Add a manageable command-line tool for AudioProcessing.
This is the start of a replacement for the venerable and unwieldly
process_test.cc (aka audioproc). It will be limited to:
- Reading WAV or aecdebug protobuf files.
- Calling the float AudioProcessing interface.
- Requiring aecdebug files for running bi-directional stream
components (e.g. AEC).
This initial version only handles WAV files.
R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7918 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/audio_processing_tests.gypi b/webrtc/modules/audio_processing/audio_processing_tests.gypi
index 627e669..7c36be7 100644
--- a/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -36,6 +36,16 @@
'sources': [ 'test/process_test.cc', ],
},
{
+ 'target_name': 'audioproc_f',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_processing',
+ 'audioproc_debug_proto',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [ 'test/audioproc_float.cc', ],
+ },
+ {
'target_name': 'unpack_aecdump',
'type': 'executable',
'dependencies': [
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
new file mode 100644
index 0000000..6286b5a
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -0,0 +1,132 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+DEFINE_string(dump, "", "The name of the debug dump file to read from.");
+DEFINE_string(c, "", "The name of the capture input file to read from.");
+DEFINE_string(o, "out.wav", "Name of the capture output file to write to.");
+DEFINE_int32(o_channels, 0, "Number of output channels. Defaults to input.");
+DEFINE_int32(o_sample_rate, 0, "Output sample rate in Hz. Defaults to input.");
+
+DEFINE_bool(aec, false, "Enable echo cancellation.");
+DEFINE_bool(agc, false, "Enable automatic gain control.");
+DEFINE_bool(hpf, false, "Enable high-pass filtering.");
+DEFINE_bool(ns, false, "Enable noise suppression.");
+DEFINE_bool(ts, false, "Enable transient suppression.");
+DEFINE_bool(all, false, "Enable all components.");
+
+DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3].");
+
+static const int kChunksPerSecond = 100;
+static const char kUsage[] =
+ "Command-line tool to run audio processing on WAV files. Accepts either\n"
+ "an input capture WAV file or protobuf debug dump and writes to an output\n"
+ "WAV file.\n"
+ "\n"
+ "All components are disabled by default. If any bi-directional components\n"
+ "are enabled, only debug dump files are permitted.";
+
+namespace webrtc {
+
+int main(int argc, char* argv[]) {
+ {
+ const std::string program_name = argv[0];
+ const std::string usage = kUsage;
+ google::SetUsageMessage(usage);
+ }
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ if (!((FLAGS_c == "") ^ (FLAGS_dump == ""))) {
+ fprintf(stderr,
+ "An input file must be specified with either -c or -dump.\n");
+ return 1;
+ }
+ if (FLAGS_dump != "") {
+ fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
+ return 1;
+ }
+
+ WavReader c_file(FLAGS_c);
+ // If the output format is uninitialized, use the input format.
+ int o_channels = FLAGS_o_channels;
+ if (!o_channels)
+ o_channels = c_file.num_channels();
+ int o_sample_rate = FLAGS_o_sample_rate;
+ if (!o_sample_rate)
+ o_sample_rate = c_file.sample_rate();
+ WavWriter o_file(FLAGS_o, o_sample_rate, o_channels);
+
+ printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+ FLAGS_c.c_str(), c_file.num_channels(), c_file.sample_rate());
+ printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
+ FLAGS_o.c_str(), o_file.num_channels(), o_file.sample_rate());
+
+ Config config;
+ config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+ scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
+ if (FLAGS_dump != "") {
+ CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
+ } else if (FLAGS_aec) {
+ fprintf(stderr, "-aec requires a -dump file.\n");
+ return -1;
+ }
+ CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
+ CHECK_EQ(kNoErr, ap->gain_control()->set_mode(GainControl::kFixedDigital));
+ CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
+ CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
+ if (FLAGS_ns_level != -1)
+ CHECK_EQ(kNoErr, ap->noise_suppression()->set_level(
+ static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
+
+ ChannelBuffer<float> c_buf(c_file.sample_rate() / kChunksPerSecond,
+ c_file.num_channels());
+ ChannelBuffer<float> o_buf(o_file.sample_rate() / kChunksPerSecond,
+ o_file.num_channels());
+
+ const size_t c_length = static_cast<size_t>(c_buf.length());
+ scoped_ptr<float[]> c_interleaved(new float[c_length]);
+ scoped_ptr<float[]> o_interleaved(new float[o_buf.length()]);
+ while (c_file.ReadSamples(c_length, c_interleaved.get()) == c_length) {
+ FloatS16ToFloat(c_interleaved.get(), c_length, c_interleaved.get());
+ Deinterleave(c_interleaved.get(), c_buf.samples_per_channel(),
+ c_buf.num_channels(), c_buf.channels());
+
+ CHECK_EQ(kNoErr,
+ ap->ProcessStream(c_buf.channels(),
+ c_buf.samples_per_channel(),
+ c_file.sample_rate(),
+ LayoutFromChannels(c_buf.num_channels()),
+ o_file.sample_rate(),
+ LayoutFromChannels(o_buf.num_channels()),
+ o_buf.channels()));
+
+ Interleave(o_buf.channels(), o_buf.samples_per_channel(),
+ o_buf.num_channels(), o_interleaved.get());
+ FloatToFloatS16(o_interleaved.get(), o_buf.length(), o_interleaved.get());
+ o_file.WriteSamples(o_interleaved.get(), o_buf.length());
+ }
+
+ return 0;
+}
+
+} // namespace webrtc
+
+int main(int argc, char* argv[]) {
+ return webrtc::main(argc, argv);
+}
