@@ -352,16 +352,6 @@ typedef struct InputStream {
AVRational framerate_guessed;
- // pts/estimated duration of the last decoded frame
- // * in decoder timebase for video,
- // * in last_frame_tb (may change during decoding) for audio
- int64_t last_frame_pts;
- int64_t last_frame_duration_est;
- AVRational last_frame_tb;
- int last_frame_sample_rate;
-
- int64_t filter_in_rescale_delta_last;
-
int64_t nb_samples; /* number of samples in the last decoded audio frame before looping */
AVDictionary *decoder_opts;
@@ -34,6 +34,15 @@
struct Decoder {
AVFrame *frame;
AVPacket *pkt;
+
+ // pts/estimated duration of the last decoded frame
+ // * in decoder timebase for video,
+ // * in last_frame_tb (may change during decoding) for audio
+ int64_t last_frame_pts;
+ int64_t last_frame_duration_est;
+ AVRational last_frame_tb;
+ int64_t last_filter_in_rescale_delta;
+ int last_frame_sample_rate;
};
void dec_free(Decoder **pdec)
@@ -67,6 +76,9 @@ static int dec_alloc(Decoder **pdec)
if (!dec->pkt)
goto fail;
+ dec->last_filter_in_rescale_delta = AV_NOPTS_VALUE;
+ dec->last_frame_pts = AV_NOPTS_VALUE;
+ dec->last_frame_tb = (AVRational){ 1, 1 };
*pdec = dec;
@@ -94,21 +106,22 @@ static int send_frame_to_filters(InputStream *ist, AVFrame *decoded_frame)
return ret;
}
-static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame)
+static AVRational audio_samplerate_update(void *logctx, Decoder *d,
+ const AVFrame *frame)
{
- const int prev = ist->last_frame_tb.den;
+ const int prev = d->last_frame_tb.den;
const int sr = frame->sample_rate;
AVRational tb_new;
int64_t gcd;
- if (frame->sample_rate == ist->last_frame_sample_rate)
+ if (frame->sample_rate == d->last_frame_sample_rate)
goto finish;
gcd = av_gcd(prev, sr);
if (prev / gcd >= INT_MAX / sr) {
- av_log(ist, AV_LOG_WARNING,
+ av_log(logctx, AV_LOG_WARNING,
"Audio timestamps cannot be represented exactly after "
"sample rate change: %d -> %d\n", prev, sr);
@@ -123,20 +136,20 @@ static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame
!(frame->time_base.den % tb_new.den))
tb_new = frame->time_base;
- if (ist->last_frame_pts != AV_NOPTS_VALUE)
- ist->last_frame_pts = av_rescale_q(ist->last_frame_pts,
- ist->last_frame_tb, tb_new);
- ist->last_frame_duration_est = av_rescale_q(ist->last_frame_duration_est,
- ist->last_frame_tb, tb_new);
+ if (d->last_frame_pts != AV_NOPTS_VALUE)
+ d->last_frame_pts = av_rescale_q(d->last_frame_pts,
+ d->last_frame_tb, tb_new);
+ d->last_frame_duration_est = av_rescale_q(d->last_frame_duration_est,
+ d->last_frame_tb, tb_new);
- ist->last_frame_tb = tb_new;
- ist->last_frame_sample_rate = frame->sample_rate;
+ d->last_frame_tb = tb_new;
+ d->last_frame_sample_rate = frame->sample_rate;
finish:
- return ist->last_frame_tb;
+ return d->last_frame_tb;
}
-static void audio_ts_process(InputStream *ist, AVFrame *frame)
+static void audio_ts_process(void *logctx, Decoder *d, AVFrame *frame)
{
AVRational tb_filter = (AVRational){1, frame->sample_rate};
AVRational tb;
@@ -145,27 +158,27 @@ static void audio_ts_process(InputStream *ist, AVFrame *frame)
// on samplerate change, choose a new internal timebase for timestamp
// generation that can represent timestamps from all the samplerates
// seen so far
- tb = audio_samplerate_update(ist, frame);
- pts_pred = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 :
- ist->last_frame_pts + ist->last_frame_duration_est;
+ tb = audio_samplerate_update(logctx, d, frame);
+ pts_pred = d->last_frame_pts == AV_NOPTS_VALUE ? 0 :
+ d->last_frame_pts + d->last_frame_duration_est;
if (frame->pts == AV_NOPTS_VALUE) {
frame->pts = pts_pred;
frame->time_base = tb;
- } else if (ist->last_frame_pts != AV_NOPTS_VALUE &&
+ } else if (d->last_frame_pts != AV_NOPTS_VALUE &&
frame->pts > av_rescale_q_rnd(pts_pred, tb, frame->time_base,
AV_ROUND_UP)) {
// there was a gap in timestamps, reset conversion state
- ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
+ d->last_filter_in_rescale_delta = AV_NOPTS_VALUE;
}
frame->pts = av_rescale_delta(frame->time_base, frame->pts,
tb, frame->nb_samples,
- &ist->filter_in_rescale_delta_last, tb);
+ &d->last_filter_in_rescale_delta, tb);
- ist->last_frame_pts = frame->pts;
- ist->last_frame_duration_est = av_rescale_q(frame->nb_samples,
- tb_filter, tb);
+ d->last_frame_pts = frame->pts;
+ d->last_frame_duration_est = av_rescale_q(frame->nb_samples,
+ tb_filter, tb);
// finally convert to filtering timebase
frame->pts = av_rescale_q(frame->pts, tb, tb_filter);
@@ -175,6 +188,7 @@ static void audio_ts_process(InputStream *ist, AVFrame *frame)
static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *frame)
{
+ const Decoder *d = ist->decoder;
const InputFile *ifile = input_files[ist->file_index];
int64_t codec_duration = 0;
@@ -202,9 +216,9 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr
// when timestamps are available, repeat last frame's actual duration
// (i.e. pts difference between this and last frame)
- if (frame->pts != AV_NOPTS_VALUE && ist->last_frame_pts != AV_NOPTS_VALUE &&
- frame->pts > ist->last_frame_pts)
- return frame->pts - ist->last_frame_pts;
+ if (frame->pts != AV_NOPTS_VALUE && d->last_frame_pts != AV_NOPTS_VALUE &&
+ frame->pts > d->last_frame_pts)
+ return frame->pts - d->last_frame_pts;
// try frame/codec duration
if (frame->duration > 0)
@@ -221,11 +235,13 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr
}
// last resort is last frame's estimated duration, and 1
- return FFMAX(ist->last_frame_duration_est, 1);
+ return FFMAX(d->last_frame_duration_est, 1);
}
static int video_frame_process(InputStream *ist, AVFrame *frame)
{
+ Decoder *d = ist->decoder;
+
// The following line may be required in some cases where there is no parser
// or the parser does not has_b_frames correctly
if (ist->par->video_delay < ist->dec_ctx->has_b_frames) {
@@ -273,13 +289,13 @@ static int video_frame_process(InputStream *ist, AVFrame *frame)
// no timestamp available - extrapolate from previous frame duration
if (frame->pts == AV_NOPTS_VALUE)
- frame->pts = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 :
- ist->last_frame_pts + ist->last_frame_duration_est;
+ frame->pts = d->last_frame_pts == AV_NOPTS_VALUE ? 0 :
+ d->last_frame_pts + d->last_frame_duration_est;
// update timestamp history
- ist->last_frame_duration_est = video_duration_estimate(ist, frame);
- ist->last_frame_pts = frame->pts;
- ist->last_frame_tb = frame->time_base;
+ d->last_frame_duration_est = video_duration_estimate(ist, frame);
+ d->last_frame_pts = frame->pts;
+ d->last_frame_tb = frame->time_base;
if (debug_ts) {
av_log(ist, AV_LOG_INFO,
@@ -404,12 +420,13 @@ static int transcode_subtitles(InputStream *ist, const AVPacket *pkt)
static int send_filter_eof(InputStream *ist)
{
+ Decoder *d = ist->decoder;
int i, ret;
for (i = 0; i < ist->nb_filters; i++) {
- int64_t end_pts = ist->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
- ist->last_frame_pts + ist->last_frame_duration_est;
- ret = ifilter_send_eof(ist->filters[i], end_pts, ist->last_frame_tb);
+ int64_t end_pts = d->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
+ d->last_frame_pts + d->last_frame_duration_est;
+ ret = ifilter_send_eof(ist->filters[i], end_pts, d->last_frame_tb);
if (ret < 0)
return ret;
}
@@ -506,7 +523,7 @@ int dec_packet(InputStream *ist, const AVPacket *pkt, int no_eof)
ist->samples_decoded += frame->nb_samples;
ist->nb_samples = frame->nb_samples;
- audio_ts_process(ist, frame);
+ audio_ts_process(ist, ist->decoder, frame);
} else {
ret = video_frame_process(ist, frame);
if (ret < 0) {
@@ -1181,11 +1181,6 @@ static void add_input_streams(const OptionsContext *o, Demuxer *d)
exit_program(1);
}
- ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
-
- ist->last_frame_pts = AV_NOPTS_VALUE;
- ist->last_frame_tb = (AVRational){ 1, 1 };
-
ist->dec_ctx = avcodec_alloc_context3(ist->dec);
if (!ist->dec_ctx)
report_and_exit(AVERROR(ENOMEM));