X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=f425cf2808cfbf70d53f3ff2bf097a24ab0ba035;hb=cc27c2716f755305d67f1e1ba828ecf37f8405dd;hp=8d3b0e1288ea5b53627a9cfb82619b0ccc125c7e;hpb=ae9b0b509787d244366eb8f69bdf9d563b6c6bb6;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 8d3b0e128..f425cf280 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of the GNU General Public License as published by
@@ -19,31 +19,100 @@
 
 #include "audio_decoder.h"
 #include "audio_buffers.h"
-#include "exceptions.h"
-#include "log.h"
+#include "audio_processor.h"
 #include "resampler.h"
 #include "util.h"
-#include "film.h"
 
 #include "i18n.h"
 
-using std::stringstream;
 using std::list;
 using std::pair;
 using std::cout;
+using std::min;
+using std::max;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
-	: Decoder (film)
-	, _audio_content (content)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+	: _audio_content (content)
 {
-	if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
-		_resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
+	if (content->resampled_audio_frame_rate() != content->audio_frame_rate() && content->audio_channels ()) {
+		_resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ()));
 	}
+
+	if (content->audio_processor ()) {
+		_processor = content->audio_processor()->clone (content->resampled_audio_frame_rate ());
+	}
+
+	reset_decoded_audio ();
+}
+
+void
+AudioDecoder::reset_decoded_audio ()
+{
+	_decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0);
 }
 
-/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling.
+shared_ptr<ContentAudio>
+AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+{
+	shared_ptr<ContentAudio> dec;
+
+	AudioFrame const end = frame + length - 1;
+		
+	if (frame < _decoded_audio.frame || end > (_decoded_audio.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, _audio_content->audio_frame_rate()), accurate);
+	}
+
+	/* Offset of the data that we want from the start of _decoded_audio.audio
+	   (to be set up shortly)
+	*/
+	AudioFrame 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_audio until we have enough data, or the subclass does not want to give us any more */
+		while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+		decoded_offset = frame - _decoded_audio.frame;
+	} else {
+		while (!pass() && _decoded_audio.audio->frames() < length) {}
+		/* Use decoded_offset of 0, as we don't really care what frames we return */
+	}
+
+	/* The amount of data available in _decoded_audio.audio starting from `frame'.  This could be -ve
+	   if pass() returned true before we got enough data.
+	*/
+	AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
+
+	/* We will return either that, or the requested amount, whichever is smaller */
+	AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
+
+	/* Copy our data to the output */
+	shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
+	out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
+
+	AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
+
+	/* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
+	_decoded_audio.audio->move (decoded_offset + to_return, 0, remaining);
+	/* And set up the number of frames we have left */
+	_decoded_audio.audio->set_frames (remaining);
+	/* Also bump where those frames are in terms of the content */
+	_decoded_audio.frame += decoded_offset + to_return;
+
+	return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
+}
+
+/** Called by subclasses when audio data is ready.
+ *
+ *  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.
  *
@@ -57,15 +126,79 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 		data = _resampler->run (data);
 	}
 
+	if (_processor) {
+		data = _processor->run (data);
+	}
+
+	AudioFrame const frame_rate = _audio_content->resampled_audio_frame_rate ();
+
+	if (_seek_reference) {
+		/* We've had an accurate seek and now we're seeing some data */
+		ContentTime const delta = time - _seek_reference.get ();
+		AudioFrame const delta_frames = delta.frames (frame_rate);
+		if (delta_frames > 0) {
+			/* This data comes after the seek time.  Pad the data with some silence. */
+			shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
+			padded->make_silent ();
+			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 */
+			AudioFrame const to_discard = min (-delta_frames, static_cast<AudioFrame> (data->frames()));
+			AudioFrame 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<AudioBuffers> 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<ContentTime> ();
+	}
+
 	if (!_audio_position) {
-		shared_ptr<const Film> film = _film.lock ();
-		assert (film);
-		FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ());
-		_audio_position = (double (time) / frc.speed_up) * film->audio_frame_rate() / TIME_HZ;
+		_audio_position = time.frames (frame_rate);
+	}
+
+	assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+	add (data);
+}
+
+void
+AudioDecoder::add (shared_ptr<const AudioBuffers> data)
+{
+	/* Resize _decoded_audio to fit the new data */
+	int new_size = 0;
+	if (_decoded_audio.audio->frames() == 0) {
+		/* There's nothing in there, so just store the new data */
+		new_size = data->frames ();
+		_decoded_audio.frame = _audio_position.get ();
+	} else {
+		/* Otherwise we need to extend _decoded_audio to include the new stuff */
+		new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
 	}
+	
+	_decoded_audio.audio->ensure_size (new_size);
+	_decoded_audio.audio->set_frames (new_size);
 
-	_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position.get ())));
+	/* Copy new data in */
+	_decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
 	_audio_position = _audio_position.get() + data->frames ();
+
+	/* Limit the amount of data we keep in case nobody is asking for it */
+	int const max_frames = _audio_content->resampled_audio_frame_rate () * 10;
+	if (_decoded_audio.audio->frames() > max_frames) {
+		int const to_remove = _decoded_audio.audio->frames() - max_frames;
+		_decoded_audio.frame += to_remove;
+		_decoded_audio.audio->move (to_remove, 0, max_frames);
+		_decoded_audio.audio->set_frames (max_frames);
+	}
 }
 
 void
@@ -77,13 +210,19 @@ AudioDecoder::flush ()
 
 	shared_ptr<const AudioBuffers> b = _resampler->flush ();
 	if (b) {
-		_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
-		_audio_position = _audio_position.get() + b->frames ();
+		add (b);
 	}
 }
 
 void
-AudioDecoder::seek (ContentTime, bool)
+AudioDecoder::seek (ContentTime t, bool accurate)
 {
 	_audio_position.reset ();
+	reset_decoded_audio ();
+	if (accurate) {
+		_seek_reference = t;
+	}
+	if (_processor) {
+		_processor->flush ();
+	}
 }