
- Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
189 lines
7.1 KiB
C++
189 lines
7.1 KiB
C++
/*
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* Copyright (c) 2011 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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// Commandline tool to unpack audioproc debug files.
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//
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// The debug files are dumped as protobuf blobs. For analysis, it's necessary
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// to unpack the file into its component parts: audio and other data.
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#include <stdio.h>
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#include "gflags/gflags.h"
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#include "webrtc/audio_processing/debug.pb.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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#include "webrtc/typedefs.h"
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// TODO(andrew): unpack more of the data.
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DEFINE_string(input_file, "input.pcm", "The name of the input stream file.");
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DEFINE_string(float_input_file, "input.float",
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"The name of the float input stream file.");
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DEFINE_string(output_file, "ref_out.pcm",
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"The name of the reference output stream file.");
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DEFINE_string(float_output_file, "ref_out.float",
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"The name of the float reference output stream file.");
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DEFINE_string(reverse_file, "reverse.pcm",
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"The name of the reverse input stream file.");
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DEFINE_string(float_reverse_file, "reverse.float",
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"The name of the float reverse input stream file.");
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DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
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DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
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DEFINE_string(level_file, "level.int32", "The name of the level file.");
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DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
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DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
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DEFINE_bool(full, false,
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"Unpack the full set of files (normally not needed).");
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namespace webrtc {
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using audioproc::Event;
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using audioproc::ReverseStream;
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using audioproc::Stream;
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using audioproc::Init;
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void WriteData(const void* data, size_t size, FILE* file,
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const std::string& filename) {
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if (fwrite(data, size, 1, file) != 1) {
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printf("Error when writing to %s\n", filename.c_str());
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exit(1);
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}
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}
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int do_main(int argc, char* argv[]) {
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std::string program_name = argv[0];
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std::string usage = "Commandline tool to unpack audioproc debug files.\n"
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"Example usage:\n" + program_name + " debug_dump.pb\n";
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google::SetUsageMessage(usage);
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google::ParseCommandLineFlags(&argc, &argv, true);
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if (argc < 2) {
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printf("%s", google::ProgramUsage());
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return 1;
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}
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FILE* debug_file = OpenFile(argv[1], "rb");
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Event event_msg;
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int frame_count = 0;
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while (ReadMessageFromFile(debug_file, &event_msg)) {
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if (event_msg.type() == Event::REVERSE_STREAM) {
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if (!event_msg.has_reverse_stream()) {
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printf("Corrupt input file: ReverseStream missing.\n");
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return 1;
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}
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const ReverseStream msg = event_msg.reverse_stream();
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if (msg.has_data()) {
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static FILE* reverse_file = OpenFile(FLAGS_reverse_file, "wb");
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WriteData(msg.data().data(), msg.data().size(), reverse_file,
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FLAGS_reverse_file);
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} else if (msg.channel_size() > 0) {
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static FILE* float_reverse_file = OpenFile(FLAGS_float_reverse_file,
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"wb");
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// TODO(ajm): Interleave multiple channels.
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assert(msg.channel_size() == 1);
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WriteData(msg.channel(0).data(), msg.channel(0).size(),
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float_reverse_file, FLAGS_reverse_file);
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}
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} else if (event_msg.type() == Event::STREAM) {
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frame_count++;
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if (!event_msg.has_stream()) {
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printf("Corrupt input file: Stream missing.\n");
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return 1;
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}
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const Stream msg = event_msg.stream();
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if (msg.has_input_data()) {
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static FILE* input_file = OpenFile(FLAGS_input_file, "wb");
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WriteData(msg.input_data().data(), msg.input_data().size(),
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input_file, FLAGS_input_file);
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} else if (msg.input_channel_size() > 0) {
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static FILE* float_input_file = OpenFile(FLAGS_float_input_file, "wb");
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// TODO(ajm): Interleave multiple channels.
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assert(msg.input_channel_size() == 1);
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WriteData(msg.input_channel(0).data(), msg.input_channel(0).size(),
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float_input_file, FLAGS_float_input_file);
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}
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if (msg.has_output_data()) {
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static FILE* output_file = OpenFile(FLAGS_output_file, "wb");
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WriteData(msg.output_data().data(), msg.output_data().size(),
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output_file, FLAGS_output_file);
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} else if (msg.output_channel_size() > 0) {
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static FILE* float_output_file = OpenFile(FLAGS_float_output_file,
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"wb");
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// TODO(ajm): Interleave multiple channels.
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assert(msg.output_channel_size() == 1);
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WriteData(msg.output_channel(0).data(), msg.output_channel(0).size(),
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float_output_file, FLAGS_float_output_file);
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}
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if (FLAGS_full) {
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if (msg.has_delay()) {
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static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
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int32_t delay = msg.delay();
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WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
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}
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if (msg.has_drift()) {
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static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
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int32_t drift = msg.drift();
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WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
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}
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if (msg.has_level()) {
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static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
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int32_t level = msg.level();
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WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
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}
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if (msg.has_keypress()) {
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static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
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bool keypress = msg.keypress();
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WriteData(&keypress, sizeof(keypress), keypress_file,
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FLAGS_keypress_file);
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}
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}
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} else if (event_msg.type() == Event::INIT) {
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if (!event_msg.has_init()) {
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printf("Corrupt input file: Init missing.\n");
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return 1;
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}
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static FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
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const Init msg = event_msg.init();
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// These should print out zeros if they're missing.
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fprintf(settings_file, "Init at frame: %d\n", frame_count);
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fprintf(settings_file, " Sample rate: %d\n", msg.sample_rate());
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fprintf(settings_file, " Device sample rate: %d\n",
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msg.device_sample_rate());
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fprintf(settings_file, " Input channels: %d\n",
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msg.num_input_channels());
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fprintf(settings_file, " Output channels: %d\n",
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msg.num_output_channels());
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fprintf(settings_file, " Reverse channels: %d\n",
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msg.num_reverse_channels());
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fprintf(settings_file, "\n");
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}
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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::do_main(argc, argv);
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}
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