fftools/ffmpeg: rework audio-decode timestamp handling

Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.

New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.

Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.

1 files changed, 81 insertions, 17 deletions

diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c
index 8829a163e0..8dcc70e879 100644
--- a/fftools/ffmpeg.c
+++ b/fftools/ffmpeg.c
@@ -881,6 +881,85 @@ static int send_frame_to_filters(InputStream *ist, AVFrame *decoded_frame)
     return ret;
 }
 
+static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame)
+{
+    const int prev = ist->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)
+        goto finish;
+
+    gcd  = av_gcd(prev, sr);
+
+    if (prev / gcd >= INT_MAX / sr) {
+        av_log(ist, AV_LOG_WARNING,
+               "Audio timestamps cannot be represented exactly after "
+               "sample rate change: %d -> %d\n", prev, sr);
+
+        // LCM of 192000, 44100, allows to represent all common samplerates
+        tb_new = (AVRational){ 1, 28224000 };
+    } else
+        tb_new = (AVRational){ 1, prev / gcd * sr };
+
+    // keep the frame timebase if it is strictly better than
+    // the samplerate-defined one
+    if (frame->time_base.num == 1 && frame->time_base.den > tb_new.den &&
+        !(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);
+
+    ist->last_frame_tb          = tb_new;
+    ist->last_frame_sample_rate = frame->sample_rate;
+
+finish:
+    return ist->last_frame_tb;
+}
+
+static void audio_ts_process(InputStream *ist, AVFrame *frame)
+{
+    AVRational tb_filter = (AVRational){1, frame->sample_rate};
+    AVRational tb;
+    int64_t pts_pred;
+
+    // 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;
+
+    if (frame->pts == AV_NOPTS_VALUE) {
+        frame->pts = pts_pred;
+        frame->time_base = tb;
+    } else if (ist->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;
+    }
+
+    frame->pts = av_rescale_delta(frame->time_base, frame->pts,
+                                  tb, frame->nb_samples,
+                                  &ist->filter_in_rescale_delta_last, tb);
+
+    ist->last_frame_pts          = frame->pts;
+    ist->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);
+    frame->duration  = frame->nb_samples;
+    frame->time_base = tb_filter;
+}
+
 static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output,
                         int *decode_failed)
 {
@@ -910,23 +989,7 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output,
     ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
                      decoded_frame->sample_rate;
 
-    if (decoded_frame->pts == AV_NOPTS_VALUE) {
-        decoded_frame->pts = ist->dts;
-        decoded_frame->time_base = AV_TIME_BASE_Q;
-    }
-    if (pkt && pkt->duration && ist->prev_pkt_pts != AV_NOPTS_VALUE &&
-        pkt->pts != AV_NOPTS_VALUE && pkt->pts - ist->prev_pkt_pts > pkt->duration)
-        ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
-    if (pkt)
-        ist->prev_pkt_pts = pkt->pts;
-    if (decoded_frame->pts != AV_NOPTS_VALUE) {
-        AVRational tb_filter = (AVRational){1, decoded_frame->sample_rate};
-        decoded_frame->pts = av_rescale_delta(decoded_frame->time_base, decoded_frame->pts,
-                                              tb_filter, decoded_frame->nb_samples,
-                                              &ist->filter_in_rescale_delta_last,
-                                              tb_filter);
-        decoded_frame->time_base = tb_filter;
-    }
+    audio_ts_process(ist, decoded_frame);
 
     ist->nb_samples = decoded_frame->nb_samples;
     err = send_frame_to_filters(ist, decoded_frame);
@@ -1076,6 +1139,7 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output, int64_
     // update timestamp history
     ist->last_frame_duration_est = video_duration_estimate(ist, decoded_frame);
     ist->last_frame_pts          = decoded_frame->pts;
+    ist->last_frame_tb           = decoded_frame->time_base;
 
     if (debug_ts) {
         av_log(ist, AV_LOG_INFO,