/*
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/>.
*/
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::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 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
{
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));
_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) {