Don't cache sample format, fix up various things.

8 files changed, 20 insertions, 74 deletions

diff --git a/src/lib/decoder.cc b/src/lib/decoder.cc
index ec046fcaf..16d815a07 100644
--- a/src/lib/decoder.cc
+++ b/src/lib/decoder.cc
@@ -90,7 +90,7 @@ Decoder::~Decoder ()
 void
 Decoder::process_begin ()
 {
-	_delay_in_bytes = _fs->audio_delay() * _fs->audio_sample_rate() * _fs->audio_channels() * _fs->bytes_per_sample() / 1000;
+	_delay_in_bytes = _fs->audio_delay() * _fs->audio_sample_rate() * _fs->audio_channels() * bytes_per_audio_sample() / 1000;
 	delete _delay_line;
 	_delay_line = new DelayLine (_delay_in_bytes);
 
@@ -104,7 +104,7 @@ Decoder::process_end ()
 	if (_delay_in_bytes < 0) {
 		uint8_t remainder[-_delay_in_bytes];
 		_delay_line->get_remaining (remainder);
-		_audio_frames_processed += _delay_in_bytes / (_fs->audio_channels() * _fs->bytes_per_sample());
+		_audio_frames_processed += _delay_in_bytes / (_fs->audio_channels() * bytes_per_audio_sample());
 		emit_audio (remainder, _delay_in_bytes);
 	}
 
@@ -122,7 +122,7 @@ Decoder::process_end ()
 		s << "Adding " << audio_short_by_frames << " frames of silence to the end.";
 		_log->log (s.str ());
 
-		int64_t bytes = audio_short_by_frames * _fs->audio_channels() * _fs->bytes_per_sample();
+		int64_t bytes = audio_short_by_frames * _fs->audio_channels() * bytes_per_audio_sample();
 		
 		int64_t const silence_size = 64 * 1024;
 		uint8_t silence[silence_size];
@@ -193,13 +193,13 @@ Decoder::emit_audio (uint8_t* data, int size)
 	/* Deinterleave and convert to float */
 
 	float* samples[_fs->audio_channels()];
-	int const total_samples = size / _fs->bytes_per_sample();
+	int const total_samples = size / bytes_per_audio_sample();
 	int const frames = total_samples / _fs->audio_channels();
 	for (int i = 0; i < _fs->audio_channels(); ++i) {
 		samples[i] = new float[frames];
 	}
 
-	switch (_fs->audio_sample_format()) {
+	switch (audio_sample_format()) {
 	case AV_SAMPLE_FMT_S16:
 	{
 		uint8_t* p = data;
@@ -213,8 +213,6 @@ Decoder::emit_audio (uint8_t* data, int size)
 			/* float sample */
 			samples[channel][sample] = float(os) / 0x8000;
 
-			cout << samples[channel][sample] << " from s16\n";
-			
 			++channel;
 			if (channel == _fs->audio_channels()) {
 				channel = 0;
@@ -230,11 +228,8 @@ Decoder::emit_audio (uint8_t* data, int size)
 	{
 		float* p = reinterpret_cast<float*> (data);
 		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			for (int j = 0; j < frames; ++j) {
-				samples[i][j] = *p++;
-				cout << samples[i][j] << " from float.\n";
-				++p;
-			}
+			memcpy (samples[i], p, frames * sizeof(float));
+			p += frames;
 		}
 	}
 	break;
@@ -460,3 +455,10 @@ Decoder::process_subtitle (shared_ptr<TimedSubtitle> s)
 		_timed_subtitle->subtitle()->set_position (Position (p.x - _fs->crop().left, p.y - _fs->crop().top));
 	}
 }
+
+
+int
+Decoder::bytes_per_audio_sample () const
+{
+	return av_get_bytes_per_sample (audio_sample_format ());
+}
diff --git a/src/lib/decoder.h b/src/lib/decoder.h
index 88dd90f0c..14e602ace 100644
--- a/src/lib/decoder.h
+++ b/src/lib/decoder.h
@@ -138,6 +138,7 @@ protected:
 private:
 	void setup_video_filters ();
 	void emit_audio (uint8_t* data, int size);
+	int bytes_per_audio_sample () const;
 	
 	/** last video frame to be processed */
 	int _video_frame;
diff --git a/src/lib/ffmpeg_decoder.h b/src/lib/ffmpeg_decoder.h
index 9abc7f49a..0d256b37e 100644
--- a/src/lib/ffmpeg_decoder.h
+++ b/src/lib/ffmpeg_decoder.h
@@ -65,6 +65,7 @@ public:
 	AVSampleFormat audio_sample_format () const;
 	int64_t audio_channel_layout () const;
 	bool has_subtitles () const;
+	int bytes_per_audio_sample () const;
 
 	std::vector<Stream> audio_streams () const;
 	std::vector<Stream> subtitle_streams () const;
diff --git a/src/lib/film_state.cc b/src/lib/film_state.cc
index f2d83e4e2..46359677c 100644
--- a/src/lib/film_state.cc
+++ b/src/lib/film_state.cc
@@ -115,7 +115,6 @@ FilmState::write_metadata () const
 	f << "length " << _length << "\n";
 	f << "audio_channels " << _audio_channels << "\n";
 	f << "audio_sample_rate " << _audio_sample_rate << "\n";
-	f << "audio_sample_format " << audio_sample_format_to_string (_audio_sample_format) << "\n";
 	f << "content_digest " << _content_digest << "\n";
 	f << "has_subtitles " << _has_subtitles << "\n";
 
@@ -224,8 +223,6 @@ FilmState::read_metadata ()
 			_audio_channels = atoi (v.c_str ());
 		} else if (k == "audio_sample_rate") {
 			_audio_sample_rate = atoi (v.c_str ());
-		} else if (k == "audio_sample_format") {
-			_audio_sample_format = audio_sample_format_from_string (v);
 		} else if (k == "content_digest") {
 			_content_digest = v;
 		} else if (k == "has_subtitles") {
@@ -354,13 +351,6 @@ FilmState::content_type () const
 	return VIDEO;
 }
 
-/** @return Number of bytes per sample of a single channel */
-int
-FilmState::bytes_per_sample () const
-{
-	return av_get_bytes_per_sample (_audio_sample_format);
-}
-
 int
 FilmState::target_sample_rate () const
 {
@@ -543,7 +533,6 @@ FilmState::set_content (string c)
 	set_frames_per_second (d->frames_per_second ());
 	set_audio_channels (d->audio_channels ());
 	set_audio_sample_rate (d->audio_sample_rate ());
-	set_audio_sample_format (d->audio_sample_format ());
 	set_has_subtitles (d->has_subtitles ());
 	set_audio_streams (d->audio_streams ());
 	set_subtitle_streams (d->subtitle_streams ());
@@ -796,13 +785,6 @@ FilmState::set_audio_sample_rate (int r)
 }
 
 void
-FilmState::set_audio_sample_format (AVSampleFormat f)
-{
-	_audio_sample_format = f;
-	_dirty = true;
-}
-
-void
 FilmState::set_content_digest (string d)
 {
 	_content_digest = d;
diff --git a/src/lib/film_state.h b/src/lib/film_state.h
index 2a00c844a..40908693f 100644
--- a/src/lib/film_state.h
+++ b/src/lib/film_state.h
@@ -72,7 +72,6 @@ public:
 		, _length (0)
 		, _audio_channels (0)
 		, _audio_sample_rate (0)
-		, _audio_sample_format (AV_SAMPLE_FMT_NONE)
 		, _has_subtitles (false)
 		, _frames_per_second (0)
 		, _dirty (false)
@@ -90,7 +89,6 @@ public:
 	std::string thumb_base (int) const;
 	int thumb_frame (int) const;
 
-	int bytes_per_sample () const;
 	int target_sample_rate () const;
 	
 	void write_metadata () const;
@@ -272,10 +270,6 @@ public:
 		return _audio_sample_rate;
 	}
 	
-	AVSampleFormat audio_sample_format () const {
-		return _audio_sample_format;
-	}
-	
 	std::string content_digest () const {
 		return _content_digest;
 	}
@@ -335,7 +329,6 @@ public:
 	void set_length (int);
 	void set_audio_channels (int);
 	void set_audio_sample_rate (int);
-	void set_audio_sample_format (AVSampleFormat);
 	void set_content_digest (std::string);
 	void set_has_subtitles (bool);
 	void set_audio_streams (std::vector<Stream>);
@@ -422,8 +415,6 @@ private:
 	int _audio_channels;
 	/** Sample rate of the source audio, in Hz */
 	int _audio_sample_rate;
-	/** Format of the audio samples */
-	AVSampleFormat _audio_sample_format;
 	/** MD5 digest of our content file */
 	std::string _content_digest;
 	/** true if the source has subtitles */
diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index 58f8a101f..7697a5e78 100644
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -351,16 +351,16 @@ J2KWAVEncoder::process_audio (float** data, int frames)
 	if (_swr_context) {
 
 		/* Compute the resampled frames count and add 32 for luck */
-		int const resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
+		int const max_resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
 
 		/* Make a buffer to put the result in */
 		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			resampled[i] = new float[resampled_frames];
+			resampled[i] = new float[max_resampled_frames];
 		}
 
 		/* Resample audio */
-		int out_frames = swr_convert (_swr_context, (uint8_t **) resampled, resampled_frames, (uint8_t const **) data, frames);
-		if (out_frames < 0) {
+		int const resampled_frames = swr_convert (_swr_context, (uint8_t **) resampled, max_resampled_frames, (uint8_t const **) data, frames);
+		if (resampled_frames < 0) {
 			throw EncodeError ("could not run sample-rate converter");
 		}
 
diff --git a/src/lib/util.cc b/src/lib/util.cc
index 47a86da9e..3f5200ead 100644
--- a/src/lib/util.cc
+++ b/src/lib/util.cc
@@ -189,35 +189,6 @@ stacktrace (ostream& out, int levels)
 }
 #endif
 
-/** @param s Sample format.
- *  @return String representation.
- */
-string
-audio_sample_format_to_string (AVSampleFormat s)
-{
-	/* Our sample format handling is not exactly complete */
-	
-	switch (s) {
-	case AV_SAMPLE_FMT_S16:
-		return "S16";
-	default:
-		assert (false);
-	}
-}
-
-/** @param s String representation of a sample format, as returned from audio_sample_format_to_string().
- *  @return Sample format.
- */
-AVSampleFormat
-audio_sample_format_from_string (string s)
-{
-	if (s == "S16") {
-		return AV_SAMPLE_FMT_S16;
-	}
-
-	assert (false);
-}
-
 /** @return Version of vobcopy that is on the path (and hence that we will use) */
 static string
 vobcopy_version ()
diff --git a/src/lib/util.h b/src/lib/util.h
index c2706a594..f2087b3f0 100644
--- a/src/lib/util.h
+++ b/src/lib/util.h
@@ -46,8 +46,6 @@ class Scaler;
 extern std::string seconds_to_hms (int);
 extern std::string seconds_to_approximate_hms (int);
 extern void stacktrace (std::ostream &, int);
-extern std::string audio_sample_format_to_string (AVSampleFormat);
-extern AVSampleFormat audio_sample_format_from_string (std::string);
 extern std::string dependency_version_summary ();
 extern double seconds (struct timeval);
 extern void dvdomatic_setup ();