X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_filter_graph.cc;h=0eeeb3c4ae5858b094ffd4f5da58ab659f9d3b75;hb=54af50c3b8e9082f9751e809d63540c51197a4a1;hp=0035fb21357a7d3dec5fd63f21e369d2f7fe2bdc;hpb=987bd787fcbecdb845e5e9bdf8435122841e7779;p=dcpomatic.git

diff --git a/src/lib/audio_filter_graph.cc b/src/lib/audio_filter_graph.cc
index 0035fb213..0eeeb3c4a 100644
--- a/src/lib/audio_filter_graph.cc
+++ b/src/lib/audio_filter_graph.cc
@@ -1,19 +1,20 @@
 /*
     Copyright (C) 2015 Carl Hetherington <cth@carlh.net>
 
-    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 <http://www.gnu.org/licenses/>.
 
 */
 
@@ -23,7 +24,9 @@
 extern "C" {
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/channel_layout.h>
 }
+#include <iostream>
 
 #include "i18n.h"
 
@@ -31,33 +34,40 @@ using std::string;
 using std::cout;
 using boost::shared_ptr;
 
-AudioFilterGraph::AudioFilterGraph (int sample_rate, int64_t channel_layout)
+AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
 	: _sample_rate (sample_rate)
-	, _channel_layout (channel_layout)
+	, _channels (channels)
 {
+	/* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
+	   so we need to tell it we're using 16 channels if we are using more than 8.
+	*/
+	if (_channels > 8) {
+		_channel_layout = av_get_default_channel_layout (16);
+	} else {
+		_channel_layout = av_get_default_channel_layout (_channels);
+	}
+
 	_in_frame = av_frame_alloc ();
 }
 
 AudioFilterGraph::~AudioFilterGraph()
 {
-	if (_in_frame) {
-		av_frame_free (&_in_frame);
-	}
+	av_frame_free (&_in_frame);
 }
 
 string
 AudioFilterGraph::src_parameters () const
 {
-	SafeStringStream a;
-
-	char buffer[64];
-	av_get_channel_layout_string (buffer, sizeof(buffer), 0, _channel_layout);
+	char layout[64];
+	av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
 
-	a << "time_base=1/1:sample_rate=" << _sample_rate << ":"
-	  << "sample_fmt=" << av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP) << ":"
-	  << "channel_layout=" << buffer;
+	char buffer[256];
+	snprintf (
+		buffer, sizeof(buffer), "time_base=1/1:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
+		_sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), layout
+		);
 
-	return a.str ();
+	return buffer;
 }
 
 void *
@@ -97,7 +107,27 @@ AudioFilterGraph::sink_name () const
 void
 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
 {
-	_in_frame->extended_data = new uint8_t*[buffers->channels()];
+	DCPOMATIC_ASSERT (buffers->frames() > 0);
+	int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
+	DCPOMATIC_ASSERT (process_channels >= buffers->channels());
+
+	if (buffers->channels() < process_channels) {
+		/* We are processing more data than we actually have (see the hack in
+		   the constructor) so we need to create new buffers with some extra
+		   silent channels.
+		*/
+		shared_ptr<AudioBuffers> extended_buffers (new AudioBuffers (process_channels, buffers->frames()));
+		for (int i = 0; i < buffers->channels(); ++i) {
+			extended_buffers->copy_channel_from (buffers.get(), i, i);
+		}
+		for (int i = buffers->channels(); i < process_channels; ++i) {
+			extended_buffers->make_silent (i);
+		}
+
+		buffers = extended_buffers;
+	}
+
+	_in_frame->extended_data = new uint8_t*[process_channels];
 	for (int i = 0; i < buffers->channels(); ++i) {
 		if (i < AV_NUM_DATA_POINTERS) {
 			_in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
@@ -109,16 +139,20 @@ AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
 	_in_frame->format = AV_SAMPLE_FMT_FLTP;
 	_in_frame->sample_rate = _sample_rate;
 	_in_frame->channel_layout = _channel_layout;
-	_in_frame->channels = av_get_channel_layout_nb_channels (_channel_layout);
+	_in_frame->channels = process_channels;
 
 	int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
 
 	delete[] _in_frame->extended_data;
+	/* Reset extended_data to its original value so that av_frame_free
+	   does not try to free it.
+	*/
+	_in_frame->extended_data = _in_frame->data;
 
 	if (r < 0) {
 		char buffer[256];
 		av_strerror (r, buffer, sizeof(buffer));
-		throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), buffer));
+		throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), &buffer[0]));
 	}
 
 	while (true) {