lib/ffmpeg/Filter: add MakeAudioBuffer{Source,Sink}()

src/lib/ffmpeg/Filter.cxx

@@ -0,0 +1,75 @@
+/*
+ * Copyright 2003-2019 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program 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,
+ * 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.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "Filter.hxx"
+#include "SampleFormat.hxx"
+#include "AudioFormat.hxx"
+#include "util/RuntimeError.hxx"
+
+#include <stdio.h>
+
+namespace Ffmpeg {
+
+static const auto &
+RequireFilterByName(const char *name)
+{
+	const auto *filter = avfilter_get_by_name(name);
+	if (filter == nullptr)
+		throw FormatRuntimeError("No such FFmpeg filter: '%s'", name);
+
+	return *filter;
+}
+
+FilterContext
+FilterContext::MakeAudioBufferSource(AudioFormat &audio_format,
+				     AVFilterGraph &graph_ctx)
+{
+	AVSampleFormat src_format = ToFfmpegSampleFormat(audio_format.format);
+	if (src_format == AV_SAMPLE_FMT_NONE) {
+		switch (audio_format.format) {
+		case SampleFormat::S24_P32:
+			audio_format.format = SampleFormat::S32;
+			src_format = AV_SAMPLE_FMT_S32;
+			break;
+
+		default:
+			audio_format.format = SampleFormat::S16;
+			src_format = AV_SAMPLE_FMT_S16;
+			break;
+		}
+	}
+
+	char abuffer_args[256];
+	sprintf(abuffer_args,
+		"sample_rate=%u:sample_fmt=%s:channels=%u:time_base=1/%u",
+		audio_format.sample_rate,
+		av_get_sample_fmt_name(src_format),
+		audio_format.channels,
+		audio_format.sample_rate);
+
+	return {RequireFilterByName("abuffer"), "abuffer", abuffer_args, nullptr, graph_ctx};
+}
+
+FilterContext
+FilterContext::MakeAudioBufferSink(AVFilterGraph &graph_ctx)
+{
+	return {RequireFilterByName("abuffersink"), "abuffersink", graph_ctx};
+}
+
+} // namespace Ffmpeg

src/lib/ffmpeg/Filter.hxx

@@ -28,6 +28,8 @@ extern "C" {
 
 #include <utility>
 
+struct AudioFormat;
+
 namespace Ffmpeg {
 
 class FilterInOut {
@@ -106,6 +108,20 @@ public:
 		return *this;
 	}
 
+	/**
+	 * Create an "abuffer" filter.
+	 *
+	 * @param the input audio format; may be modified by the
+	 * function to ask the caller to do format conversion
+	 */
+	static FilterContext MakeAudioBufferSource(AudioFormat &audio_format,
+						   AVFilterGraph &graph_ctx);
+
+	/**
+	 * Create an "abuffersink" filter.
+	 */
+	static FilterContext MakeAudioBufferSink(AVFilterGraph &graph_ctx);
+
 	auto &operator*() noexcept {
 		return *context;
 	}

src/lib/ffmpeg/meson.build

@@ -17,6 +17,11 @@ if not enable_ffmpeg
   subdir_done()
 endif
 
+ffmpeg_sources = []
+if libavfilter_dep.found()
+  ffmpeg_sources += 'Filter.cxx'
+endif
+
 ffmpeg = static_library(
   'ffmpeg',
   'Init.cxx',
@@ -24,6 +29,7 @@ ffmpeg = static_library(
   'LogCallback.cxx',
   'Error.cxx',
   'Domain.cxx',
+  ffmpeg_sources,
   include_directories: inc,
   dependencies: [
     libavformat_dep,