Basic grunt-work, untested and unfinished, but it compiles.

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

diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
index a82ebc4cf4b04e8b4d48dec0c20875994078d06c..8f0905e0d7b58f34c8edbc8143b4ff4d2b4b2ae0 100644 (file)
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -66,102 +66,6 @@ AudioDecoderStream::reset_decoded ()
        _decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
 }
 
-ContentAudio
-AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
-{
-       shared_ptr<ContentAudio> dec;
-
-       _log->log (
-               String::compose (
-                       "ADS has request for %1 %2; has %3 %4",
-                       frame, length, _decoded.frame, _decoded.audio->frames()
-                       ), LogEntry::TYPE_DEBUG_DECODE
-               );
-
-       Frame const from = frame;
-       Frame const to = from + length;
-       Frame const have_from = _decoded.frame;
-       Frame const have_to = _decoded.frame + _decoded.audio->frames();
-
-       optional<Frame> missing;
-       if (have_from > from || have_to < to) {
-               /* We need something */
-               if (have_from <= from && from < have_to) {
-                       missing = have_to;
-               } else {
-                       missing = from;
-               }
-       }
-
-       if (missing) {
-               optional<ContentTime> pos = _audio_decoder->position ();
-               _log->log (
-                       String::compose ("ADS suggests seek to %1 (now at %2)", *missing, pos ? to_string(pos.get()) : "none"),
-                       LogEntry::TYPE_DEBUG_DECODE
-                       );
-               _audio_decoder->maybe_seek (ContentTime::from_frames (*missing, _content->resampled_frame_rate()), accurate);
-       }
-
-       /* Offset of the data that we want from the start of _decoded.audio
-          (to be set up shortly)
-       */
-       Frame decoded_offset = 0;
-
-       /* Now enough pass() calls will either:
-        *  (a) give us what we want, or
-        *  (b) hit the end of the decoder.
-        *
-        * If we are being accurate, we want the right frames,
-        * otherwise any frames will do.
-        */
-       if (accurate) {
-               /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
-               while (
-                       (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) <= to) &&
-                       !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
-                       )
-               {}
-
-               decoded_offset = frame - _decoded.frame;
-
-               _log->log (
-                       String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, have_from),
-                       LogEntry::TYPE_DEBUG_DECODE
-                       );
-       } else {
-               while (
-                       _decoded.audio->frames() < length &&
-                       !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
-                       )
-               {}
-
-               /* Use decoded_offset of 0, as we don't really care what frames we return */
-       }
-
-       /* The amount of data available in _decoded.audio starting from `frame'.  This could be -ve
-          if pass() returned true before we got enough data.
-       */
-       Frame const available = _decoded.audio->frames() - decoded_offset;
-
-       /* We will return either that, or the requested amount, whichever is smaller */
-       Frame const to_return = max ((Frame) 0, min (available, length));
-
-       /* Copy our data to the output */
-       shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
-       out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
-
-       Frame const remaining = max ((Frame) 0, available - to_return);
-
-       /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
-       _decoded.audio->move (decoded_offset + to_return, 0, remaining);
-       /* And set up the number of frames we have left */
-       _decoded.audio->set_frames (remaining);
-       /* Also bump where those frames are in terms of the content */
-       _decoded.frame += decoded_offset + to_return;
-
-       return ContentAudio (out, frame);
-}
-
 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
  *  We have to assume that we are feeding continuous data into the resampler, and so we get continuous
  *  data out.  Hence we do the timestamping here, post-resampler, just by counting samples.
@@ -255,16 +159,6 @@ AudioDecoderStream::flush ()
        }
 }
 
-void
-AudioDecoderStream::seek (ContentTime t, bool accurate)
-{
-       _position.reset ();
-       reset_decoded ();
-       if (accurate) {
-               _seek_reference = t;
-       }
-}
-
 void
 AudioDecoderStream::set_fast ()
 {