mirror of
https://github.com/joel16/VitaShell.git
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329 lines
9.0 KiB
C
329 lines
9.0 KiB
C
/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2004 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public
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License along with this library; if not, write to the Free
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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Sam Lantinga
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slouken@libsdl.org
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*/
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#include <stdio.h>
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#include <string.h>
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#include <stdlib.h>
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#include "music_mad.h"
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mad_data *
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mad_openFile(const char *filename, SDL_AudioSpec *mixer) {
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SDL_RWops *rw;
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rw = SDL_RWFromFile(filename, "rb");
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if (rw == NULL) {
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return NULL;
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}
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return mad_openFileRW(rw, mixer);
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}
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mad_data *
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mad_openFileRW(SDL_RWops *rw, SDL_AudioSpec *mixer) {
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mad_data *mp3_mad;
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mp3_mad = (mad_data *)malloc(sizeof(mad_data));
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mp3_mad->rw = rw;
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mad_stream_init(&mp3_mad->stream);
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mad_frame_init(&mp3_mad->frame);
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mad_synth_init(&mp3_mad->synth);
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mp3_mad->frames_read = 0;
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mad_timer_reset(&mp3_mad->next_frame_start);
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mp3_mad->volume = 128;
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mp3_mad->status = 0;
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mp3_mad->output_begin = 0;
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mp3_mad->output_end = 0;
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mp3_mad->mixer = *mixer;
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return mp3_mad;
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}
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void
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mad_closeFile(mad_data *mp3_mad) {
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SDL_FreeRW(mp3_mad->rw);
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mad_stream_finish(&mp3_mad->stream);
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mad_frame_finish(&mp3_mad->frame);
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mad_synth_finish(&mp3_mad->synth);
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free(mp3_mad);
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}
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/* Starts the playback. */
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void
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mad_start(mad_data *mp3_mad) {
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mp3_mad->status |= MS_playing;
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}
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/* Stops the playback. */
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void
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mad_stop(mad_data *mp3_mad) {
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mp3_mad->status &= ~MS_playing;
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}
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/* Returns true if the playing is engaged, false otherwise. */
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int
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mad_isPlaying(mad_data *mp3_mad) {
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return ((mp3_mad->status & MS_playing) != 0);
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}
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/* Reads the next frame from the file. Returns true on success or
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false on failure. */
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static int
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read_next_frame(mad_data *mp3_mad) {
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if (mp3_mad->stream.buffer == NULL ||
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mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
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size_t read_size;
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size_t remaining;
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unsigned char *read_start;
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/* There might be some bytes in the buffer left over from last
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time. If so, move them down and read more bytes following
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them. */
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if (mp3_mad->stream.next_frame != NULL) {
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remaining = mp3_mad->stream.bufend - mp3_mad->stream.next_frame;
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memmove(mp3_mad->input_buffer, mp3_mad->stream.next_frame, remaining);
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read_start = mp3_mad->input_buffer + remaining;
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read_size = MAD_INPUT_BUFFER_SIZE - remaining;
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} else {
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read_size = MAD_INPUT_BUFFER_SIZE;
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read_start = mp3_mad->input_buffer;
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remaining = 0;
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}
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/* Now read additional bytes from the input file. */
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read_size = SDL_RWread(mp3_mad->rw, read_start, 1, read_size);
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if (read_size <= 0) {
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if ((mp3_mad->status & (MS_input_eof | MS_input_error)) == 0) {
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if (read_size == 0) {
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mp3_mad->status |= MS_input_eof;
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} else {
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mp3_mad->status |= MS_input_error;
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}
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/* At the end of the file, we must stuff MAD_BUFFER_GUARD
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number of 0 bytes. */
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memset(read_start + read_size, 0, MAD_BUFFER_GUARD);
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read_size += MAD_BUFFER_GUARD;
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}
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}
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/* Now feed those bytes into the libmad stream. */
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mad_stream_buffer(&mp3_mad->stream, mp3_mad->input_buffer,
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read_size + remaining);
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mp3_mad->stream.error = MAD_ERROR_NONE;
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}
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/* Now ask libmad to extract a frame from the data we just put in
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its buffer. */
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if (mad_frame_decode(&mp3_mad->frame, &mp3_mad->stream)) {
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if (MAD_RECOVERABLE(mp3_mad->stream.error)) {
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return 0;
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} else if (mp3_mad->stream.error == MAD_ERROR_BUFLEN) {
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return 0;
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} else {
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mp3_mad->status |= MS_decode_error;
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return 0;
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}
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}
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mp3_mad->frames_read++;
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mad_timer_add(&mp3_mad->next_frame_start, mp3_mad->frame.header.duration);
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return 1;
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}
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/* Scale a MAD sample to 16 bits for output. */
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static signed int
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scale(mad_fixed_t sample) {
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/* round */
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sample += (1L << (MAD_F_FRACBITS - 16));
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/* clip */
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if (sample >= MAD_F_ONE)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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/* quantize */
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return sample >> (MAD_F_FRACBITS + 1 - 16);
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}
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/* Once the frame has been read, copies its samples into the output
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buffer. */
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static void
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decode_frame(mad_data *mp3_mad) {
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struct mad_pcm *pcm;
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unsigned int nchannels, nsamples;
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mad_fixed_t const *left_ch, *right_ch;
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unsigned char *out;
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mad_synth_frame(&mp3_mad->synth, &mp3_mad->frame);
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pcm = &mp3_mad->synth.pcm;
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out = mp3_mad->output_buffer + mp3_mad->output_end;
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if ((mp3_mad->status & MS_cvt_decoded) == 0) {
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mp3_mad->status |= MS_cvt_decoded;
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/* The first frame determines some key properties of the stream.
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In particular, it tells us enough to set up the convert
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structure now. */
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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);
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}
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/* pcm->samplerate contains the sampling frequency */
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nchannels = pcm->channels;
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nsamples = pcm->length;
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left_ch = pcm->samples[0];
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right_ch = pcm->samples[1];
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while (nsamples--) {
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signed int sample;
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/* output sample(s) in 16-bit signed little-endian PCM */
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sample = scale(*left_ch++);
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*out++ = ((sample >> 0) & 0xff);
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*out++ = ((sample >> 8) & 0xff);
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if (nchannels == 2) {
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sample = scale(*right_ch++);
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*out++ = ((sample >> 0) & 0xff);
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*out++ = ((sample >> 8) & 0xff);
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}
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}
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mp3_mad->output_end = out - mp3_mad->output_buffer;
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/*assert(mp3_mad->output_end <= MAD_OUTPUT_BUFFER_SIZE);*/
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}
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void
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mad_getSamples(mad_data *mp3_mad, Uint8 *stream, int len) {
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int bytes_remaining;
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int num_bytes;
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Uint8 *out;
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if ((mp3_mad->status & MS_playing) == 0) {
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/* We're not supposed to be playing, so send silence instead. */
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memset(stream, 0, len);
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return;
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}
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out = stream;
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bytes_remaining = len;
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while (bytes_remaining > 0) {
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if (mp3_mad->output_end == mp3_mad->output_begin) {
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/* We need to get a new frame. */
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mp3_mad->output_begin = 0;
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mp3_mad->output_end = 0;
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if (!read_next_frame(mp3_mad)) {
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if ((mp3_mad->status & MS_error_flags) != 0) {
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/* Couldn't read a frame; either an error condition or
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end-of-file. Stop. */
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memset(out, 0, bytes_remaining);
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mp3_mad->status &= ~MS_playing;
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return;
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}
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} else {
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decode_frame(mp3_mad);
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/* Now convert the frame data to the appropriate format for
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output. */
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mp3_mad->cvt.buf = mp3_mad->output_buffer;
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mp3_mad->cvt.len = mp3_mad->output_end;
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mp3_mad->output_end = (int)(mp3_mad->output_end * mp3_mad->cvt.len_ratio);
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/*assert(mp3_mad->output_end <= MAD_OUTPUT_BUFFER_SIZE);*/
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SDL_ConvertAudio(&mp3_mad->cvt);
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}
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}
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num_bytes = mp3_mad->output_end - mp3_mad->output_begin;
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if (bytes_remaining < num_bytes) {
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num_bytes = bytes_remaining;
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}
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if (mp3_mad->volume == 128) {
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memcpy(out, mp3_mad->output_buffer + mp3_mad->output_begin, num_bytes);
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} else {
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SDL_MixAudio(out, mp3_mad->output_buffer + mp3_mad->output_begin,
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num_bytes, mp3_mad->volume);
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}
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out += num_bytes;
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mp3_mad->output_begin += num_bytes;
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bytes_remaining -= num_bytes;
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}
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}
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void
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mad_seek(mad_data *mp3_mad, double position) {
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mad_timer_t target;
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int int_part;
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int_part = (int)position;
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mad_timer_set(&target, int_part,
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(int)((position - int_part) * 1000000), 1000000);
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if (mad_timer_compare(mp3_mad->next_frame_start, target) > 0) {
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/* In order to seek backwards in a VBR file, we have to rewind and
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start again from the beginning. This isn't necessary if the
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file happens to be CBR, of course; in that case we could seek
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directly to the frame we want. But I leave that little
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optimization for the future developer who discovers she really
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needs it. */
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mp3_mad->frames_read = 0;
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mad_timer_reset(&mp3_mad->next_frame_start);
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mp3_mad->status &= ~MS_error_flags;
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mp3_mad->output_begin = 0;
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mp3_mad->output_end = 0;
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SDL_RWseek(mp3_mad->rw, 0, SEEK_SET);
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}
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/* Now we have to skip frames until we come to the right one.
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Again, only truly necessary if the file is VBR. */
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while (mad_timer_compare(mp3_mad->next_frame_start, target) < 0) {
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if (!read_next_frame(mp3_mad)) {
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if ((mp3_mad->status & MS_error_flags) != 0) {
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/* Couldn't read a frame; either an error condition or
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end-of-file. Stop. */
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mp3_mad->status &= ~MS_playing;
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return;
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}
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}
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}
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/* Here we are, at the beginning of the frame that contains the
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target time. Ehh, I say that's close enough. If we wanted to,
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we could get more precise by decoding the frame now and counting
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the appropriate number of samples out of it. */
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}
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void
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mad_setVolume(mad_data *mp3_mad, int volume) {
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mp3_mad->volume = volume;
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}
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