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rockbox/firmware/export/enc_base.h
Michael Sevakis 4888131972 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>
2013-06-30 00:40:27 +02:00

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Base declarations for working with software encoders
*
* Copyright (C) 2006-2013 Michael Sevakis
*
* 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 ENC_BASE_H
#define ENC_BASE_H
#include <sys/types.h>
/** Encoder config structures **/
/** aiff_enc.codec **/
struct aiff_enc_config
{
#if 0
unsigned long sample_depth;
#endif
};
/** mp3_enc.codec **/
#define MP3_BITR_CAP_8 (1 << 0)
#define MP3_BITR_CAP_16 (1 << 1)
#define MP3_BITR_CAP_24 (1 << 2)
#define MP3_BITR_CAP_32 (1 << 3)
#define MP3_BITR_CAP_40 (1 << 4)
#define MP3_BITR_CAP_48 (1 << 5)
#define MP3_BITR_CAP_56 (1 << 6)
#define MP3_BITR_CAP_64 (1 << 7)
#define MP3_BITR_CAP_80 (1 << 8)
#define MP3_BITR_CAP_96 (1 << 9)
#define MP3_BITR_CAP_112 (1 << 10)
#define MP3_BITR_CAP_128 (1 << 11)
#define MP3_BITR_CAP_144 (1 << 12)
#define MP3_BITR_CAP_160 (1 << 13)
#define MP3_BITR_CAP_192 (1 << 14)
#define MP3_BITR_CAP_224 (1 << 15)
#define MP3_BITR_CAP_256 (1 << 16)
#define MP3_BITR_CAP_320 (1 << 17)
#define MP3_ENC_NUM_BITR 18
/* MPEG 1 */
#define MPEG1_SAMPR_CAPS (SAMPR_CAP_32 | SAMPR_CAP_48 | SAMPR_CAP_44)
#define MPEG1_BITR_CAPS (MP3_BITR_CAP_32 | MP3_BITR_CAP_40 | \
MP3_BITR_CAP_48 | MP3_BITR_CAP_56 | \
MP3_BITR_CAP_64 | MP3_BITR_CAP_80 | \
MP3_BITR_CAP_96 | MP3_BITR_CAP_112 | \
MP3_BITR_CAP_128 | MP3_BITR_CAP_160 | \
MP3_BITR_CAP_192 | MP3_BITR_CAP_224 | \
MP3_BITR_CAP_256 | MP3_BITR_CAP_320)
/* MPEG 2 */
#define MPEG2_SAMPR_CAPS (SAMPR_CAP_22 | SAMPR_CAP_24 | SAMPR_CAP_16)
#define MPEG2_BITR_CAPS (MP3_BITR_CAP_8 | MP3_BITR_CAP_16 | \
MP3_BITR_CAP_24 | MP3_BITR_CAP_32 | \
MP3_BITR_CAP_40 | MP3_BITR_CAP_48 | \
MP3_BITR_CAP_56 | MP3_BITR_CAP_64 | \
MP3_BITR_CAP_80 | MP3_BITR_CAP_96 | \
MP3_BITR_CAP_112 | MP3_BITR_CAP_128 | \
MP3_BITR_CAP_144 | MP3_BITR_CAP_160)
#if 0
/* MPEG 2.5 */
#define MPEG2_5_SAMPR_CAPS (SAMPR_CAP_8 | SAMPR_CAP_12 | SAMPR_CAP_11)
#define MPEG2_5_BITR_CAPS MPEG2_BITR_CAPS
#endif
/* HAVE_MPEG* defines mainly apply to the bitrate menu */
#if (REC_SAMPR_CAPS & MPEG1_SAMPR_CAPS) || defined (HAVE_SPDIF_REC)
#define HAVE_MPEG1_SAMPR
#endif
#if (REC_SAMPR_CAPS & MPEG2_SAMPR_CAPS) || defined (HAVE_SPDIF_REC)
#define HAVE_MPEG2_SAMPR
#endif
#if 0
#if (REC_SAMPR_CAPS & MPEG2_5_SAMPR_CAPS) || defined (HAVE_SPDIF_REC)
#define HAVE_MPEG2_5_SAMPR
#endif
#endif /* 0 */
#define MP3_ENC_SAMPR_CAPS (MPEG1_SAMPR_CAPS | MPEG2_SAMPR_CAPS)
/* This number is count of full encoder set */
#define MP3_ENC_NUM_SAMPR 6
#if 0
extern const unsigned long mp3_enc_sampr[MP3_ENC_NUM_SAMPR];
#endif
extern const unsigned long mp3_enc_bitr[MP3_ENC_NUM_BITR];
struct mp3_enc_config
{
unsigned long bitrate;
};
#define MP3_ENC_BITRATE_CFG_DEFAULT 11 /* 128 */
#define MP3_ENC_BITRATE_CFG_VALUE_LIST "8,16,24,32,40,48,56,64,80,96," \
"112,128,144,160,192,224,256,320"
/** wav_enc.codec **/
#define WAV_ENC_SAMPR_CAPS SAMPR_CAP_ALL
struct wav_enc_config
{
#if 0
unsigned long sample_depth;
#endif
};
/** wavpack_enc.codec **/
#define WAVPACK_ENC_SAMPR_CAPS SAMPR_CAP_ALL
struct wavpack_enc_config
{
#if 0
unsigned long sample_depth;
#endif
};
/* General config information about any encoder */
struct encoder_config
{
union
{
/* states which *_enc_config member is valid */
int rec_format; /* REC_FORMAT_* value */
int afmt; /* AFMT_* value */
};
union
{
struct mp3_enc_config mp3_enc;
struct wavpack_enc_config wavpack_enc;
struct wav_enc_config wav_enc;
};
};
/** Encoder chunk macros and definitions **/
/* What sort of data does the header describe? */
enum CHUNK_T
{
CHUNK_T_DATA = 0x0, /* Encoded audio data */
CHUNK_T_STREAM_START = 0x1, /* Stream start marker */
CHUNK_T_STREAM_END = 0x2, /* Stream end marker */
CHUNK_T_WRAP = 0x3 /* Buffer early wrap marker */
};
/* Header for every buffer slot and chunk */
union enc_chunk_hdr
{
struct
{
uint32_t type : 2; /* Chunk type (CHUNK_T_*) */
uint32_t err : 1; /* Encoder error */
uint32_t pre : 1; /* Chunk is prerecorded data */
uint32_t aux0 : 1; /* Aux flag 0 - for encoder */
uint32_t unused : 3; /* */
uint32_t size : 24; /* size of data */
};
uint32_t zero; /* Zero-out struct access */
intptr_t reserved1; /* Want it at least pointer-sized */
} __attribute__((__may_alias__));
#define ENC_HDR_SIZE (sizeof (union enc_chunk_hdr))
/* When hdr.type is CHUNK_T_STREAM_START */
struct enc_chunk_file
{
union enc_chunk_hdr hdr; /* This chunk's header */
/* hdr.size = slot count of chunk */
char path[]; /* NULL-terminated path of file */
} __attribute__((__may_alias__));
/* If flags = CHUNK_T_STREAM_END, just the header exists */
/* When hdr.type is CHUNK_T_DATA */
struct enc_chunk_data
{
union enc_chunk_hdr hdr; /* IN,OUT: This chunk's header */
/* hdr.size = total size of data[] */
uint32_t pcm_count; /* OUT: number of PCM samples encoded */
uint8_t data[]; /* OUT: encoded audio data */
} __attribute__((__may_alias__));
/* CHUNK_T_STREAM_END and CHUNK_T_WRAP consist of only the header */
#define ENC_FILE_HDR(hdr) ((struct enc_chunk_file *)(hdr))
#define ENC_DATA_HDR(hdr) ((struct enc_chunk_data *)(hdr))
/* Audio and encoder stream parameters */
struct enc_inputs
{
/* IN parameters */
unsigned long sample_rate; /* PCM samplerate setting */
int num_channels; /* Number of audio channels */
struct encoder_config *config; /* Encoder settings */
/* IN,OUT parameters */
unsigned long enc_sample_rate; /* Actual sample rate accepted by encoder
(for recorded time calculation) */
};
enum enc_callback_reason
{
ENC_CB_INPUTS, /* 'params' is struct enc_inputs * */
ENC_CB_STREAM, /* 'params' is union enc_chunk_hdr * */
};
typedef int (* enc_callback_t)(enum enc_callback_reason reason, void *params);
#endif /* ENC_BASE_H */
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