void audio_callback(void *userdata, Uint8 *stream, int len)
{
VideoState *is = (VideoState*)userdata;
int len1, audio_size;
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
double pts;
while(len > 0) {
if(audio_buf_index >= audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is, audio_buf, audio_buf_size,&pts);
if(audio_size < 0) {
/* If error, output silence */
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
// 一次 audio_decode_frame 出来的 audio_buf_size 可能比 len 更大， memcpy 不完，
// 下次 audio_callback 继续 memcpy 剩下的数据，而不再调用 audio_decode_frame
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
while(len > 0) {
if(audio_buf_index >= audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
if(audio_size < 0) {
/* If error, output silence */
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len){
//VideoState *is = (VideoState *) userdata;
SDL_LockMutex(global_mutex_lock);
VideoState *is = global_video_state[global_audioIndex];
SDL_UnlockMutex(global_mutex_lock);
int len1, audio_size;
double pts;
while (len > 0){
if (is->audio_buf_index >= is->audio_buf_size){
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is, &pts);
if (audio_size < 0){
/* If error, output silence */
is->audio_buf_size = 1024;
memset(is->audio_buf, 0, is->audio_buf_size);
}else{
audio_size = synchronize_audio(is, (int16_t *) is->audio_buf,audio_size, pts);
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len){
len1 = len;
}
memcpy(stream, (uint8_t *) is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
/*
* Read up to len samples of type sox_sample_t from file into buf[].
* Return number of samples read.
*/
static size_t read_samples(sox_format_t * ft, sox_sample_t *buf, size_t len)
{
priv_t * ffmpeg = (priv_t *)ft->priv;
AVPacket *pkt = &ffmpeg->audio_pkt;
int ret;
size_t nsamp = 0, nextra;
/* Read data repeatedly until buf is full or no more can be read */
do {
/* If input buffer empty, read more data */
if (ffmpeg->audio_buf_index * 2 >= ffmpeg->audio_buf_size) {
if ((ret = av_read_frame(ffmpeg->ctxt, pkt)) < 0)
break;
ffmpeg->audio_buf_size = audio_decode_frame(ffmpeg, ffmpeg->audio_buf, AVCODEC_MAX_AUDIO_FRAME_SIZE);
ffmpeg->audio_buf_index = 0;
}
/* Convert data into SoX samples up to size of buffer */
nextra = min((ffmpeg->audio_buf_size - ffmpeg->audio_buf_index) / 2, (int)(len - nsamp));
for (; nextra > 0; nextra--)
buf[nsamp++] = SOX_SIGNED_16BIT_TO_SAMPLE(((int16_t *)ffmpeg->audio_buf)[ffmpeg->audio_buf_index++], ft->clips);
} while (nsamp < len && nextra > 0);
return nsamp;
}
void audio_callback(void *userdata, Uint8 *stream, int len)
{
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
while(len > 0)
{
if(audio_buf_index >= audio_buf_size)
{
audio_size = audio_decode_frame(aCodecCtx, audio_buf,sizeof(audio_buf));
if(audio_size < 0)
{
audio_buf_size = 1024;
memset(audio_buf, 0, audio_buf_size);
}
else
{
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
SDL_MixAudio(stream, (uint8_t * )audio_buf + audio_buf_index, len1, volume);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
VideoState *is = (VideoState *)userdata;
int len1, audio_size;
double pts;
while(len > 0) {
if(is->audio_buf_index >= is->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is, &pts);
if(audio_size < 0) {
/* If error, output silence */
is->audio_buf_size = 1024;
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
/* prepare a new audio buffer */
void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
/*SDL AUDIO THREAD*/
FFMovie *movie = opaque;
int audio_size, len1;
double pts;
while (len > 0) {
if (movie->audio_buf_index >= movie->audio_buf_size) {
audio_size = audio_decode_frame(movie, movie->audio_buf, &pts);
if (audio_size < 0) {
/* if error, just output silence */
movie->audio_buf_size = 1024;
memset(movie->audio_buf, 0, movie->audio_buf_size);
} else {
audio_size = synchronize_audio(movie, (int16_t*)movie->audio_buf, audio_size, pts);
movie->audio_buf_size = audio_size;
}
movie->audio_buf_index = 0;
}
len1 = movie->audio_buf_size - movie->audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)movie->audio_buf + movie->audio_buf_index, len1);
len -= len1;
stream += len1;
movie->audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
FFmpegState *st = (FFmpegState *)userdata;
int len1, audio_data_size;
while (len > 0) {
if (st->audio_buf_index >= st->audio_buf_size) {
audio_data_size = audio_decode_frame(st);
if (audio_data_size < 0) {
/* silence */
st->audio_buf_size = 1024;
memset(st->audio_buf, 0, st->audio_buf_size);
}
else {
st->audio_buf_size = audio_data_size;
}
st->audio_buf_index = 0;
}
len1 = st->audio_buf_size - st->audio_buf_index;
if (len1 > len) {
len1 = len;
}
memcpy(stream, (uint8_t *)st->audio_buf + st->audio_buf_index, len1);
len -= len1;
stream += len1;
st->audio_buf_index += len1;
}
}
// SDL audio callback: to get decoded audio frame data.
// stream: [out] decoded audio frame data.
// len : [in ] data length that SDL needs.
void audio_callback(void *userdata, Uint8 *stream, int len)
{
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size; // Decoded audio data size
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
printf("\r\n ==============>audio_callback start\r\n");
while(len > 0) { // 仍未满足SDL需求数据量, 需继续向解码器要数据.
if(audio_buf_index >= audio_buf_size) { // 上一次解码出来的数据已经推完, 需继续解码.
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf)); // 解码音频帧后的数据放入audio_buf
printf("\r\n audio_size:%d!\r\n", audio_size);
if(audio_size < 0) {
/* If error, output silence */
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
// printf("\r\n len1:%d!\r\n", len1);
// printf("\r\n len:%d!\r\n", len);
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void sdl_audio_callback(void *userdata, Uint8 *stream, int len) {
VideoState *is = (VideoState *)userdata;
int len1, audio_size;
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is);
if (audio_size < 0) {
//TODO
/* if error, just output silence */
}
else {
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
void audio_callback(void* userdata, Uint8* stream, int len) {
VideoState *is = (VideoState*) userdata;
int len1, audio_size;
double pts;
printf("audio_callback len:%d\n", len);
fflush(stdout);
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(is, is->audio_buf,
sizeof(is->audio_buf), &pts);
printf("audio_decode_frame audio_size=%d\n", audio_size);
fflush(stdout);
if (audio_size < 0) {
is->audio_buf_size = 1024;
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
audio_size = synchronize_audio(is, (int16_t *) is->audio_buf,
audio_size, pts);
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *) is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
AVCodecContext *aCodeCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
while (len > 0) {
if (audio_buf_index >= audio_buf_size) {
// we have already sent all our data; get more
audio_size = audio_decode_frame(aCodeCtx, audio_buf, audio_buf_size);
if (audio_size < 0) {
// If error, output silence
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if (len1 > len) {
len1 = len;
}
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback( void *userdata, uint8_t *stream, int len )
{
//printf( "In Audio callback : Thread\n" );
//printf("Length is %d\n",len);
static int stream_buf_index = 0;
static int audio_data_size = 0;
static int audio_buf_index = 0;
int free_flag = 0;
while ( len > 0 && gMedia->quit[2] == no ) {
if ( audio_data_size == 0 ) {
audio_buf_index = 0;
audio_data_size = audio_decode_frame();
//printf("%d\n",audio_data_size);
free_flag = 1;
}
if ( audio_data_size == -1 ) {
return;
}
if ( len <= audio_data_size ) {
memcpy( stream + stream_buf_index,
*( gMedia->audio_buf ) + audio_buf_index,
len );
//printf("%d bytes copied : %d bytes left in buffer\t",len,audio_data_size);
audio_data_size -= len;
audio_buf_index += len;
// printf("%d\n",audio_buf_index);
len = 0;
stream_buf_index = 0;
break;
}
if ( len > audio_data_size ) {
memcpy( stream + stream_buf_index,
*( gMedia->audio_buf ) + audio_buf_index,
audio_data_size );
len -= audio_data_size;
stream_buf_index += audio_data_size;
audio_data_size = 0;
audio_buf_index = 0;
continue;
}
}
if ( free_flag && !audio_data_size ) {
av_free( gMedia->audio_buf[0] );
}
}
static void audio_callback(void *userdata, Uint8 *stream, int len) {
VideoState *is = (VideoState *) userdata;
int len1, audio_data_size;
double pts;
/* len是由SDL传入的SDL缓冲区的大小，如果这个缓冲未满，我们就一直往里填充数据 */
while (len > 0) {
/* audio_buf_index 和 audio_buf_size 标示我们自己用来放置解码出来的数据的缓冲区，*/
/* 这些数据待copy到SDL缓冲区， 当audio_buf_index >= audio_buf_size的时候意味着我*/
/* 们的缓冲为空，没有数据可供copy，这时候需要调用audio_decode_frame来解码出更
/* 多的桢数据 */
if (is->audio_buf_index >= is->audio_buf_size) {
audio_data_size = audio_decode_frame(is, &pts);
/* audio_data_size < 0 标示没能解码出数据，我们默认播放静音 */
if (audio_data_size < 0) {
/* silence */
is->audio_buf_size = 1024;
/* 清零，静音 */
if (is->audio_buf == NULL) return;
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
is->audio_buf_size = audio_data_size;
}
is->audio_buf_index = 0;
}
/* 查看stream可用空间，决定一次copy多少数据，剩下的下次继续copy */
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len) {
len1 = len;
}
if (is->audio_buf == NULL) return;
memcpy(stream, (uint8_t *) is->audio_buf + is->audio_buf_index, len1);
// SDL_MixAudio(stream, (uint8_t * )is->audio_buf + is->audio_buf_index, len1, 50);
// SDL_MixAudioFormat(stream, (uint8_t * )is->audio_buf + is->audio_buf_index, AUDIO_S16SYS, len1, 50);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len)
{
//ffplay_info("Start.\n");
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
ffplay_info("len = %d, audio_buf_index = %d, audio_buf_size = %d\n",len,audio_buf_index,audio_buf_size);
while(len > 0)
{
ffplay_info("len = %d, audio_buf_index = %d, audio_buf_size = %d\n",len,audio_buf_index,audio_buf_size);
if(audio_buf_index >= audio_buf_size)
{
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
ffplay_info("audio_size = %d\n",audio_size);
if(audio_size < 0)
{
/* If error, output silence */
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
}
else
{
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
{
len1 = len;
}
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
ffplay_info("len = %d\n",len);
}
//ffplay_info("end.\n");
}
static int audio_thread(void *data)
{
audio_context *audio_ctx = (audio_context *)data;
int i, j;
while(!quit_video)
{
if(musicchannel.paused)
{
continue;
}
// Just to be sure: check if all goes well...
for(i = 0; i < MUSIC_NUM_BUFFERS; i++)
{
if(musicchannel.fp_playto[i] > INT_TO_FIX(MUSIC_BUF_SIZE))
{
musicchannel.fp_playto[i] = 0;
}
}
// Need to update?
for(j = 0, i = musicchannel.playing_buffer + 1; j < MUSIC_NUM_BUFFERS; j++, i++)
{
i %= MUSIC_NUM_BUFFERS;
if(musicchannel.fp_playto[i] == 0)
{
// Buffer needs to be filled
if (audio_decode_frame(audio_ctx, (uint8_t*)musicchannel.buf[i], MUSIC_BUF_SIZE * sizeof(short)) < 0)
return 0;
musicchannel.fp_playto[i] = INT_TO_FIX(MUSIC_BUF_SIZE);
if(!musicchannel.active)
{
musicchannel.playing_buffer = i;
musicchannel.active = 1;
}
}
}
// Sleep for 1 ms so that this thread doesn't waste CPU cycles busywaiting
usleep(1000);
}
return 0;
}
static void audio_callback(void *userdata, uint8_t * stream, int len)
{
VideoState *is = (VideoState *) userdata;
int len1, audio_size;
double pts;
if (!is->first) return;
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
/* We have already sent all our data; get more */
audio_size = -1;
if (!is->paused) {
audio_size = audio_decode_frame(is, AUDIO_BUF(is),
AUDIO_BUF_SIZE, &pts);
}
if (audio_size < 0) {
/* If error, output silence */
is->audio_buf_size = 1024;
memset(AUDIO_BUF(is), 0, is->audio_buf_size);
}
else {
audio_size = synchronize_audio(is, (int16_t *) AUDIO_BUF(is),
audio_size, pts);
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *) AUDIO_BUF(is) + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1 = 0;
int audio_size = 0;
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = (MAX_AUDIO_FRAME_SIZE * 3) / 2;
static unsigned int audio_buf_index = 0;
// int bytes_per_sec = 2* aCodecCtx->channels * aCodecCtx->sample_rate;
while(len > 0)
{
if(audio_buf_index >= audio_buf_size)
{
/* We have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, audio_buf, audio_buf_size);
if(audio_size < 0)
{
/* If error, output silence */
audio_buf_size = 1024; // arbitrary?
memset(audio_buf, 0, audio_buf_size);
}
else
{
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback(void *userdata, uint8_t *stream, int len)
{
AVCodecContext *aCodecCtx;
aCodecCtx = (AVCodecContext*)userdata;
int len1, audio_size;
static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
AVPacket *pkt = av_mallocz(sizeof(AVPacket));
AVPacket *pkt_temp = av_mallocz(sizeof(AVPacket));
AVFrame *frame = NULL;
while(len > 0)
{
if(audio_buf_index >= audio_buf_size)
{
/*we have already sent all our data; get more */
audio_size = audio_decode_frame(aCodecCtx, pkt, pkt_temp, frame, audio_buf);
if(audio_size < 0)
{
/*if error, output silence*/
audio_buf_size = 1024;
memset(audio_buf, 0, audio_buf_size);
}
else
{
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t*)audio_buf + audio_buf_index, len1);
len -= len1;
stream += len1;
audio_buf_index += len1;
}
}
void audio_callback(void *userdata, Uint8 *stream, int len) {
AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
int len1, audio_size;
static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
static unsigned int audio_buf_size = 0;
static unsigned int audio_buf_index = 0;
printf("audio_callback-len:%d\n", len);
while(len > 0) {
if(audio_buf_index >= audio_buf_size) {
// We have already sent all our data; get more
audio_size = audio_decode_frame(aCodecCtx, audio_buf, audio_buf_size);
if(audio_size < 0) {
// If error, output silence
audio_buf_size = 1024; //
memset(audio_buf, 0, audio_buf_size);
} else {
audio_buf_size = audio_size;
}
audio_buf_index = 0;
}
len1 = audio_buf_size - audio_buf_index;
if(len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
len -= len1; // if len > 0, it means decode something again
stream += len1; // update stream location(destination buffer)
audio_buf_index += len1; // update audio_buf location(source buffer)
}
}
static int decode_one_frame(struct GroovePlaylist *playlist, struct GrooveFile *file) {
struct GroovePlaylistPrivate *p = (struct GroovePlaylistPrivate *) playlist;
struct GrooveFilePrivate *f = (struct GrooveFilePrivate *) file;
AVPacket *pkt = &f->audio_pkt;
// might need to rebuild the filter graph if certain things changed
if (maybe_init_filter_graph(playlist, file) < 0)
return -1;
// abort_request is set if we are destroying the file
if (f->abort_request)
return -1;
// handle pause requests
// only read p->paused once so that we don't need a mutex
int paused = p->paused;
if (paused != p->last_paused) {
p->last_paused = paused;
if (paused) {
av_read_pause(f->ic);
} else {
av_read_play(f->ic);
}
}
// handle seek requests
pthread_mutex_lock(&f->seek_mutex);
if (f->seek_pos >= 0) {
if (av_seek_frame(f->ic, f->audio_stream_index, f->seek_pos, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "%s: error while seeking\n", f->ic->filename);
} else if (f->seek_flush) {
every_sink_flush(playlist);
}
avcodec_flush_buffers(f->audio_st->codec);
f->seek_pos = -1;
f->eof = 0;
}
pthread_mutex_unlock(&f->seek_mutex);
if (f->eof) {
if (f->audio_st->codec->codec->capabilities & CODEC_CAP_DELAY) {
av_init_packet(pkt);
pkt->data = NULL;
pkt->size = 0;
pkt->stream_index = f->audio_stream_index;
if (audio_decode_frame(playlist, file) > 0) {
// keep flushing
return 0;
}
}
// this file is complete. move on
return -1;
}
int err = av_read_frame(f->ic, pkt);
if (err < 0) {
// treat all errors as EOF, but log non-EOF errors.
if (err != AVERROR_EOF) {
av_log(NULL, AV_LOG_WARNING, "error reading frames\n");
}
f->eof = 1;
return 0;
}
if (pkt->stream_index != f->audio_stream_index) {
// we're only interested in the One True Audio Stream
av_free_packet(pkt);
return 0;
}
audio_decode_frame(playlist, file);
av_free_packet(pkt);
return 0;
}
void *audio_thread(void *arg) {
JNIEnv* env;
if((*g_jvm)->AttachCurrentThread(g_jvm, (void**)&env, NULL) != JNI_OK)
{
LOGE("%s: AttachCurrentThread() failed", __FUNCTION__);
return ((void *)-1);;
}
VideoState *is = (VideoState*)arg;
int remain, audio_size;//remain 解码出的音频缓冲区剩余的数据长度
int pcmBufferLen;//音频数据写入的缓冲区的长度
jclass audio_track_cls = (*env)->FindClass(env,"android/media/AudioTrack");
jmethodID min_buff_size_id = (*env)->GetStaticMethodID(
env,
audio_track_cls,
"getMinBufferSize",
"(III)I");
int buffer_size = (*env)->CallStaticIntMethod(env,audio_track_cls,min_buff_size_id, frequency,
12, /*CHANNEL_IN_STEREO*/
2); /*ENCODING_PCM_16BIT*/
LOGI(10,"buffer_size=%i",buffer_size);
pcmBufferLen = AVCODEC_MAX_AUDIO_FRAME_SIZE * 3/2;
jbyteArray buffer = (*env)->NewByteArray(env,pcmBufferLen);
jmethodID constructor_id = (*env)->GetMethodID(env,audio_track_cls, "<init>",
"(IIIIII)V");
jobject audio_track = (*env)->NewObject(env,audio_track_cls,
constructor_id,
3, /*AudioManager.STREAM_MUSIC*/
frequency, /*sampleRateInHz*/
12, /*CHANNEL_IN_STEREO*/
2, /*ENCODING_PCM_16BIT*/
buffer_size*10, /*bufferSizeInBytes*/
1 /*AudioTrack.MODE_STREAM*/
);
//setvolume
jmethodID setStereoVolume = (*env)->GetMethodID(env,audio_track_cls,"setStereoVolume","(FF)I");
(*env)->CallIntMethod(env,audio_track,setStereoVolume,1.0,1.0);
//play
jmethodID method_play = (*env)->GetMethodID(env,audio_track_cls, "play",
"()V");
(*env)->CallVoidMethod(env,audio_track, method_play);
//write
jmethodID method_write = (*env)->GetMethodID(env,audio_track_cls,"write","([BII)I");
//release
jmethodID method_release = (*env)->GetMethodID(env,audio_track_cls,"release","()V");
//double ref_clock, sync_threshold, diff;
double pts;
LOGI(1, "pcmBufferLen = %d, AVCODEC_MAX_AUDIO_FRAME_SIZE = %d", pcmBufferLen, AVCODEC_MAX_AUDIO_FRAME_SIZE);
while(!is->quit) {
if(is->audio_buf_index >= is->audio_buf_size) {//audio_buf中的数据已经转移完毕了
audio_size = audio_decode_frame(is, is->audio_buf, sizeof(is->audio_buf), &pts);
if (audio_size <= 0) {
is->audio_buf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 3/2;
memset(is->audio_buf, 0, is->audio_buf_size);
}else {
audio_size = synchronize_audio(is, (int16_t *)is->audio_buf,audio_size,pts);
is->audio_buf_size = audio_size;
//fwrite(is->audio_buf, 1, audio_size, fp);
}
//每次解码出音频之后，就把音频的索引audio_buf_index值0 从头开始索引
is->audio_buf_index = 0;
}
//剩余的数据长度超过音频数据写入的缓冲区的长度
remain = is->audio_buf_size - is->audio_buf_index;
if(remain > pcmBufferLen) {
remain = pcmBufferLen;
}
(*env)->SetByteArrayRegion(env,buffer, 0, remain, (jbyte *)is->audio_buf);
(*env)->CallIntMethod(env,audio_track,method_write,buffer,0,remain);
is->audio_buf_index += remain;
}
(*env)->CallVoidMethod(env,audio_track, method_release);
if(debug) LOGI(1, "### decode audio thread exit.");
if((*g_jvm)->DetachCurrentThread(g_jvm) != JNI_OK) {
LOGE(1,"### detach audio thread error");
}
pthread_exit(0);
((void *)0);
}
