Rename AudioContent frame_rate methods and move resampled_audio_frame_rate into AudioContent.

10 files changed, 46 insertions, 54 deletions

diff --git a/src/lib/audio_content.cc b/src/lib/audio_content.cc
index c84f57130..d9e00ff14 100644
--- a/src/lib/audio_content.cc
+++ b/src/lib/audio_content.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2013 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2013-2014 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
@@ -150,11 +150,11 @@ string
 AudioContent::technical_summary () const
 {
 	return String::compose (
-		"audio: channels %1, length %2, raw rate %3, out rate %4",
+		"audio: channels %1, length %2, content rate %3, resampled rate %4",
 		audio_channels(),
 		audio_length().seconds(),
-		content_audio_frame_rate(),
-		output_audio_frame_rate()
+		audio_frame_rate(),
+		resampled_audio_frame_rate()
 		);
 }
 
@@ -163,3 +163,29 @@ AudioContent::set_audio_mapping (AudioMapping)
 {
 	signal_changed (AudioContentProperty::AUDIO_MAPPING);
 }
+
+/** @return the frame rate that this content should be resampled to in order
+ *  that it is in sync with the active video content at its start time.
+ */
+int
+AudioContent::resampled_audio_frame_rate () const
+{
+	shared_ptr<const Film> film = _film.lock ();
+	assert (film);
+	
+	/* Resample to a DCI-approved sample rate */
+	double t = dcp_audio_frame_rate (audio_frame_rate ());
+
+	FrameRateChange frc = film->active_frame_rate_change (position ());
+
+	/* Compensate if the DCP is being run at a different frame rate
+	   to the source; that is, if the video is run such that it will
+	   look different in the DCP compared to the source (slower or faster).
+	*/
+
+	if (frc.change_speed) {
+		t /= frc.speed_up;
+	}
+
+	return rint (t);
+}
diff --git a/src/lib/audio_content.h b/src/lib/audio_content.h
index 60d53b343..09f90b85c 100644
--- a/src/lib/audio_content.h
+++ b/src/lib/audio_content.h
@@ -63,12 +63,13 @@ public:
 	/** @return the length of the audio in the content */
 	virtual ContentTime audio_length () const = 0;
 	/** @return the frame rate of the content */
-	virtual int content_audio_frame_rate () const = 0;
-	virtual int output_audio_frame_rate () const = 0;
+	virtual int audio_frame_rate () const = 0;
 	virtual AudioMapping audio_mapping () const = 0;
 	virtual void set_audio_mapping (AudioMapping);
 	virtual boost::filesystem::path audio_analysis_path () const;
 
+	int resampled_audio_frame_rate () const;
+
 	boost::signals2::connection analyse_audio (boost::function<void()>);
 
 	void set_audio_gain (float);
diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index f914ecf8a..4a543cea9 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -39,8 +39,8 @@ using boost::shared_ptr;
 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 ()));
 	}
 
 	reset_decoded_audio ();
@@ -61,7 +61,7 @@ AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
 		
 	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->content_audio_frame_rate()), accurate);
+		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
@@ -126,7 +126,7 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 	}
 
 	if (!_audio_position) {
-		_audio_position = time.frames (_audio_content->output_audio_frame_rate ());
+		_audio_position = time.frames (_audio_content->resampled_audio_frame_rate ());
 	}
 
 	assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
diff --git a/src/lib/ffmpeg_content.cc b/src/lib/ffmpeg_content.cc
index 4e14802e8..a51cb3de8 100644
--- a/src/lib/ffmpeg_content.cc
+++ b/src/lib/ffmpeg_content.cc
@@ -288,7 +288,7 @@ FFmpegContent::audio_channels () const
 }
 
 int
-FFmpegContent::content_audio_frame_rate () const
+FFmpegContent::audio_frame_rate () const
 {
 	boost::mutex::scoped_lock lm (_mutex);
 
@@ -299,29 +299,6 @@ FFmpegContent::content_audio_frame_rate () const
 	return _audio_stream->frame_rate;
 }
 
-int
-FFmpegContent::output_audio_frame_rate () const
-{
-	shared_ptr<const Film> film = _film.lock ();
-	assert (film);
-	
-	/* Resample to a DCI-approved sample rate */
-	double t = dcp_audio_frame_rate (content_audio_frame_rate ());
-
-	FrameRateChange frc (video_frame_rate(), film->video_frame_rate());
-
-	/* Compensate if the DCP is being run at a different frame rate
-	   to the source; that is, if the video is run such that it will
-	   look different in the DCP compared to the source (slower or faster).
-	*/
-
-	if (frc.change_speed) {
-		t /= frc.speed_up;
-	}
-
-	return rint (t);
-}
-
 bool
 operator== (FFmpegStream const & a, FFmpegStream const & b)
 {
diff --git a/src/lib/ffmpeg_content.h b/src/lib/ffmpeg_content.h
index d86c90af9..1ab0a92d0 100644
--- a/src/lib/ffmpeg_content.h
+++ b/src/lib/ffmpeg_content.h
@@ -146,8 +146,7 @@ public:
 	/* AudioContent */
 	int audio_channels () const;
 	ContentTime audio_length () const;
-	int content_audio_frame_rate () const;
-	int output_audio_frame_rate () const;
+	int audio_frame_rate () const;
 	AudioMapping audio_mapping () const;
 	void set_audio_mapping (AudioMapping);
 	boost::filesystem::path audio_analysis_path () const;
diff --git a/src/lib/player.cc b/src/lib/player.cc
index 5fe35e34d..75b550093 100644
--- a/src/lib/player.cc
+++ b/src/lib/player.cc
@@ -439,7 +439,7 @@ Player::get_audio (DCPTime time, DCPTime length, bool accurate)
 		shared_ptr<AudioDecoder> decoder = dynamic_pointer_cast<AudioDecoder> ((*i)->decoder);
 		assert (decoder);
 
-		if (content->content_audio_frame_rate() == 0) {
+		if (content->audio_frame_rate() == 0) {
 			/* This AudioContent has no audio (e.g. if it is an FFmpegContent with no
 			 * audio stream).
 			 */
diff --git a/src/lib/sndfile_content.cc b/src/lib/sndfile_content.cc
index 5b408f2da..0cf65967f 100644
--- a/src/lib/sndfile_content.cc
+++ b/src/lib/sndfile_content.cc
@@ -80,8 +80,8 @@ SndfileContent::information () const
 	s << String::compose (
 		_("%1 channels, %2kHz, %3 samples"),
 		audio_channels(),
-		content_audio_frame_rate() / 1000.0,
-		audio_length().frames (content_audio_frame_rate ())
+		audio_frame_rate() / 1000.0,
+		audio_length().frames (audio_frame_rate ())
 		);
 	
 	return s.str ();
@@ -134,7 +134,7 @@ SndfileContent::as_xml (xmlpp::Node* node) const
 
 	node->add_child("AudioChannels")->add_child_text (lexical_cast<string> (audio_channels ()));
 	node->add_child("AudioLength")->add_child_text (lexical_cast<string> (audio_length().get ()));
-	node->add_child("AudioFrameRate")->add_child_text (lexical_cast<string> (content_audio_frame_rate ()));
+	node->add_child("AudioFrameRate")->add_child_text (lexical_cast<string> (audio_frame_rate ()));
 	_audio_mapping.as_xml (node->add_child("AudioMapping"));
 }
 
@@ -146,15 +146,6 @@ SndfileContent::full_length () const
 	return DCPTime (audio_length(), film->active_frame_rate_change (position ()));
 }
 
-int
-SndfileContent::output_audio_frame_rate () const
-{
-	shared_ptr<const Film> film = _film.lock ();
-	assert (film);
-	
-	return film->audio_frame_rate ();
-}
-
 void
 SndfileContent::set_audio_mapping (AudioMapping m)
 {
diff --git a/src/lib/sndfile_content.h b/src/lib/sndfile_content.h
index 04d157131..bbc5111ff 100644
--- a/src/lib/sndfile_content.h
+++ b/src/lib/sndfile_content.h
@@ -57,13 +57,11 @@ public:
 		return _audio_length;
 	}
 	
-	int content_audio_frame_rate () const {
+	int audio_frame_rate () const {
 		boost::mutex::scoped_lock lm (_mutex);
 		return _audio_frame_rate;
 	}
 
-	int output_audio_frame_rate () const;
-
 	AudioMapping audio_mapping () const {
 		boost::mutex::scoped_lock lm (_mutex);
 		return _audio_mapping;
diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index c37a84474..602014d58 100644
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -74,7 +74,7 @@ SndfileDecoder::pass ()
 	/* Do things in half second blocks as I think there may be limits
 	   to what FFmpeg (and in particular the resampler) can cope with.
 	*/
-	sf_count_t const block = _sndfile_content->content_audio_frame_rate() / 2;
+	sf_count_t const block = _sndfile_content->audio_frame_rate() / 2;
 	sf_count_t const this_time = min (block, _remaining);
 
 	int const channels = _sndfile_content->audio_channels ();
diff --git a/src/wx/audio_panel.cc b/src/wx/audio_panel.cc
index 1c679d336..eb95e17ab 100644
--- a/src/wx/audio_panel.cc
+++ b/src/wx/audio_panel.cc
@@ -237,7 +237,7 @@ AudioPanel::setup_stream_description ()
 		} else {
 			s << fcs->audio_channels() << wxT (" ") << _("channels");
 		}
-		s << wxT (", ") << fcs->content_audio_frame_rate() << _("Hz");
+		s << wxT (", ") << fcs->audio_frame_rate() << _("Hz");
 		_description->SetLabel (s);
 	}
 }