VitaShell/libmad/music_mad.c
2016-09-11 16:59:31 +02:00

329 lines
9.0 KiB
C

/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2004 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
slouken@libsdl.org
*/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include "music_mad.h"
mad_data *
mad_openFile(const char *filename, SDL_AudioSpec *mixer) {
SDL_RWops *rw;
rw = SDL_RWFromFile(filename, "rb");
if (rw == NULL) {
return NULL;
}
return mad_openFileRW(rw, mixer);
}
mad_data *
mad_openFileRW(SDL_RWops *rw, SDL_AudioSpec *mixer) {
mad_data *mp3_mad;
mp3_mad = (mad_data *)malloc(sizeof(mad_data));
mp3_mad->rw = rw;
mad_stream_init(&mp3_mad->stream);
mad_frame_init(&mp3_mad->frame);
mad_synth_init(&mp3_mad->synth);
mp3_mad->frames_read = 0;
mad_timer_reset(&mp3_mad->next_frame_start);
mp3_mad->volume = 128;
mp3_mad->status = 0;
mp3_mad->output_begin = 0;
mp3_mad->output_end = 0;
mp3_mad->mixer = *mixer;
return mp3_mad;
}
void
mad_closeFile(mad_data *mp3_mad) {
SDL_FreeRW(mp3_mad->rw);
mad_stream_finish(&mp3_mad->stream);
mad_frame_finish(&mp3_mad->frame);
mad_synth_finish(&mp3_mad->synth);
free(mp3_mad);
}
/* Starts the playback. */
void
mad_start(mad_data *mp3_mad) {
mp3_mad->status |= MS_playing;
}
/* Stops the playback. */
void
mad_stop(mad_data *mp3_mad) {
mp3_mad->status &= ~MS_playing;
}
/* Returns true if the playing is engaged, false otherwise. */
int
mad_isPlaying(mad_data *mp3_mad) {
return ((mp3_mad->status & MS_playing) != 0);
}
/* Reads the next frame from the file. Returns true on success or
false on failure. */
static int
read_next_frame(mad_data *mp3_mad) {
if (mp3_mad->stream.buffer == NULL ||
mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
size_t read_size;
size_t remaining;
unsigned char *read_start;
/* There might be some bytes in the buffer left over from last
time. If so, move them down and read more bytes following
them. */
if (mp3_mad->stream.next_frame != NULL) {
remaining = mp3_mad->stream.bufend - mp3_mad->stream.next_frame;
memmove(mp3_mad->input_buffer, mp3_mad->stream.next_frame, remaining);
read_start = mp3_mad->input_buffer + remaining;
read_size = MAD_INPUT_BUFFER_SIZE - remaining;
} else {
read_size = MAD_INPUT_BUFFER_SIZE;
read_start = mp3_mad->input_buffer;
remaining = 0;
}
/* Now read additional bytes from the input file. */
read_size = SDL_RWread(mp3_mad->rw, read_start, 1, read_size);
if (read_size <= 0) {
if ((mp3_mad->status & (MS_input_eof | MS_input_error)) == 0) {
if (read_size == 0) {
mp3_mad->status |= MS_input_eof;
} else {
mp3_mad->status |= MS_input_error;
}
/* At the end of the file, we must stuff MAD_BUFFER_GUARD
number of 0 bytes. */
memset(read_start + read_size, 0, MAD_BUFFER_GUARD);
read_size += MAD_BUFFER_GUARD;
}
}
/* Now feed those bytes into the libmad stream. */
mad_stream_buffer(&mp3_mad->stream, mp3_mad->input_buffer,
read_size + remaining);
mp3_mad->stream.error = MAD_ERROR_NONE;
}
/* Now ask libmad to extract a frame from the data we just put in
its buffer. */
if (mad_frame_decode(&mp3_mad->frame, &mp3_mad->stream)) {
if (MAD_RECOVERABLE(mp3_mad->stream.error)) {
return 0;
} else if (mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
return 0;
} else {
mp3_mad->status |= MS_decode_error;
return 0;
}
}
mp3_mad->frames_read++;
mad_timer_add(&mp3_mad->next_frame_start, mp3_mad->frame.header.duration);
return 1;
}
/* Scale a MAD sample to 16 bits for output. */
static signed int
scale(mad_fixed_t sample) {
/* round */
sample += (1L << (MAD_F_FRACBITS - 16));
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
/* Once the frame has been read, copies its samples into the output
buffer. */
static void
decode_frame(mad_data *mp3_mad) {
struct mad_pcm *pcm;
unsigned int nchannels, nsamples;
mad_fixed_t const *left_ch, *right_ch;
unsigned char *out;
mad_synth_frame(&mp3_mad->synth, &mp3_mad->frame);
pcm = &mp3_mad->synth.pcm;
out = mp3_mad->output_buffer + mp3_mad->output_end;
if ((mp3_mad->status & MS_cvt_decoded) == 0) {
mp3_mad->status |= MS_cvt_decoded;
/* The first frame determines some key properties of the stream.
In particular, it tells us enough to set up the convert
structure now. */
SDL_BuildAudioCVT(&mp3_mad->cvt, AUDIO_S16, (Uint8)pcm->channels, mp3_mad->frame.header.samplerate, mp3_mad->mixer.format, mp3_mad->mixer.channels, mp3_mad->mixer.freq);
}
/* pcm->samplerate contains the sampling frequency */
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
while (nsamples--) {
signed int sample;
/* output sample(s) in 16-bit signed little-endian PCM */
sample = scale(*left_ch++);
*out++ = ((sample >> 0) & 0xff);
*out++ = ((sample >> 8) & 0xff);
if (nchannels == 2) {
sample = scale(*right_ch++);
*out++ = ((sample >> 0) & 0xff);
*out++ = ((sample >> 8) & 0xff);
}
}
mp3_mad->output_end = out - mp3_mad->output_buffer;
/*assert(mp3_mad->output_end <= MAD_OUTPUT_BUFFER_SIZE);*/
}
void
mad_getSamples(mad_data *mp3_mad, Uint8 *stream, int len) {
int bytes_remaining;
int num_bytes;
Uint8 *out;
if ((mp3_mad->status & MS_playing) == 0) {
/* We're not supposed to be playing, so send silence instead. */
memset(stream, 0, len);
return;
}
out = stream;
bytes_remaining = len;
while (bytes_remaining > 0) {
if (mp3_mad->output_end == mp3_mad->output_begin) {
/* We need to get a new frame. */
mp3_mad->output_begin = 0;
mp3_mad->output_end = 0;
if (!read_next_frame(mp3_mad)) {
if ((mp3_mad->status & MS_error_flags) != 0) {
/* Couldn't read a frame; either an error condition or
end-of-file. Stop. */
memset(out, 0, bytes_remaining);
mp3_mad->status &= ~MS_playing;
return;
}
} else {
decode_frame(mp3_mad);
/* Now convert the frame data to the appropriate format for
output. */
mp3_mad->cvt.buf = mp3_mad->output_buffer;
mp3_mad->cvt.len = mp3_mad->output_end;
mp3_mad->output_end = (int)(mp3_mad->output_end * mp3_mad->cvt.len_ratio);
/*assert(mp3_mad->output_end <= MAD_OUTPUT_BUFFER_SIZE);*/
SDL_ConvertAudio(&mp3_mad->cvt);
}
}
num_bytes = mp3_mad->output_end - mp3_mad->output_begin;
if (bytes_remaining < num_bytes) {
num_bytes = bytes_remaining;
}
if (mp3_mad->volume == 128) {
memcpy(out, mp3_mad->output_buffer + mp3_mad->output_begin, num_bytes);
} else {
SDL_MixAudio(out, mp3_mad->output_buffer + mp3_mad->output_begin,
num_bytes, mp3_mad->volume);
}
out += num_bytes;
mp3_mad->output_begin += num_bytes;
bytes_remaining -= num_bytes;
}
}
void
mad_seek(mad_data *mp3_mad, double position) {
mad_timer_t target;
int int_part;
int_part = (int)position;
mad_timer_set(&target, int_part,
(int)((position - int_part) * 1000000), 1000000);
if (mad_timer_compare(mp3_mad->next_frame_start, target) > 0) {
/* In order to seek backwards in a VBR file, we have to rewind and
start again from the beginning. This isn't necessary if the
file happens to be CBR, of course; in that case we could seek
directly to the frame we want. But I leave that little
optimization for the future developer who discovers she really
needs it. */
mp3_mad->frames_read = 0;
mad_timer_reset(&mp3_mad->next_frame_start);
mp3_mad->status &= ~MS_error_flags;
mp3_mad->output_begin = 0;
mp3_mad->output_end = 0;
SDL_RWseek(mp3_mad->rw, 0, SEEK_SET);
}
/* Now we have to skip frames until we come to the right one.
Again, only truly necessary if the file is VBR. */
while (mad_timer_compare(mp3_mad->next_frame_start, target) < 0) {
if (!read_next_frame(mp3_mad)) {
if ((mp3_mad->status & MS_error_flags) != 0) {
/* Couldn't read a frame; either an error condition or
end-of-file. Stop. */
mp3_mad->status &= ~MS_playing;
return;
}
}
}
/* Here we are, at the beginning of the frame that contains the
target time. Ehh, I say that's close enough. If we wanted to,
we could get more precise by decoding the frame now and counting
the appropriate number of samples out of it. */
}
void
mad_setVolume(mad_data *mp3_mad, int volume) {
mp3_mad->volume = volume;
}