123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218 |
- // 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;
- virtual int OnMoreData(base::TimeDelta delay,
- base::TimeTicks delay_timestamp,
- int prior_frames_skipped,
- AudioBus* dest,
- bool is_mixing);
- // 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_
|