我们这篇文章分为三部分
第一部分解封装音频
static int read_thread(void *arg)
{
......
/* open the streams */
if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) {
stream_component_open(ffp, st_index[AVMEDIA_TYPE_AUDIO]);
} else {
ffp->av_sync_type = AV_SYNC_VIDEO_MASTER;
is->av_sync_type = ffp->av_sync_type;
}
ret = -1;
if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) {
ret = stream_component_open(ffp, st_index[AVMEDIA_TYPE_VIDEO]);
}
if (is->show_mode == SHOW_MODE_NONE)
is->show_mode = ret >= 0 ? SHOW_MODE_VIDEO : SHOW_MODE_RDFT;
if (st_index[AVMEDIA_TYPE_SUBTITLE] >= 0) {
stream_component_open(ffp, st_index[AVMEDIA_TYPE_SUBTITLE]);
}
......
for (;;) {
........
从流中取出相应的packet，并保存到相应queue中，便于解码的时候使用
ret = av_read_frame(ic, pkt);
if (ret < 0) {
int pb_eof = 0;
int pb_error = 0;
if ((ret == AVERROR_EOF || avio_feof(ic->pb)) && !is->eof) {
ffp_check_buffering_l(ffp);
pb_eof = 1;
// check error later
}
if (ic->pb && ic->pb->error) {
pb_eof = 1;
pb_error = ic->pb->error;
}
if (ret == AVERROR_EXIT) {
pb_eof = 1;
pb_error = AVERROR_EXIT;
}
if (pb_eof) {
......
if (is->audio_stream >= 0)
packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
......
is->eof = 1;
}
if (pb_error) {
......
if (is->audio_stream >= 0)
packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
......
is->eof = 1;
ffp->error = pb_error;
av_log(ffp, AV_LOG_ERROR, "av_read_frame error: %s\n", ffp_get_error_string(ffp->error));
// break;
} else {
ffp->error = 0;
}
if (is->eof) {
ffp_toggle_buffering(ffp, 0);
SDL_Delay(100);
}
SDL_LockMutex(wait_mutex);
SDL_CondWaitTimeout(is->continue_read_thread, wait_mutex, 10);
SDL_UnlockMutex(wait_mutex);
ffp_statistic_l(ffp);
continue;
} else {
is->eof = 0;
}
if (pkt->flags & AV_PKT_FLAG_DISCONTINUITY) {
if (is->audio_stream >= 0) {
packet_queue_put(&is->audioq, &flush_pkt);
}
......
}
/* check if packet is in play range specified by user, then queue, otherwise discard */
stream_start_time = ic->streams[pkt->stream_index]->start_time;
pkt_ts = pkt->pts == AV_NOPTS_VALUE ? pkt->dts : pkt->pts;
pkt_in_play_range = ffp->duration == AV_NOPTS_VALUE ||
(pkt_ts - (stream_start_time != AV_NOPTS_VALUE ? stream_start_time : 0)) *
av_q2d(ic->streams[pkt->stream_index]->time_base) -
(double)(ffp->start_time != AV_NOPTS_VALUE ? ffp->start_time : 0) / 1000000
<= ((double)ffp->duration / 1000000);
if (pkt->stream_index == is->audio_stream && pkt_in_play_range) {
packet_queue_put(&is->audioq, pkt);
}
.......
}
}从上面的代码我们可以看到，解封装出来后的AvPacket放入了is->audioq
第二部分音频解码
通过这篇文章 我们知道通过av_find_best_stream，找到音频流的索引，
if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) {
stream_component_open(ffp, st_index[AVMEDIA_TYPE_AUDIO]);
}* open a given stream. Return 0 if OK */
static int stream_component_open(FFPlayer *ffp, int stream_index)
{
VideoState *is = ffp->is;
AVFormatContext *ic = is->ic;
AVCodecContext *avctx;
AVCodec *codec = NULL;
const char *forced_codec_name = NULL;
AVDictionary *opts = NULL;
AVDictionaryEntry *t = NULL;
int sample_rate, nb_channels;
int64_t channel_layout;
int ret = 0;
int stream_lowres = ffp->lowres;
if (stream_index < 0 || stream_index >= ic->nb_streams)
return -1;
//创建解码器上下文
avctx = avcodec_alloc_context3(NULL);
if (!avctx)
return AVERROR(ENOMEM);
//为解码器复制流的参数
ret = avcodec_parameters_to_context(avctx, ic->streams[stream_index]->codecpar);
if (ret < 0)
goto fail;
//解码器设置时间基
av_codec_set_pkt_timebase(avctx, ic->streams[stream_index]->time_base);
//根据code_id 找到解码器
codec = avcodec_find_decoder(avctx->codec_id);
//我们可以在VideoState初始化的时候，设置专门的解码器名字，这里会根据名字找到解码器
switch (avctx->codec_type) {
case AVMEDIA_TYPE_AUDIO : is->last_audio_stream = stream_index; forced_codec_name = ffp->audio_codec_name; break;
case AVMEDIA_TYPE_SUBTITLE: is->last_subtitle_stream = stream_index; forced_codec_name = ffp->subtitle_codec_name; break;
case AVMEDIA_TYPE_VIDEO : is->last_video_stream = stream_index; forced_codec_name = ffp->video_codec_name; break;
default: break;
}
if (forced_codec_name)
codec = avcodec_find_decoder_by_name(forced_codec_name);
if (!codec) {
if (forced_codec_name) av_log(NULL, AV_LOG_WARNING,
"No codec could be found with name '%s'\n", forced_codec_name);
else av_log(NULL, AV_LOG_WARNING,
"No codec could be found with id %d\n", avctx->codec_id);
ret = AVERROR(EINVAL);
goto fail;
}
//解码器id设置到解码器上下文当中
avctx->codec_id = codec->id;
if(stream_lowres > av_codec_get_max_lowres(codec)){
av_log(avctx, AV_LOG_WARNING, "The maximum value for lowres supported by the decoder is %d\n",
av_codec_get_max_lowres(codec));
stream_lowres = av_codec_get_max_lowres(codec);
}
av_codec_set_lowres(avctx, stream_lowres);
#if FF_API_EMU_EDGE
if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
#endif
if (ffp->fast)
avctx->flags2 |= AV_CODEC_FLAG2_FAST;
#if FF_API_EMU_EDGE
if(codec->capabilities & AV_CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
#endif
//打开解码器
opts = filter_codec_opts(ffp->codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
if (!av_dict_get(opts, "threads", NULL, 0))
av_dict_set(&opts, "threads", "auto", 0);
if (stream_lowres)
av_dict_set_int(&opts, "lowres", stream_lowres, 0);
if (avctx->codec_type == AVMEDIA_TYPE_VIDEO || avctx->codec_type == AVMEDIA_TYPE_AUDIO)
av_dict_set(&opts, "refcounted_frames", "1", 0);
if ((ret = avcodec_open2(avctx, codec, &opts)) < 0) {
goto fail;
}
if ((t = av_dict_get(opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
av_log(NULL, AV_LOG_ERROR, "Option %s not found.\n", t->key);
#ifdef FFP_MERGE
ret = AVERROR_OPTION_NOT_FOUND;
goto fail;
#endif
}
is->eof = 0;
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
switch (avctx->codec_type) {
case AVMEDIA_TYPE_AUDIO:
#if CONFIG_AVFILTER
{
AVFilterContext *sink;
is->audio_filter_src.freq = avctx->sample_rate; //采样率
is->audio_filter_src.channels = avctx->channels; //声道数
is->audio_filter_src.channel_layout = get_valid_channel_layout(avctx->channel_layout, avctx->channels); //声道布局
is->audio_filter_src.fmt = avctx->sample_fmt; //位深
SDL_LockMutex(ffp->af_mutex);
if ((ret = configure_audio_filters(ffp, ffp->afilters, 0)) < 0) {
SDL_UnlockMutex(ffp->af_mutex);
goto fail;
}
ffp->af_changed = 0;
SDL_UnlockMutex(ffp->af_mutex);
sink = is->out_audio_filter;
sample_rate = av_buffersink_get_sample_rate(sink);
nb_channels = av_buffersink_get_channels(sink);
channel_layout = av_buffersink_get_channel_layout(sink);
}
#else
sample_rate = avctx->sample_rate;
nb_channels = avctx->channels;
channel_layout = avctx->channel_layout;
#endif
/* prepare audio output */
if ((ret = audio_open(ffp, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
goto fail;
ffp_set_audio_codec_info(ffp, AVCODEC_MODULE_NAME, avcodec_get_name(avctx->codec_id));
is->audio_hw_buf_size = ret;
is->audio_src = is->audio_tgt;
is->audio_buf_size = 0;
is->audio_buf_index = 0;
/* init averaging filter */
is->audio_diff_avg_coef = exp(log(0.01) / AUDIO_DIFF_AVG_NB);
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio FIFO fullness,
we correct audio sync only if larger than this threshold */
is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec;
is->audio_stream = stream_index;
is->audio_st = ic->streams[stream_index];
decoder_init(&is->auddec, avctx, &is->audioq, is->continue_read_thread);
if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek) {
is->auddec.start_pts = is->audio_st->start_time;
is->auddec.start_pts_tb = is->audio_st->time_base;
}
if ((ret = decoder_start(&is->auddec, audio_thread, ffp, "ff_audio_dec")) < 0)
goto out;
SDL_AoutPauseAudio(ffp->aout, 0);
break;
......
}
goto out;
fail:
avcodec_free_context(&avctx);
out:
av_dict_free(&opts);
return ret;
}我们可以看到最后通过decoder_start，走到了audio_thread
我们接下来分析audio_thread
static int audio_thread(void *arg)
{
FFPlayer *ffp = arg;
VideoState *is = ffp->is;
AVFrame *frame = av_frame_alloc();
Frame *af;
#if CONFIG_AVFILTER
int last_serial = -1;
int64_t dec_channel_layout;
int reconfigure;
#endif
int got_frame = 0;
AVRational tb;
int ret = 0;
int audio_accurate_seek_fail = 0;
int64_t audio_seek_pos = 0;
double frame_pts = 0;
double audio_clock = 0;
int64_t now = 0;
double samples_duration = 0;
int64_t deviation = 0;
int64_t deviation2 = 0;
int64_t deviation3 = 0;
if (!frame)
return AVERROR(ENOMEM);
do {
ffp_audio_statistic_l(ffp);
//通过decoder_decode_frame来进行解码，解码后的数据放在frame中。frame是帧的概念，在音频中，一帧代表的是几毫秒的音频
if ((got_frame = decoder_decode_frame(ffp, &is->auddec, frame, NULL)) < 0)
goto the_end;
if (got_frame) {
tb = (AVRational){1, frame->sample_rate}; 每次采样的时间刻度
if (ffp->enable_accurate_seek && is->audio_accurate_seek_req && !is->seek_req) {
计算出当前帧相对于第一帧的时间戳
frame_pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
now = av_gettime_relative() / 1000;
if (!isnan(frame_pts)) {
//下面这句话可以这样理解，一个采样需要播放1/frame->sample_rate,那么nb_samples需要多少呢？明显就是nb_samples * (1/sample_rate),进而就是nb_samples/sample_rate
samples_duration = (double) frame->nb_samples / frame->sample_rate;
//相对于当前系统时间，什么时候开始播放
audio_clock = frame_pts + samples_duration;
//精准寻址转化为微妙
is->accurate_seek_aframe_pts = audio_clock * 1000 * 1000;
要调转到的时间点
audio_seek_pos = is->seek_pos;
deviation = llabs((int64_t)(audio_clock * 1000 * 1000) - is->seek_pos);
if ((audio_clock * 1000 * 1000 < is->seek_pos ) || deviation > MAX_DEVIATION) {
if (is->drop_aframe_count == 0) {
SDL_LockMutex(is->accurate_seek_mutex);
if (is->accurate_seek_start_time <= 0 && (is->video_stream < 0 || is->video_accurate_seek_req)) {
is->accurate_seek_start_time = now;
}
SDL_UnlockMutex(is->accurate_seek_mutex);
av_log(NULL, AV_LOG_INFO, "audio accurate_seek start, is->seek_pos=%lld, audio_clock=%lf, is->accurate_seek_start_time = %lld\n", is->seek_pos, audio_clock, is->accurate_seek_start_time);
}
is->drop_aframe_count++;
while (is->video_accurate_seek_req && !is->abort_request) {
int64_t vpts = is->accurate_seek_vframe_pts;
deviation2 = vpts - audio_clock * 1000 * 1000;
deviation3 = vpts - is->seek_pos;
if (deviation2 > -100 * 1000 && deviation3 < 0) {
break;
} else {
av_usleep(20 * 1000);
}
now = av_gettime_relative() / 1000;
if ((now - is->accurate_seek_start_time) > ffp->accurate_seek_timeout) {
break;
}
}
if(!is->video_accurate_seek_req && is->video_stream >= 0 && audio_clock * 1000 * 1000 > is->accurate_seek_vframe_pts) {
audio_accurate_seek_fail = 1;
} else {
now = av_gettime_relative() / 1000;
if ((now - is->accurate_seek_start_time) <= ffp->accurate_seek_timeout) {
av_frame_unref(frame);
continue; // drop some old frame when do accurate seek
} else {
audio_accurate_seek_fail = 1;
}
}
} else {
if (audio_seek_pos == is->seek_pos) {
av_log(NULL, AV_LOG_INFO, "audio accurate_seek is ok, is->drop_aframe_count=%d, audio_clock = %lf\n", is->drop_aframe_count, audio_clock);
is->drop_aframe_count = 0;
SDL_LockMutex(is->accurate_seek_mutex);
is->audio_accurate_seek_req = 0;
SDL_CondSignal(is->video_accurate_seek_cond);
if (audio_seek_pos == is->seek_pos && is->video_accurate_seek_req && !is->abort_request) {
SDL_CondWaitTimeout(is->audio_accurate_seek_cond, is->accurate_seek_mutex, ffp->accurate_seek_timeout);
} else {
ffp_notify_msg2(ffp, FFP_MSG_ACCURATE_SEEK_COMPLETE, (int)(audio_clock * 1000));
}
if (audio_seek_pos != is->seek_pos && !is->abort_request) {
is->audio_accurate_seek_req = 1;
SDL_UnlockMutex(is->accurate_seek_mutex);
av_frame_unref(frame);
continue;
}
SDL_UnlockMutex(is->accurate_seek_mutex);
}
}
} else {
audio_accurate_seek_fail = 1;
}
if (audio_accurate_seek_fail) {
av_log(NULL, AV_LOG_INFO, "audio accurate_seek is error, is->drop_aframe_count=%d, now = %lld, audio_clock = %lf\n", is->drop_aframe_count, now, audio_clock);
is->drop_aframe_count = 0;
SDL_LockMutex(is->accurate_seek_mutex);
is->audio_accurate_seek_req = 0;
SDL_CondSignal(is->video_accurate_seek_cond);
if (is->video_accurate_seek_req && !is->abort_request) {
SDL_CondWaitTimeout(is->audio_accurate_seek_cond, is->accurate_seek_mutex, ffp->accurate_seek_timeout);
} else {
ffp_notify_msg2(ffp, FFP_MSG_ACCURATE_SEEK_COMPLETE, (int)(audio_clock * 1000));
}
SDL_UnlockMutex(is->accurate_seek_mutex);
}
is->accurate_seek_start_time = 0;
audio_accurate_seek_fail = 0;
}
#if CONFIG_AVFILTER
dec_channel_layout = get_valid_channel_layout(frame->channel_layout, frame->channels);
reconfigure =
cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
frame->format, frame->channels) ||
is->audio_filter_src.channel_layout != dec_channel_layout ||
is->audio_filter_src.freq != frame->sample_rate ||
is->auddec.pkt_serial != last_serial ||
ffp->af_changed;
if (reconfigure) {
SDL_LockMutex(ffp->af_mutex);
ffp->af_changed = 0;
char buf1[1024], buf2[1024];
av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
av_log(NULL, AV_LOG_DEBUG,
"Audio frame changed from rate:%d ch:%d fmt:%s layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
is->audio_filter_src.freq, is->audio_filter_src.channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, last_serial,
frame->sample_rate, frame->channels, av_get_sample_fmt_name(frame->format), buf2, is->auddec.pkt_serial);
is->audio_filter_src.fmt = frame->format;
is->audio_filter_src.channels = frame->channels;
is->audio_filter_src.channel_layout = dec_channel_layout;
is->audio_filter_src.freq = frame->sample_rate;
last_serial = is->auddec.pkt_serial;
if ((ret = configure_audio_filters(ffp, ffp->afilters, 1)) < 0) {
SDL_UnlockMutex(ffp->af_mutex);
goto the_end;
}
SDL_UnlockMutex(ffp->af_mutex);
}
if ((ret = av_buffersrc_add_frame(is->in_audio_filter, frame)) < 0)
goto the_end;
while ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, frame, 0)) >= 0) {
tb = av_buffersink_get_time_base(is->out_audio_filter);
#endif
if (!(af = frame_queue_peek_writable(&is->sampq)))
goto the_end;
af->pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
af->pos = frame->pkt_pos;
af->serial = is->auddec.pkt_serial;
af->duration = av_q2d((AVRational){frame->nb_samples, frame->sample_rate});
av_frame_move_ref(af->frame, frame);
frame_queue_push(&is->sampq);
#if CONFIG_AVFILTER
if (is->audioq.serial != is->auddec.pkt_serial)
break;
}
if (ret == AVERROR_EOF)
is->auddec.finished = is->auddec.pkt_serial;
#endif
}
} while (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF);
the_end:
#if CONFIG_AVFILTER
avfilter_graph_free(&is->agraph);
#endif
av_frame_free(&frame);
return ret;
}上面这段代码我们知道解码出来的数据放到了is->sampq。
第三部分音频播放
static int stream_component_open(FFPlayer *ffp, int stream_index){
......
if ((ret = audio_open(ffp, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
goto fail;
......
}audio_hw_params 打开的设备的音频参数
static int audio_open(FFPlayer *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params)
{
FFPlayer *ffp = opaque;
VideoState *is = ffp->is;
SDL_AudioSpec wanted_spec, spec;
const char *env;
static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
#ifdef FFP_MERGE
static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};
#endif
static const int next_sample_rates[] = {0, 44100, 48000};
int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;
env = SDL_getenv("SDL_AUDIO_CHANNELS");
//设置采样率 采样大小（位深）通道数
if (env) {
wanted_nb_channels = atoi(env);
wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
}
if (!wanted_channel_layout || wanted_nb_channels != av_get_channel_layout_nb_channels(wanted_channel_layout)) {
wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
}
wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
wanted_spec.channels = wanted_nb_channels; 通道数
wanted_spec.freq = wanted_sample_rate; 采样率
if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
av_log(NULL, AV_LOG_ERROR, "Invalid sample rate or channel count!\n");
return -1;
}
while (next_sample_rate_idx && next_sample_rates[next_sample_rate_idx] >= wanted_spec.freq)
next_sample_rate_idx--;
wanted_spec.format = AUDIO_S16SYS; 采样大小
wanted_spec.silence = 0;
wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AoutGetAudioPerSecondCallBacks(ffp->aout)));
wanted_spec.callback = sdl_audio_callback;
wanted_spec.userdata = opaque;
while (SDL_AoutOpenAudio(ffp->aout, &wanted_spec, &spec) < 0) {
/* avoid infinity loop on exit. --by bbcallen */
if (is->abort_request)
return -1;
av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s\n",
wanted_spec.channels, wanted_spec.freq, SDL_GetError());
wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
if (!wanted_spec.channels) {
wanted_spec.freq = next_sample_rates[next_sample_rate_idx--];
wanted_spec.channels = wanted_nb_channels;
if (!wanted_spec.freq) {
av_log(NULL, AV_LOG_ERROR,
"No more combinations to try, audio open failed\n");
return -1;
}
}
wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
}
if (spec.format != AUDIO_S16SYS) {
av_log(NULL, AV_LOG_ERROR,
"SDL advised audio format %d is not supported!\n", spec.format);
return -1;
}
if (spec.channels != wanted_spec.channels) {
wanted_channel_layout = av_get_default_channel_layout(spec.channels);
if (!wanted_channel_layout) {
av_log(NULL, AV_LOG_ERROR,
"SDL advised channel count %d is not supported!\n", spec.channels);
return -1;
}
}
audio_hw_params->fmt = AV_SAMPLE_FMT_S16;
audio_hw_params->freq = spec.freq;
audio_hw_params->channel_layout = wanted_channel_layout;
audio_hw_params->channels = spec.channels;
audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
return -1;
}
SDL_AoutSetDefaultLatencySeconds(ffp->aout, ((double)(2 * spec.size)) / audio_hw_params->bytes_per_sec);
return spec.size;
} wanted_spec.callback = sdl_audio_callback;
wanted_spec.userdata = opaque;
可以知道数据是从sdl_audio_callback来的
我们看下sdl_audio_callback
/* prepare a new audio buffer */
static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
//opaque就是上面说的 wanted_spec.userdata = opaque;
FFPlayer *ffp = opaque;
VideoState *is = ffp->is;
int audio_size, len1;
if (!ffp || !is) {
memset(stream, 0, len);
return;
}
ffp->audio_callback_time = av_gettime_relative();
if (ffp->pf_playback_rate_changed) {
ffp->pf_playback_rate_changed = 0;
#if defined(__ANDROID__)
if (!ffp->soundtouch_enable) {
SDL_AoutSetPlaybackRate(ffp->aout, ffp->pf_playback_rate);
}
#else
SDL_AoutSetPlaybackRate(ffp->aout, ffp->pf_playback_rate);
#endif
}
if (ffp->pf_playback_volume_changed) {
ffp->pf_playback_volume_changed = 0;
SDL_AoutSetPlaybackVolume(ffp->aout, ffp->pf_playback_volume);
}
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
audio_size = audio_decode_frame(ffp);
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf = NULL;
is->audio_buf_size = SDL_AUDIO_MIN_BUFFER_SIZE / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
if (is->auddec.pkt_serial != is->audioq.serial) {
is->audio_buf_index = is->audio_buf_size;
memset(stream, 0, len);
// stream += len;
// len = 0;
SDL_AoutFlushAudio(ffp->aout);
break;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len)
len1 = len;
if (!is->muted && is->audio_buf && is->audio_volume == SDL_MIX_MAXVOLUME)
memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
else {
memset(stream, 0, len1);
if (!is->muted && is->audio_buf)
SDL_MixAudio(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1, is->audio_volume);
}
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
/* Let's assume the audio driver that is used by SDL has two periods. */
if (!isnan(is->audio_clock)) {
set_clock_at(&is->audclk, is->audio_clock - (double)(is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec - SDL_AoutGetLatencySeconds(ffp->aout), is->audio_clock_serial, ffp->audio_callback_time / 1000000.0);
sync_clock_to_slave(&is->extclk, &is->audclk);
}
if (!ffp->first_audio_frame_rendered) {
ffp->first_audio_frame_rendered = 1;
ffp_notify_msg1(ffp, FFP_MSG_AUDIO_RENDERING_START);
}
if (is->latest_audio_seek_load_serial == is->audio_clock_serial) {
int latest_audio_seek_load_serial = __atomic_exchange_n(&(is->latest_audio_seek_load_serial), -1, memory_order_seq_cst);
if (latest_audio_seek_load_serial == is->audio_clock_serial) {
if (ffp->av_sync_type == AV_SYNC_AUDIO_MASTER) {
ffp_notify_msg2(ffp, FFP_MSG_AUDIO_SEEK_RENDERING_START, 1);
} else {
ffp_notify_msg2(ffp, FFP_MSG_AUDIO_SEEK_RENDERING_START, 0);
}
}
}
if (ffp->render_wait_start && !ffp->start_on_prepared && is->pause_req) {
while (is->pause_req && !is->abort_request) {
SDL_Delay(20);
}
}
}copy到stream中实现了播放功能
if (!is->muted && is->audio_buf)
SDL_MixAudio(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1, is->audio_volume);本文章为转载内容，我们尊重原作者对文章享有的著作权。如有内容错误或侵权问题，欢迎原作者联系我们进行内容更正或删除文章。
