |
| virtual bool SetAutomaticGainControl(bool enabled) = 0; |
| |
| // Returns the Automatic Gain Control (AGC) state. |
| virtual bool GetAutomaticGainControl() = 0; |
| |
| // Returns the current muting state for the microphone. |
| virtual bool IsMuted() = 0; |
| |
| // Sets the output device from which to cancel echo, if echo cancellation is |
| // supported by this stream. E.g. called by WebRTC when it changes playback |
| // devices. |
| virtual void SetOutputDeviceForAec(const std::string& output_device_id) = 0; |
| }; |
| |
| } // namespace media |
| |
| #endif // MEDIA_AUDIO_AUDIO_IO_H_ |