X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder_stream.cc;h=8f0905e0d7b58f34c8edbc8143b4ff4d2b4b2ae0;hb=de2af791bdfdcd653752cba970e59efc7bf810c7;hp=36274b502692e3d95bbd76e7a8162d7968a287b3;hpb=a0d1dd5d91c81ec9907cbc7b890905c463c18f62;p=dcpomatic.git diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc index 36274b502..8f0905e0d 100644 --- a/src/lib/audio_decoder_stream.cc +++ b/src/lib/audio_decoder_stream.cc @@ -1,19 +1,20 @@ /* - Copyright (C) 2012-2015 Carl Hetherington + Copyright (C) 2012-2016 Carl Hetherington - This program is free software; you can redistribute it and/or modify + This file is part of DCP-o-matic. + + DCP-o-matic is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. - This program is distributed in the hope that it will be useful, + DCP-o-matic is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + along with DCP-o-matic. If not, see . */ @@ -25,6 +26,8 @@ #include "util.h" #include "film.h" #include "log.h" +#include "audio_content.h" +#include "compose.hpp" #include #include "i18n.h" @@ -37,13 +40,21 @@ using std::max; using boost::optional; using boost::shared_ptr; -AudioDecoderStream::AudioDecoderStream (shared_ptr content, AudioStreamPtr stream, AudioDecoder* decoder) +AudioDecoderStream::AudioDecoderStream ( + shared_ptr content, AudioStreamPtr stream, Decoder* decoder, AudioDecoder* audio_decoder, shared_ptr log + ) : _content (content) , _stream (stream) , _decoder (decoder) + , _audio_decoder (audio_decoder) + , _log (log) + /* We effectively start having done a seek to zero; this allows silence-padding of the first + data that comes out of our decoder. + */ + , _seek_reference (ContentTime ()) { - if (content->resampled_audio_frame_rate() != _stream->frame_rate()) { - _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels ())); + if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) { + _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ())); } reset_decoded (); @@ -55,75 +66,6 @@ AudioDecoderStream::reset_decoded () _decoded = ContentAudio (shared_ptr (new AudioBuffers (_stream->channels(), 0)), 0); } -ContentAudio -AudioDecoderStream::get (Frame frame, Frame length, bool accurate) -{ - shared_ptr dec; - - _content->film()->log()->log (String::compose ("ADS has request for %1 %2", frame, length), Log::TYPE_DEBUG_DECODE); - - Frame const end = frame + length - 1; - - if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) { - /* Either we have no decoded data, or what we do have is a long way from what we want: seek */ - seek (ContentTime::from_frames (frame, _content->resampled_audio_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()) < end) && - !_decoder->pass () - ) - {} - - decoded_offset = frame - _decoded.frame; - } else { - while ( - _decoded.audio->frames() < length && - !_decoder->pass () - ) - {} - - /* 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 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. @@ -134,13 +76,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate) void AudioDecoderStream::audio (shared_ptr data, ContentTime time) { - _content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), Log::TYPE_DEBUG_DECODE); + _log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE); if (_resampler) { data = _resampler->run (data); } - Frame const frame_rate = _content->resampled_audio_frame_rate (); + Frame const frame_rate = _content->resampled_frame_rate (); if (_seek_reference) { /* We've had an accurate seek and now we're seeing some data */ @@ -153,20 +95,6 @@ AudioDecoderStream::audio (shared_ptr data, ContentTime time padded->copy_from (data.get(), data->frames(), 0, delta_frames); data = padded; time -= delta; - } else if (delta_frames < 0) { - /* This data comes before the seek time. Throw some data away */ - Frame const to_discard = min (-delta_frames, static_cast (data->frames())); - Frame const to_keep = data->frames() - to_discard; - if (to_keep == 0) { - /* We have to throw all this data away, so keep _seek_reference and - try again next time some data arrives. - */ - return; - } - shared_ptr trimmed (new AudioBuffers (data->channels(), to_keep)); - trimmed->copy_from (data.get(), to_keep, to_discard, 0); - data = trimmed; - time += ContentTime::from_frames (to_discard, frame_rate); } _seek_reference = optional (); } @@ -209,7 +137,7 @@ AudioDecoderStream::add (shared_ptr data) _position = _position.get() + data->frames (); /* Limit the amount of data we keep in case nobody is asking for it */ - int const max_frames = _content->resampled_audio_frame_rate () * 10; + int const max_frames = _content->resampled_frame_rate () * 10; if (_decoded.audio->frames() > max_frames) { int const to_remove = _decoded.audio->frames() - max_frames; _decoded.frame += to_remove; @@ -232,11 +160,19 @@ AudioDecoderStream::flush () } void -AudioDecoderStream::seek (ContentTime t, bool accurate) +AudioDecoderStream::set_fast () { - _position.reset (); - reset_decoded (); - if (accurate) { - _seek_reference = t; + if (_resampler) { + _resampler->set_fast (); } } + +optional +AudioDecoderStream::position () const +{ + if (!_position) { + return optional (); + } + + return ContentTime::from_frames (_position.get(), _content->resampled_frame_rate()); +}