blob: 1c0fb0b497ba9db279e6d6828242a3f97b1ae582 [file] [log] [blame]
// Copyright 2017 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.
#include <string>
#include <vector>
#include "chromecast_export.h"
#include "volume_control.h"
namespace chromecast {
namespace media {
class AudioPostProcessor;
} // namespace media
} // namespace chromecast
// Creates an AudioPostProcessor.
// This is applicable only to Alsa CMA backend.
// Please refer to
// chromecast/media/cma/backend/post_processors/
// as an example, but OEM's implementations should not have any
// Chromium dependencies.
// Called from StreamMixerAlsa when shared objects are listed in
// /etc/cast_audio.json
// AudioPostProcessors are created on startup and only destroyed/reset
// if the output sample rate changes.
extern "C" CHROMECAST_EXPORT chromecast::media::AudioPostProcessor*
AudioPostProcessorShlib_Create(const std::string& config, int channels);
namespace chromecast {
namespace media {
// The maximum amount of data that will ever be processed in one call.
const int kMaxAudioWriteTimeMilliseconds = 20;
// Interface for AudioPostProcessors used for applying DSP in StreamMixerAlsa.
class AudioPostProcessor {
// Updates the sample rate of the processor.
// Returns |false| if the processor cannot support |sample_rate|
// Returning false will result in crashing cast_shell.
virtual bool SetSampleRate(int sample_rate) = 0;
// Processes audio frames from |data|, overwriting contents.
// |data| will always be 32-bit interleaved float.
// |frames| is the number of audio frames in data and is
// always non-zero and less than or equal to 20ms of audio.
// AudioPostProcessor must always provide |frames| frames of data back
// (may output 0’s)
// |system_volume| is the Cast Volume applied to the stream
// (normalized to 0-1). It is the same as the cast volume set via alsa.
// |volume_dbfs| is the actual attenuation in dBFS (-inf to 0), equivalent to
// VolumeMap::VolumeToDbFS(|volume|).
// AudioPostProcessor should assume that volume has already been applied.
// Returns the current rendering delay of the filter in frames,
// or negative if an error occurred during processing.
// If an error occurred during processing, |data| should be unchanged.
virtual int ProcessFrames(float* data,
int frames,
float system_volume,
float volume_dbfs) = 0;
// Returns the number of frames of silence it will take for the
// processor to come to rest.
// This may be the actual number of frames stored,
// or may be calculated from internal resonators or similar.
// When inputs are paused, at least this |GetRingingTimeInFrames()| of
// silence will be passed through the processor.
// This is not expected to be real-time;
// It should only change when SetSampleRate is called.
virtual int GetRingingTimeInFrames() = 0;
// Sends a message to the PostProcessor. Implementations are responsible
// for the format and parsing of messages.
// OEM's do not need to implement this method.
virtual void UpdateParameters(const std::string& message) {}
// Set content type to the PostProcessor so it could change processing
// settings accordingly.
virtual void SetContentType(AudioContentType content_type) {}
// Called when device is playing as part of a stereo pair.
// |channel| is the playout channel on this device (0 for left, 1 for right).
// or -1 if the device is not part of a stereo pair.
virtual void SetPlayoutChannel(int channel) {}
virtual ~AudioPostProcessor() = default;
} // namespace media
} // namespace chromecast