SDL_mixer: An audio mixer library based on the SDL library

Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>

This software is provided 'as-is', without any express or implied

warranty. In no event will the authors be held liable for any damages

arising from the use of this software.

Permission is granted to anyone to use this software for any purpose,

including commercial applications, and to alter it and redistribute it

freely, subject to the following restrictions:

1. The origin of this software must not be misrepresented; you must not

claim that you wrote the original software. If you use this software

in a product, an acknowledgment in the product documentation would be

appreciated but is not required.

2. Altered source versions must be plainly marked as such, and must not be

misrepresented as being the original software.

3. This notice may not be removed or altered from any source distribution.

*/

#ifdef MP3_MAD_MUSIC

#include <string.h>

#include "music_mad.h"

mad_data *

mad_openFileRW(SDL_RWops *rw, SDL_AudioSpec *mixer, int freerw)

{

mad_data *mp3_mad;

mp3_mad = (mad_data *)malloc(sizeof(mad_data));

if (mp3_mad) {

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 = MIX_MAX_VOLUME;

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)

{

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;

int ret;

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, 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);*/

}

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;

}

} 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 == MIX_MAX_VOLUME) {

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;

}

/* 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;

}

#endif /* MP3_MAD_MUSIC */