Hacky workaround for FFmpeg not having a channel layout for any counts between 8...
authorCarl Hetherington <cth@carlh.net>
Mon, 15 Feb 2016 22:34:31 +0000 (22:34 +0000)
committerCarl Hetherington <cth@carlh.net>
Mon, 15 Feb 2016 22:34:31 +0000 (22:34 +0000)
ChangeLog
src/lib/analyse_audio_job.cc
src/lib/audio_filter_graph.cc
src/lib/audio_filter_graph.h
test/audio_analysis_test.cc

index ac5fe0333efb62b9a07d2e96a3e398a1d6873f6d..a96740d1f2c14d36ae46eb0cc13a14a61183dd75 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,8 @@
+2016-02-15  Carl Hetherington  <cth@carlh.net>
+
+       * Fix exception when analysing audio of projects with more
+       than 8 DCP channels.
+
 2016-02-12  Carl Hetherington  <cth@carlh.net>
 
        * Add basic support for SSA (SubStation Alpha) subtitles (#128).
index 769f3762bee98ee214c9972ddb0c6018c0cc8767..d17c4c30bfb06b51b43594064b4e392b3c8a7a0e 100644 (file)
@@ -56,7 +56,7 @@ AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const
        , _sample_peak (0)
        , _sample_peak_frame (0)
 #ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
-       , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), av_get_default_channel_layout(film->audio_channels())))
+       , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), film->audio_channels()))
 #endif
 {
 #ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
index fd2492d3b8960f9a0da48a350c40fa9fe281279b..a43f1881ea515755ed34176bc43784d82807867e 100644 (file)
@@ -31,10 +31,19 @@ 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 ();
 }
 
@@ -95,7 +104,26 @@ AudioFilterGraph::sink_name () const
 void
 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
 {
-       _in_frame->extended_data = new uint8_t*[buffers->channels()];
+       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));
@@ -107,7 +135,7 @@ 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);
 
index 8efff5d8d5df244e6963f909ecc68ec763540ab8..90518e2edfa5fd06582bd070887008a3165c6774 100644 (file)
@@ -27,7 +27,7 @@ class AudioBuffers;
 class AudioFilterGraph : public FilterGraph
 {
 public:
-       AudioFilterGraph (int sample_rate, int64_t channel_layout);
+       AudioFilterGraph (int sample_rate, int channels);
        ~AudioFilterGraph ();
 
        void process (boost::shared_ptr<const AudioBuffers> audio);
@@ -40,6 +40,7 @@ protected:
 
 private:
        int _sample_rate;
+       int _channels;
        int64_t _channel_layout;
        AVFrame* _in_frame;
 };
index c2c06734af922e9f64e399201841d692bcaaf9a9..83ed458caee197de2be99a422c0be20a868c7227 100644 (file)
@@ -135,3 +135,33 @@ BOOST_AUTO_TEST_CASE (audio_analysis_test2)
        JobManager::instance()->add (job);
        wait_for_jobs ();
 }
+
+
+static bool done = false;
+
+static void
+analysis_finished ()
+{
+       done = true;
+}
+
+/* Test a case which was reported to throw an exception; analysing
+ * a 12-channel DCP's audio.
+ */
+BOOST_AUTO_TEST_CASE (audio_analysis_test3)
+{
+       shared_ptr<Film> film = new_test_film ("analyse_audio_test");
+       film->set_container (Ratio::from_id ("185"));
+       film->set_dcp_content_type (DCPContentType::from_isdcf_name ("TLR"));
+       film->set_name ("frobozz");
+
+       shared_ptr<SndfileContent> content (new SndfileContent (film, "test/data/white.wav"));
+       film->examine_and_add_content (content);
+       wait_for_jobs ();
+
+       film->set_audio_channels (12);
+       boost::signals2::connection connection;
+       JobManager::instance()->analyse_audio (film, film->playlist(), connection, boost::bind (&analysis_finished));
+       wait_for_jobs ();
+       BOOST_CHECK (done);
+}