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
This commit is contained in:
andrew@webrtc.org
parent
cf6d0b64ef
commit
08df9b2841
@ -35,6 +35,16 @@
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],
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'sources': [ 'test/process_test.cc', ],
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},
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{
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'target_name': 'audioproc_f',
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'type': 'executable',
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'dependencies': [
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'audio_processing',
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'audioproc_debug_proto',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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],
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'sources': [ 'test/audioproc_float.cc', ],
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},
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{
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'target_name': 'unpack_aecdump',
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'type': 'executable',
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132
webrtc/modules/audio_processing/test/audioproc_float.cc
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132
webrtc/modules/audio_processing/test/audioproc_float.cc
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@ -0,0 +1,132 @@
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include "gflags/gflags.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/wav_file.h"
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#include "webrtc/modules/audio_processing/channel_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/test/test_utils.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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DEFINE_string(dump, "", "The name of the debug dump file to read from.");
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DEFINE_string(c, "", "The name of the capture input file to read from.");
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DEFINE_string(o, "out.wav", "Name of the capture output file to write to.");
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DEFINE_int32(o_channels, 0, "Number of output channels. Defaults to input.");
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DEFINE_int32(o_sample_rate, 0, "Output sample rate in Hz. Defaults to input.");
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DEFINE_bool(aec, false, "Enable echo cancellation.");
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DEFINE_bool(agc, false, "Enable automatic gain control.");
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DEFINE_bool(hpf, false, "Enable high-pass filtering.");
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DEFINE_bool(ns, false, "Enable noise suppression.");
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DEFINE_bool(ts, false, "Enable transient suppression.");
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DEFINE_bool(all, false, "Enable all components.");
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DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3].");
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static const int kChunksPerSecond = 100;
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static const char kUsage[] =
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"Command-line tool to run audio processing on WAV files. Accepts either\n"
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"an input capture WAV file or protobuf debug dump and writes to an output\n"
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"WAV file.\n"
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"\n"
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"All components are disabled by default. If any bi-directional components\n"
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"are enabled, only debug dump files are permitted.";
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namespace webrtc {
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int main(int argc, char* argv[]) {
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{
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const std::string program_name = argv[0];
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const std::string usage = kUsage;
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google::SetUsageMessage(usage);
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}
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google::ParseCommandLineFlags(&argc, &argv, true);
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if (!((FLAGS_c == "") ^ (FLAGS_dump == ""))) {
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fprintf(stderr,
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"An input file must be specified with either -c or -dump.\n");
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return 1;
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}
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if (FLAGS_dump != "") {
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fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
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return 1;
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}
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WavReader c_file(FLAGS_c);
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// If the output format is uninitialized, use the input format.
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int o_channels = FLAGS_o_channels;
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if (!o_channels)
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o_channels = c_file.num_channels();
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int o_sample_rate = FLAGS_o_sample_rate;
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if (!o_sample_rate)
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o_sample_rate = c_file.sample_rate();
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WavWriter o_file(FLAGS_o, o_sample_rate, o_channels);
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printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
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FLAGS_c.c_str(), c_file.num_channels(), c_file.sample_rate());
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printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
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FLAGS_o.c_str(), o_file.num_channels(), o_file.sample_rate());
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Config config;
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config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
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scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
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if (FLAGS_dump != "") {
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CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
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} else if (FLAGS_aec) {
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fprintf(stderr, "-aec requires a -dump file.\n");
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return -1;
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}
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CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
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CHECK_EQ(kNoErr, ap->gain_control()->set_mode(GainControl::kFixedDigital));
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CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
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CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
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if (FLAGS_ns_level != -1)
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CHECK_EQ(kNoErr, ap->noise_suppression()->set_level(
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static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
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ChannelBuffer<float> c_buf(c_file.sample_rate() / kChunksPerSecond,
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c_file.num_channels());
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ChannelBuffer<float> o_buf(o_file.sample_rate() / kChunksPerSecond,
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o_file.num_channels());
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const size_t c_length = static_cast<size_t>(c_buf.length());
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scoped_ptr<float[]> c_interleaved(new float[c_length]);
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scoped_ptr<float[]> o_interleaved(new float[o_buf.length()]);
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while (c_file.ReadSamples(c_length, c_interleaved.get()) == c_length) {
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FloatS16ToFloat(c_interleaved.get(), c_length, c_interleaved.get());
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Deinterleave(c_interleaved.get(), c_buf.samples_per_channel(),
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c_buf.num_channels(), c_buf.channels());
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CHECK_EQ(kNoErr,
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ap->ProcessStream(c_buf.channels(),
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c_buf.samples_per_channel(),
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c_file.sample_rate(),
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LayoutFromChannels(c_buf.num_channels()),
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o_file.sample_rate(),
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LayoutFromChannels(o_buf.num_channels()),
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o_buf.channels()));
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Interleave(o_buf.channels(), o_buf.samples_per_channel(),
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o_buf.num_channels(), o_interleaved.get());
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FloatToFloatS16(o_interleaved.get(), o_buf.length(), o_interleaved.get());
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o_file.WriteSamples(o_interleaved.get(), o_buf.length());
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}
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return 0;
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}
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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return webrtc::main(argc, argv);
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}
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