// 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. // // AudioConverter is a complete mixing, resampling, buffering, and channel // mixing solution for converting data from one set of AudioParameters to // another. // // For efficiency, pieces are only invoked when necessary; i.e., // - The resampler is only used if sample rates differ. // - The FIFO is only used if buffer sizes differ. // - The channel mixer is only used if channel layouts differ. // // Additionally, since resampling is the most expensive operation, input mixing // and channel down mixing are done prior to resampling. Likewise, channel up // mixing is performed after resampling. #ifndef MEDIA_BASE_AUDIO_CONVERTER_H_ #define MEDIA_BASE_AUDIO_CONVERTER_H_ #include #include "base/callback.h" #include "base/memory/scoped_ptr.h" #include "base/time/time.h" #include "media/audio/audio_parameters.h" #include "media/base/media_export.h" namespace media { class AudioBus; class AudioPullFifo; class ChannelMixer; class MultiChannelResampler; // Converts audio data between two AudioParameters formats. Sample usage: // AudioParameters input(...), output(...); // AudioConverter ac(input, output); // scoped_ptr output_audio_bus = AudioBus::Create(output); // ac.AddInput(); // ac.AddInput(); // ac.Convert(output_audio_bus.get()); // // Convert() will ask for input audio data from each InputCallback and convert // the data into the provided AudioBus. class MEDIA_EXPORT AudioConverter { public: // Interface for inputs into the converter. Each InputCallback is added or // removed from Convert() processing via AddInput() and RemoveInput(). class MEDIA_EXPORT InputCallback { public: // Method for providing more data into the converter. Expects |audio_bus| // to be completely filled with data upon return; zero padded if not enough // frames are available to satisfy the request. The return value is the // volume level of the provided audio data. If a volume level of zero is // returned no further processing will be done on the provided data, else // the volume level will be used to scale the provided audio data. virtual double ProvideInput(AudioBus* audio_bus, base::TimeDelta buffer_delay) = 0; protected: virtual ~InputCallback() {} }; // Constructs an AudioConverter for converting between the given input and // output parameters. Specifying |disable_fifo| means all InputCallbacks are // capable of handling arbitrary buffer size requests; i.e. one call might ask // for 10 frames of data (indicated by the size of AudioBus provided) and the // next might ask for 20. In synthetic testing, disabling the FIFO yields a // ~20% speed up for common cases. AudioConverter(const AudioParameters& input_params, const AudioParameters& output_params, bool disable_fifo); ~AudioConverter(); // Converts audio from all inputs into the |dest|. If an |initial_delay| is // specified, it will be propagated to each input. void Convert(AudioBus* dest); void ConvertWithDelay(const base::TimeDelta& initial_delay, AudioBus* dest); // Adds or removes an input from the converter. RemoveInput() will call // Reset() if no inputs remain after the specified input is removed. void AddInput(InputCallback* input); void RemoveInput(InputCallback* input); // Flushes all buffered data. void Reset(); // The maximum size in frames that guarantees we will only make a single call // to each input's ProvideInput for more data. int ChunkSize() const; private: // Provides input to the MultiChannelResampler. Called by the resampler when // more data is necessary. void ProvideInput(int resampler_frame_delay, AudioBus* audio_bus); // Provides input to the AudioPullFifo. Called by the fifo when more data is // necessary. void SourceCallback(int fifo_frame_delay, AudioBus* audio_bus); // (Re)creates the temporary |unmixed_audio_| buffer if necessary. void CreateUnmixedAudioIfNecessary(int frames); // Set of inputs for Convert(). typedef std::list InputCallbackSet; InputCallbackSet transform_inputs_; // Used to buffer data between the client and the output device in cases where // the client buffer size is not the same as the output device buffer size. scoped_ptr audio_fifo_; int chunk_size_; // Handles resampling. scoped_ptr resampler_; // Handles channel transforms. |unmixed_audio_| is a temporary destination // for audio data before it goes into the channel mixer. scoped_ptr channel_mixer_; scoped_ptr unmixed_audio_; // Temporary AudioBus destination for mixing inputs. scoped_ptr mixer_input_audio_bus_; // Since resampling is expensive, figure out if we should downmix channels // before resampling. bool downmix_early_; // Used to calculate buffer delay information for InputCallbacks. base::TimeDelta input_frame_duration_; base::TimeDelta output_frame_duration_; base::TimeDelta initial_delay_; int resampler_frame_delay_; // Number of channels of input audio data. Set during construction via the // value from the input AudioParameters class. Preserved to recreate internal // AudioBus structures on demand in response to varying frame size requests. const int input_channel_count_; DISALLOW_COPY_AND_ASSIGN(AudioConverter); }; } // namespace media #endif // MEDIA_BASE_AUDIO_CONVERTER_H_