Supporters update.
[dcpomatic.git] / src / lib / audio_filter_graph.cc
index fd2492d3b8960f9a0da48a350c40fa9fe281279b..4e3052d57c519eccc3f390e01d5f3e4c3ad6da5f 100644 (file)
@@ -1,41 +1,63 @@
 /*
     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/>.
 
 */
 
-#include "audio_filter_graph.h"
+
 #include "audio_buffers.h"
+#include "audio_filter_graph.h"
 #include "compose.hpp"
+#include "dcpomatic_assert.h"
+#include "exceptions.h"
 extern "C" {
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/opt.h>
 }
+#include <iostream>
 
 #include "i18n.h"
 
-using std::string;
+
 using std::cout;
-using boost::shared_ptr;
+using std::make_shared;
+using std::shared_ptr;
+using std::string;
+
 
-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 ();
+       if (_in_frame == nullptr) {
+               throw std::bad_alloc();
+       }
 }
 
 AudioFilterGraph::~AudioFilterGraph()
@@ -46,40 +68,36 @@ AudioFilterGraph::~AudioFilterGraph()
 string
 AudioFilterGraph::src_parameters () const
 {
-       SafeStringStream a;
+       char layout[64];
+       av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
 
-       char buffer[64];
-       av_get_channel_layout_string (buffer, sizeof(buffer), 0, _channel_layout);
+       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
+               );
 
-       a << "time_base=1/1:sample_rate=" << _sample_rate << ":"
-         << "sample_fmt=" << av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP) << ":"
-         << "channel_layout=" << buffer;
-
-       return a.str ();
+       return buffer;
 }
 
-void *
-AudioFilterGraph::sink_parameters () const
-{
-       AVABufferSinkParams* sink_params = av_abuffersink_params_alloc ();
-
-       AVSampleFormat* sample_fmts = new AVSampleFormat[2];
-       sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
-       sample_fmts[1] = AV_SAMPLE_FMT_NONE;
-       sink_params->sample_fmts = sample_fmts;
 
-       int64_t* channel_layouts = new int64_t[2];
-       channel_layouts[0] = _channel_layout;
-       channel_layouts[1] = -1;
-       sink_params->channel_layouts = channel_layouts;
+void
+AudioFilterGraph::set_parameters (AVFilterContext* context) const
+{
+       AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
+       int r = av_opt_set_int_list (context, "sample_fmts", sample_fmts, AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+       DCPOMATIC_ASSERT (r >= 0);
 
-       sink_params->sample_rates = new int[2];
-       sink_params->sample_rates[0] = _sample_rate;
-       sink_params->sample_rates[1] = -1;
+       int64_t channel_layouts[] = { _channel_layout, -1 };
+       r = av_opt_set_int_list (context, "channel_layouts", channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
+       DCPOMATIC_ASSERT (r >= 0);
 
-       return sink_params;
+       int sample_rates[] = { _sample_rate, -1 };
+       r = av_opt_set_int_list (context, "sample_rates", sample_rates, -1, AV_OPT_SEARCH_CHILDREN);
+       DCPOMATIC_ASSERT (r >= 0);
 }
 
+
 string
 AudioFilterGraph::src_name () const
 {
@@ -93,9 +111,29 @@ AudioFilterGraph::sink_name () const
 }
 
 void
-AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
+AudioFilterGraph::process (shared_ptr<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.
+               */
+               auto extended_buffers = make_shared<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));
@@ -107,7 +145,7 @@ 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);
 
@@ -120,7 +158,7 @@ AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
        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) {