Work around deadlock when destroying J2KEncoder with a full writer queue (#2784).

[dcpomatic.git] / src / lib / audio_content.cc

diff --git a/src/lib/audio_content.cc b/src/lib/audio_content.cc
index 05c388a3e711812dc3ad3ceef1c39d8b01002edd..16f2bd5f1a250d7188cb1a73740a415d530b2b08 100644 (file)
--- a/src/lib/audio_content.cc
+++ b/src/lib/audio_content.cc
@@ -22,6 +22,7 @@
 #include "audio_content.h"
 #include "compose.hpp"
 #include "config.h"
+#include "constants.h"
 #include "exceptions.h"
 #include "film.h"
 #include "frame_rate_change.h"
@@ -191,7 +192,6 @@ AudioContent::mapping () const
        merged.make_zero ();
 
        int c = 0;
-       int s = 0;
        for (auto i: streams()) {
                auto mapping = i->mapping ();
                for (int j = 0; j < mapping.input_channels(); ++j) {
@@ -202,7 +202,6 @@ AudioContent::mapping () const
                        }
                        ++c;
                }
-               ++s;
        }
 
        return merged;
@@ -312,7 +311,10 @@ AudioContent::add_properties (shared_ptr<const Film> film, list<UserProperty>& p
 
        if (stream) {
                p.push_back (UserProperty(UserProperty::AUDIO, _("Channels"), stream->channels()));
-               p.push_back (UserProperty(UserProperty::AUDIO, _("Content audio sample rate"), stream->frame_rate(), _("Hz")));
+               p.push_back (UserProperty(UserProperty::AUDIO, _("Content sample rate"), stream->frame_rate(), _("Hz")));
+               if (auto bits = stream->bit_depth()) {
+                       p.push_back(UserProperty(UserProperty::AUDIO, _("Content bit depth"), *bits, _("bits")));
+               }
        }
 
        FrameRateChange const frc (_parent->active_video_frame_rate(film), film->video_frame_rate());
@@ -347,18 +349,6 @@ AudioContent::add_properties (shared_ptr<const Film> film, list<UserProperty>& p
 }
 
 
-void
-AudioContent::set_streams (vector<AudioStreamPtr> streams)
-{
-       ContentChangeSignaller cc (_parent, AudioContentProperty::STREAMS);
-
-       {
-               boost::mutex::scoped_lock lm (_mutex);
-               _streams = streams;
-       }
-}
-
-
 AudioStreamPtr
 AudioContent::stream () const
 {
@@ -401,13 +391,12 @@ AudioContent::take_settings_from (shared_ptr<const AudioContent> c)
        set_fade_in (c->fade_in());
        set_fade_out (c->fade_out());
 
-       size_t i = 0;
-       size_t j = 0;
+       auto const streams_to_take = std::min(_streams.size(), c->_streams.size());
 
-       while (i < _streams.size() && j < c->_streams.size()) {
-               _streams[i]->set_mapping (c->_streams[j]->mapping());
-               ++i;
-               ++j;
+       for (auto i = 0U; i < streams_to_take; ++i) {
+               auto mapping = _streams[i]->mapping();
+               mapping.take_from(c->_streams[i]->mapping());
+               _streams[i]->set_mapping(mapping);
        }
 }
 
@@ -422,9 +411,11 @@ AudioContent::modify_position (shared_ptr<const Film> film, DCPTime& pos) const
 void
 AudioContent::modify_trim_start(shared_ptr<const Film> film, ContentTime& trim) const
 {
-       DCPOMATIC_ASSERT (!_streams.empty());
-       /* XXX: we're in trouble if streams have different rates */
-       trim = trim.round (_streams.front()->frame_rate());
+       /* When this trim is used it the audio will have been resampled, and using the
+        * DCP rate here reduces the chance of rounding errors causing audio glitches
+        * due to errors in placement of audio frames (#2373).
+        */
+       trim = trim.round(film ? film->audio_frame_rate() : 48000);
 }