rockbox/firmware/export/audio.h

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 by Linus Nielsen Feltzing
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#ifndef __AUDIO_H
#define __AUDIO_H
#include <stdbool.h>
#include <string.h> /* size_t */
#include "config.h"
/* These must always be included with audio.h for this to compile under
cetain conditions. Do it here or else spread the complication around to
many files. */
#include "pcm_sampr.h"
#include "pcm.h"
#ifdef HAVE_RECORDING
#include "enc_base.h"
#endif /* HAVE_RECORDING */
#define AUDIO_STATUS_PLAY 0x0001
#define AUDIO_STATUS_PAUSE 0x0002
#define AUDIO_STATUS_RECORD 0x0004
#define AUDIO_STATUS_PRERECORD 0x0008
#define AUDIO_STATUS_ERROR 0x0010
#define AUDIO_STATUS_WARNING 0x0020
#define AUDIOERR_DISK_FULL 1
#define AUDIO_GAIN_LINEIN 0
#define AUDIO_GAIN_MIC 1
void audio_init(void) INIT_ATTR;
void audio_play(unsigned long elapsed, unsigned long offset);
void audio_stop(void);
/* Stops audio from serving playback and frees resources*/
void audio_hard_stop(void);
void audio_pause(void);
void audio_resume(void);
void audio_next(void);
void audio_prev(void);
int audio_status(void);
/* size of the audio buffer */
size_t audio_buffer_size(void);
/* size of the buffer available for allocating memory from the audio buffer using core_*()
* returns core_available() if audio buffer is not allocated yet */
size_t audio_buffer_available(void);
void audio_ff_rewind(long newpos);
void audio_flush_and_reload_tracks(void);
struct mp3entry* audio_current_track(void);
struct mp3entry* audio_next_track(void);
bool audio_peek_track(struct mp3entry* id3, int offset);
#ifdef HAVE_DISK_STORAGE
void audio_set_buffer_margin(int seconds);
#endif
void audio_error_clear(void);
int audio_get_file_pos(void);
void audio_beep(int duration);
void audio_next_dir(void);
void audio_prev_dir(void);
/* channel modes */
enum rec_channel_modes
{
__CHN_MODE_START_INDEX = -1,
CHN_MODE_STEREO,
CHN_MODE_MONO,
CHN_NUM_MODES
};
/* channel mode capability bits */
#define CHN_CAP_STEREO (1 << CHN_MODE_STEREO)
#define CHN_CAP_MONO (1 << CHN_MODE_MONO)
#define CHN_CAP_ALL (CHN_CAP_STEREO | CHN_CAP_MONO)
/* the enums below must match prestr[] in recording.c */
enum audio_sources
{
AUDIO_SRC_PLAYBACK = -1, /* Virtual source */
HAVE_MIC_IN_(AUDIO_SRC_MIC,)
HAVE_LINE_IN_(AUDIO_SRC_LINEIN,)
HAVE_SPDIF_IN_(AUDIO_SRC_SPDIF,)
HAVE_FMRADIO_IN_(AUDIO_SRC_FMRADIO,)
AUDIO_NUM_SOURCES,
AUDIO_SRC_MAX = AUDIO_NUM_SOURCES-1,
AUDIO_SRC_DEFAULT = AUDIO_SRC_PLAYBACK
};
extern int audio_channels;
extern int audio_output_source;
#ifdef HAVE_RECORDING
/* Recordable source implies it has the input as well */
/* For now there's no restrictions on any targets with which inputs
are recordable so define them as equivalent - if they do differ,
special handling is needed right now. */
enum rec_sources
{
__REC_SRC_FIRST = -1,
HAVE_MIC_REC_(REC_SRC_MIC,)
HAVE_LINE_REC_(REC_SRC_LINEIN,)
HAVE_SPDIF_REC_(REC_SRC_SPDIF,)
HAVE_FMRADIO_REC_(REC_SRC_FMRADIO,)
REC_NUM_SOURCES
};
#endif /* HAVE_RECORDING */
/* selects a source to monitor for recording or playback */
#define SRCF_PLAYBACK 0x0000 /* default */
#define SRCF_RECORDING 0x1000
#if CONFIG_TUNER
/* for AUDIO_SRC_FMRADIO */
#define SRCF_FMRADIO_PLAYING 0x0000 /* default */
#define SRCF_FMRADIO_PAUSED 0x2000
#endif
#ifdef HAVE_RECORDING
/* parameters for audio_set_recording_options */
struct audio_recording_options
{
int rec_source;
int rec_frequency;
int rec_channels;
int rec_prerecord_time;
int rec_mono_mode;
int rec_source_flags; /* for rec_set_source */
struct encoder_config enc_config;
};
/* audio recording functions */
void audio_init_recording(void);
void audio_close_recording(void);
void audio_record(const char *filename);
void audio_stop_recording(void);
void audio_pause_recording(void);
void audio_resume_recording(void);
void audio_new_file(const char *filename);
void audio_set_recording_options(struct audio_recording_options *options);
void audio_set_recording_gain(int left, int right, int type);
unsigned long audio_recorded_time(void);
unsigned long audio_num_recorded_bytes(void);
Update software recording engine to latest codec interface. Basically, just give it a good rewrite. Software codec recording can be implemented in a more straightforward and simple manner and made more robust through the better codec control now available. Encoded audio buffer uses a packed format instead of fixed-size chunks and uses smaller data headers leading to more efficient usage. The greatest benefit is with a VBR format like wavpack which needs to request a maximum size but only actually ends up committing part of that request. No guard buffers are used for either PCM or encoded audio. PCM is read into the codec's provided buffer and mono conversion done at that time in the core if required. Any highly-specialized sample conversion is still done within the codec itself, such as 32-bit (wavpack) or interleaved mono (mp3). There is no longer a separate filename array. All metadata goes onto the main encoded audio buffer, eliminating any predermined file limit on the buffer as well as not wasting the space for unused path queue slots. The core and codec interface is less awkward and a bit more sensible. Some less useful interface features were removed. Threads are kept on narrow code paths ie. the audio thread never calls encoding functions and the codec thread never calls file functions as before. Codecs no longer call file functions directly. Writes are buffered in the core and data written to storage in larger chunks to speed up flushing of data. In fact, codecs are no longer aware of the stream being a file at all and have no access to the fd. SPDIF frequency detection no longer requires a restart of recording or plugging the source before entering the screen. It will poll for changes and update when stopped or prerecording (which does discard now-invalid prerecorded data). I've seen to it that writing a proper header on full disk works when the format makes it reasonably practical to do so. Other cases may have incorrect data sizes but sample info will be in tact. File left that way may play anyway. mp3_enc.codec acquires the ability to write 'Info' headers with LAME tags to make it gapless (bonus). Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653 Reviewed-on: http://gerrit.rockbox.org/493 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2013-06-22 20:41:16 +00:00
unsigned long audio_prerecorded_time(void);
#endif /* HAVE_RECORDING */
#if INPUT_SRC_CAPS != 0
/* audio.c */
void audio_set_input_source(int source, unsigned flags);
/* audio_input_mux: target-specific implementation used by audio_set_source
to set hardware inputs and audio paths */
void audio_input_mux(int source, unsigned flags);
void audio_set_output_source(int source);
#endif /* INPUT_SRC_CAPS */
#ifdef HAVE_SPDIF_IN
/* returns index into rec_master_sampr_list */
int audio_get_spdif_sample_rate(void);
/* > 0: monitor EBUin, 0: Monitor IISrecv, <0: reset only */
void audio_spdif_set_monitor(int monitor_spdif);
#endif /* HAVE_SPDIF_IN */
#ifdef HAVE_SPEAKER
/* enable/disable the speaker: 0=off, 1=on, 2=on if jack unpluged, off otherwise
* NOTE this is a one time thing, this function doesn't implement the logic to
* check for jack events, it merely changes the speaker state to the expected
* state based on the requested mode.
*/
void audio_enable_speaker(int mode);
#endif
/***********************************************************************/
/* audio event handling */
enum track_event_flags
{
TEF_NONE = 0x0, /* no flags are set */
TEF_CURRENT = 0x1, /* event is for the current track */
TEF_AUTO_SKIP = 0x2, /* event is sent in context of auto skip */
TEF_REWIND = 0x4, /* interpret as rewind, id3->elapsed is the
position before the seek back to 0 */
};
struct track_event
{
unsigned int flags; /* combo of enum track_event_flags values */
struct mp3entry *id3; /* pointer to mp3entry describing track */
};
#endif /* __AUDIO_H */