int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
if(pts == INT64_MIN)
return s->outpts;
if (s->firstpts == AV_NOPTS_VALUE)
s->outpts = s->firstpts = pts;
if(s->min_compensation >= FLT_MAX) {
return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
} else {
int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
if(fabs(fdelta) > s->min_compensation) {
if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
int ret;
if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
if(ret<0){
av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
}
} else if(s->soft_compensation_duration && s->max_soft_compensation) {
int duration = s->out_sample_rate * s->soft_compensation_duration;
double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
swr_set_compensation(s, comp, duration);
}
}
return s->outpts;
}
}
static int audio_decode_frame(VideoState *is, double *pts_ptr)
{
int len1, len2, decoded_data_size;
AVPacket *pkt = &is->audio_pkt;
int got_frame = 0;
int64_t dec_channel_layout;
int wanted_nb_samples, resampled_data_size, n;
double pts;
for (;;) {
while (is->audio_pkt_size > 0) {
if (is->isPause == true) //判断暂停
{
SDL_Delay(10);
continue;
}
if (!is->audio_frame) {
if (!(is->audio_frame = av_frame_alloc())) {
return AVERROR(ENOMEM);
}
} else
{
// avcodec_get_frame_default(is->audio_frame);
memset(is->audio_frame, 0, sizeof(AVFrame));
av_frame_unref(is->audio_frame);
}
len1 = avcodec_decode_audio4(is->audio_st->codec, is->audio_frame,
&got_frame, pkt);
if (len1 < 0) {
// error, skip the frame
is->audio_pkt_size = 0;
break;
}
is->audio_pkt_data += len1;
is->audio_pkt_size -= len1;
if (!got_frame)
continue;
/* 计算解码出来的桢需要的缓冲大小 */
decoded_data_size = av_samples_get_buffer_size(NULL,
is->audio_frame->channels, is->audio_frame->nb_samples,
(AVSampleFormat)is->audio_frame->format, 1);
dec_channel_layout =
(is->audio_frame->channel_layout
&& is->audio_frame->channels
== av_get_channel_layout_nb_channels(
is->audio_frame->channel_layout)) ?
is->audio_frame->channel_layout :
av_get_default_channel_layout(
is->audio_frame->channels);
wanted_nb_samples = is->audio_frame->nb_samples;
if (is->audio_frame->format != is->audio_src_fmt
|| dec_channel_layout != is->audio_src_channel_layout
|| is->audio_frame->sample_rate != is->audio_src_freq
|| (wanted_nb_samples != is->audio_frame->nb_samples
&& !is->swr_ctx)) {
if (is->swr_ctx)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt_channel_layout, (AVSampleFormat)is->audio_tgt_fmt,
is->audio_tgt_freq, dec_channel_layout,
(AVSampleFormat)is->audio_frame->format, is->audio_frame->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
//fprintf(stderr,"swr_init() failed\n");
break;
}
is->audio_src_channel_layout = dec_channel_layout;
is->audio_src_channels = is->audio_st->codec->channels;
is->audio_src_freq = is->audio_st->codec->sample_rate;
is->audio_src_fmt = is->audio_st->codec->sample_fmt;
}
/* 这里我们可以对采样数进行调整，增加或者减少，一般可以用来做声画同步 */
if (is->swr_ctx) {
const uint8_t **in =
(const uint8_t **) is->audio_frame->extended_data;
uint8_t *out[] = { is->audio_buf2 };
if (wanted_nb_samples != is->audio_frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx,
(wanted_nb_samples - is->audio_frame->nb_samples)
* is->audio_tgt_freq
/ is->audio_frame->sample_rate,
wanted_nb_samples * is->audio_tgt_freq
/ is->audio_frame->sample_rate) < 0) {
//fprintf(stderr,"swr_set_compensation() failed\n");
break;
}
}
len2 = swr_convert(is->swr_ctx, out,
sizeof(is->audio_buf2) / is->audio_tgt_channels
/ av_get_bytes_per_sample(is->audio_tgt_fmt),
in, is->audio_frame->nb_samples);
if (len2 < 0) {
//fprintf(stderr,"swr_convert() failed\n");
break;
}
if (len2
== sizeof(is->audio_buf2) / is->audio_tgt_channels
/ av_get_bytes_per_sample(is->audio_tgt_fmt)) {
//fprintf(stderr,"warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx);
}
is->audio_buf = is->audio_buf2;
resampled_data_size = len2 * is->audio_tgt_channels
* av_get_bytes_per_sample(is->audio_tgt_fmt);
} else {
resampled_data_size = decoded_data_size;
is->audio_buf = is->audio_frame->data[0];
}
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * is->audio_st->codec->channels;
is->audio_clock += (double) resampled_data_size
/ (double) (n * is->audio_st->codec->sample_rate);
if (is->seek_flag_audio)
{
//发生了跳转 则跳过关键帧到目的时间的这几帧
if (is->audio_clock < is->seek_time)
{
break;
}
else
{
is->seek_flag_audio = 0;
}
}
// We have data, return it and come back for more later
return resampled_data_size;
}
if (pkt->data)
av_free_packet(pkt);
memset(pkt, 0, sizeof(*pkt));
if (is->quit)
return -1;
if (is->isPause == true) //判断暂停
{
SDL_Delay(10);
continue;
}
if (packet_queue_get(&is->audioq, pkt, 0) <= 0)
{
return -1;
}
//收到这个数据 说明刚刚执行过跳转 现在需要把解码器的数据 清除一下
if(strcmp((char*)pkt->data,FLUSH_DATA) == 0)
{
avcodec_flush_buffers(is->audio_st->codec);
av_free_packet(pkt);
continue;
}
is->audio_pkt_data = pkt->data;
is->audio_pkt_size = pkt->size;
/* if update, update the audio clock w/pts */
if (pkt->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base) * pkt->pts;
}
}
return 0;
}
int audio_decode_frame(FFmpegState *st) {
int len1, len2, decoded_data_size;
AVPacket *pkt = &st->audio_pkt;
int got_frame = 0;
int64_t dec_channel_layout;
int wanted_nb_samples, resampled_data_size;
for (;;) {
while (st->audio_pkt_size > 0) {
if (!st->aFrame) {
if (!(st->aFrame = av_frame_alloc())) {
return AVERROR(ENOMEM);
}
}
//else
// avcodec_get_frame_defaults(st->aFrame);
len1 = avcodec_decode_audio4(st->audio_st->codec, st->aFrame, &got_frame, pkt);
if (len1 < 0) {
// error, skip the frame
st->audio_pkt_size = 0;
break;
}
st->audio_pkt_data += len1;
st->audio_pkt_size -= len1;
if (!got_frame)
continue;
decoded_data_size = av_samples_get_buffer_size(NULL,
st->aFrame->channels,
st->aFrame->nb_samples,
(AVSampleFormat)st->aFrame->format, 1);
dec_channel_layout = (st->aFrame->channel_layout && st->aFrame->channels
== av_get_channel_layout_nb_channels(st->aFrame->channel_layout))
? st->aFrame->channel_layout
: av_get_default_channel_layout(st->aFrame->channels);
wanted_nb_samples = st->aFrame->nb_samples;
//fprintf(stderr, "wanted_nb_samples = %d\n", wanted_nb_samples);
if (st->swr) {
// const uint8_t *in[] = { is->audio_frame->data[0] };
const uint8_t **in = (const uint8_t **)st->aFrame->extended_data;
uint8_t *out[] = { st->audio_buf2 };
if (wanted_nb_samples != st->aFrame->nb_samples) {
if (swr_set_compensation(st->swr, (wanted_nb_samples - st->aFrame->nb_samples)
* st->audio_tgt_freq / st->aFrame->sample_rate,
wanted_nb_samples * st->audio_tgt_freq / st->aFrame->sample_rate) < 0) {
fprintf(stderr, "swr_set_compensation() failed\n");
break;
}
}
// SWR Convert
len2 = swr_convert(st->swr, out,
sizeof(st->audio_buf2)
/ st->audio_tgt_channels
/ av_get_bytes_per_sample(st->audio_tgt_fmt),
in, st->aFrame->nb_samples);
if (len2 < 0) {
fprintf(stderr, "swr_convert() failed\n");
break;
}
if (len2 == sizeof(st->audio_buf2) / st->audio_tgt_channels / av_get_bytes_per_sample(st->audio_tgt_fmt)) {
fprintf(stderr, "warning: audio buffer is probably too small\n");
swr_init(st->swr);
}
st->audio_buf = st->audio_buf2;
resampled_data_size = len2 * st->audio_tgt_channels * av_get_bytes_per_sample(st->audio_tgt_fmt);
}
else {
resampled_data_size = decoded_data_size;
st->audio_buf = st->aFrame->data[0];
}
// Update the wave buffer
if (st->wave) {
update_wave_buffer(st->audio_buf, resampled_data_size);
}
// We have data, return it and come back for more later
return resampled_data_size;
}
// Read packet from queue
if (pkt->data) av_free_packet(pkt);
memset(pkt, 0, sizeof(*pkt));
if (st->quit) return -1;
if (packet_queue_get(&st->audioq, pkt, 1) < 0) return -1;
st->audio_pkt_data = pkt->data;
st->audio_pkt_size = pkt->size;
}
}
