Unpacking aecdumps generates wav files
BUG=webrtc:3359
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7018 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/common_audio/wav_writer.h b/common_audio/wav_writer.h
index 45bcbac..0966727 100644
--- a/common_audio/wav_writer.h
+++ b/common_audio/wav_writer.h
@@ -33,13 +33,13 @@
// [-32768,32767], and there must be the previously specified number of
// interleaved channels.
void WriteSamples(const float* samples, size_t num_samples);
+ void WriteSamples(const int16_t* samples, size_t num_samples);
int sample_rate() const { return sample_rate_; }
int num_channels() const { return num_channels_; }
uint32_t num_samples() const { return num_samples_; }
private:
- void WriteSamples(const int16_t* samples, size_t num_samples);
void Close();
const int sample_rate_;
const int num_channels_;
diff --git a/modules/audio_processing/audio_processing_tests.gypi b/modules/audio_processing/audio_processing_tests.gypi
index 82aa7fd..99b80f2 100644
--- a/modules/audio_processing/audio_processing_tests.gypi
+++ b/modules/audio_processing/audio_processing_tests.gypi
@@ -41,6 +41,7 @@
'dependencies': [
'audioproc_debug_proto',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [ 'test/unpack.cc', ],
diff --git a/modules/audio_processing/test/unpack.cc b/modules/audio_processing/test/unpack.cc
index c90ba82..a225d58 100644
--- a/modules/audio_processing/test/unpack.cc
+++ b/modules/audio_processing/test/unpack.cc
@@ -14,25 +14,28 @@
// to unpack the file into its component parts: audio and other data.
#include <stdio.h>
+#include <limits>
#include "gflags/gflags.h"
#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/wav_writer.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
// TODO(andrew): unpack more of the data.
DEFINE_string(input_file, "input.pcm", "The name of the input stream file.");
-DEFINE_string(float_input_file, "input.float",
- "The name of the float input stream file.");
+DEFINE_string(input_wav_file, "input.wav",
+ "The name of the WAV input stream file.");
DEFINE_string(output_file, "ref_out.pcm",
"The name of the reference output stream file.");
-DEFINE_string(float_output_file, "ref_out.float",
- "The name of the float reference output stream file.");
+DEFINE_string(output_wav_file, "ref_out.wav",
+ "The name of the WAV reference output stream file.");
DEFINE_string(reverse_file, "reverse.pcm",
"The name of the reverse input stream file.");
-DEFINE_string(float_reverse_file, "reverse.float",
- "The name of the float reverse input stream file.");
+DEFINE_string(reverse_wav_file, "reverse.wav",
+ "The name of the WAV reverse input stream file.");
DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
DEFINE_string(level_file, "level.int32", "The name of the level file.");
@@ -40,6 +43,7 @@
DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
DEFINE_bool(full, false,
"Unpack the full set of files (normally not needed).");
+DEFINE_bool(pcm, false, "Write to PCM instead of WAV file.");
namespace webrtc {
@@ -48,6 +52,36 @@
using audioproc::Stream;
using audioproc::Init;
+class PcmFile {
+ public:
+ PcmFile(const std::string& filename)
+ : file_handle_(fopen(filename.c_str(), "wb")) {}
+
+ ~PcmFile() {
+ fclose(file_handle_);
+ }
+
+ void WriteSamples(const int16_t* samples, size_t num_samples) {
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert samples to little-endian when writing to PCM file"
+#endif
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+ }
+
+ void WriteSamples(const float* samples, size_t num_samples) {
+ static const size_t kChunksize = 4096 / sizeof(uint16_t);
+ for (size_t i = 0; i < num_samples; i += kChunksize) {
+ int16_t isamples[kChunksize];
+ const size_t chunk = std::min(kChunksize, num_samples - i);
+ RoundToInt16(samples + i, chunk, isamples);
+ WriteSamples(isamples, chunk);
+ }
+ }
+
+ private:
+ FILE* file_handle_;
+};
+
void WriteData(const void* data, size_t size, FILE* file,
const std::string& filename) {
if (fwrite(data, size, 1, file) != 1) {
@@ -56,6 +90,40 @@
}
}
+void WriteIntData(const int16_t* data,
+ size_t length,
+ WavFile* wav_file,
+ PcmFile* pcm_file) {
+ if (wav_file) {
+ wav_file->WriteSamples(data, length);
+ }
+ if (pcm_file) {
+ pcm_file->WriteSamples(data, length);
+ }
+}
+
+void WriteFloatData(const float* const* data,
+ size_t samples_per_channel,
+ int num_channels,
+ WavFile* wav_file,
+ PcmFile* pcm_file) {
+ size_t length = num_channels * samples_per_channel;
+ scoped_ptr<float[]> buffer(new float[length]);
+ Interleave(data, samples_per_channel, num_channels, buffer.get());
+ // TODO(aluebs): Use ScaleToInt16Range() from audio_util
+ for (size_t i = 0; i < length; ++i) {
+ buffer[i] = buffer[i] > 0 ?
+ buffer[i] * std::numeric_limits<int16_t>::max() :
+ -buffer[i] * std::numeric_limits<int16_t>::min();
+ }
+ if (wav_file) {
+ wav_file->WriteSamples(buffer.get(), length);
+ }
+ if (pcm_file) {
+ pcm_file->WriteSamples(buffer.get(), length);
+ }
+}
+
int do_main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage = "Commandline tool to unpack audioproc debug files.\n"
@@ -72,7 +140,16 @@
Event event_msg;
int frame_count = 0;
-while (ReadMessageFromFile(debug_file, &event_msg)) {
+ int num_input_channels = 0;
+ int num_output_channels = 0;
+ int num_reverse_channels = 0;
+ scoped_ptr<WavFile> reverse_wav_file;
+ scoped_ptr<WavFile> input_wav_file;
+ scoped_ptr<WavFile> output_wav_file;
+ scoped_ptr<PcmFile> reverse_pcm_file;
+ scoped_ptr<PcmFile> input_pcm_file;
+ scoped_ptr<PcmFile> output_pcm_file;
+ while (ReadMessageFromFile(debug_file, &event_msg)) {
if (event_msg.type() == Event::REVERSE_STREAM) {
if (!event_msg.has_reverse_stream()) {
printf("Corrupt input file: ReverseStream missing.\n");
@@ -81,17 +158,20 @@
const ReverseStream msg = event_msg.reverse_stream();
if (msg.has_data()) {
- static FILE* reverse_file = OpenFile(FLAGS_reverse_file, "wb");
- WriteData(msg.data().data(), msg.data().size(), reverse_file,
- FLAGS_reverse_file);
-
+ WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
+ msg.data().size() / sizeof(int16_t),
+ reverse_wav_file.get(),
+ reverse_pcm_file.get());
} else if (msg.channel_size() > 0) {
- static FILE* float_reverse_file = OpenFile(FLAGS_float_reverse_file,
- "wb");
- // TODO(ajm): Interleave multiple channels.
- assert(msg.channel_size() == 1);
- WriteData(msg.channel(0).data(), msg.channel(0).size(),
- float_reverse_file, FLAGS_reverse_file);
+ scoped_ptr<const float*[]> data(new const float*[num_reverse_channels]);
+ for (int i = 0; i < num_reverse_channels; ++i) {
+ data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
+ }
+ WriteFloatData(data.get(),
+ msg.channel(0).size() / sizeof(float),
+ num_reverse_channels,
+ reverse_wav_file.get(),
+ reverse_pcm_file.get());
}
} else if (event_msg.type() == Event::STREAM) {
frame_count++;
@@ -102,30 +182,38 @@
const Stream msg = event_msg.stream();
if (msg.has_input_data()) {
- static FILE* input_file = OpenFile(FLAGS_input_file, "wb");
- WriteData(msg.input_data().data(), msg.input_data().size(),
- input_file, FLAGS_input_file);
-
+ WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
+ msg.input_data().size() / sizeof(int16_t),
+ input_wav_file.get(),
+ input_pcm_file.get());
} else if (msg.input_channel_size() > 0) {
- static FILE* float_input_file = OpenFile(FLAGS_float_input_file, "wb");
- // TODO(ajm): Interleave multiple channels.
- assert(msg.input_channel_size() == 1);
- WriteData(msg.input_channel(0).data(), msg.input_channel(0).size(),
- float_input_file, FLAGS_float_input_file);
+ scoped_ptr<const float*[]> data(new const float*[num_input_channels]);
+ for (int i = 0; i < num_input_channels; ++i) {
+ data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
+ }
+ WriteFloatData(data.get(),
+ msg.input_channel(0).size() / sizeof(float),
+ num_input_channels,
+ input_wav_file.get(),
+ input_pcm_file.get());
}
if (msg.has_output_data()) {
- static FILE* output_file = OpenFile(FLAGS_output_file, "wb");
- WriteData(msg.output_data().data(), msg.output_data().size(),
- output_file, FLAGS_output_file);
-
+ WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
+ msg.output_data().size() / sizeof(int16_t),
+ output_wav_file.get(),
+ output_pcm_file.get());
} else if (msg.output_channel_size() > 0) {
- static FILE* float_output_file = OpenFile(FLAGS_float_output_file,
- "wb");
- // TODO(ajm): Interleave multiple channels.
- assert(msg.output_channel_size() == 1);
- WriteData(msg.output_channel(0).data(), msg.output_channel(0).size(),
- float_output_file, FLAGS_float_output_file);
+ scoped_ptr<const float*[]> data(new const float*[num_output_channels]);
+ for (int i = 0; i < num_output_channels; ++i) {
+ data[i] =
+ reinterpret_cast<const float*>(msg.output_channel(i).data());
+ }
+ WriteFloatData(data.get(),
+ msg.output_channel(0).size() / sizeof(float),
+ num_output_channels,
+ output_wav_file.get(),
+ output_pcm_file.get());
}
if (FLAGS_full) {
@@ -164,15 +252,51 @@
const Init msg = event_msg.init();
// These should print out zeros if they're missing.
fprintf(settings_file, "Init at frame: %d\n", frame_count);
- fprintf(settings_file, " Sample rate: %d\n", msg.sample_rate());
- fprintf(settings_file, " Input channels: %d\n",
- msg.num_input_channels());
- fprintf(settings_file, " Output channels: %d\n",
- msg.num_output_channels());
- fprintf(settings_file, " Reverse channels: %d\n",
- msg.num_reverse_channels());
+ int input_sample_rate = msg.sample_rate();
+ fprintf(settings_file, " Input sample rate: %d\n", input_sample_rate);
+ int output_sample_rate = msg.output_sample_rate();
+ fprintf(settings_file, " Output sample rate: %d\n", output_sample_rate);
+ int reverse_sample_rate = msg.reverse_sample_rate();
+ fprintf(settings_file,
+ " Reverse sample rate: %d\n",
+ reverse_sample_rate);
+ num_input_channels = msg.num_input_channels();
+ fprintf(settings_file, " Input channels: %d\n", num_input_channels);
+ num_output_channels = msg.num_output_channels();
+ fprintf(settings_file, " Output channels: %d\n", num_output_channels);
+ num_reverse_channels = msg.num_reverse_channels();
+ fprintf(settings_file, " Reverse channels: %d\n", num_reverse_channels);
fprintf(settings_file, "\n");
+
+ if (reverse_sample_rate == 0) {
+ reverse_sample_rate = input_sample_rate;
+ }
+ if (output_sample_rate == 0) {
+ output_sample_rate = input_sample_rate;
+ }
+
+ if (FLAGS_pcm) {
+ if (!reverse_pcm_file.get()) {
+ reverse_pcm_file.reset(new PcmFile(FLAGS_reverse_file));
+ }
+ if (!input_pcm_file.get()) {
+ input_pcm_file.reset(new PcmFile(FLAGS_input_file));
+ }
+ if (!output_pcm_file.get()) {
+ output_pcm_file.reset(new PcmFile(FLAGS_output_file));
+ }
+ } else {
+ reverse_wav_file.reset(new WavFile(FLAGS_reverse_wav_file,
+ reverse_sample_rate,
+ num_reverse_channels));
+ input_wav_file.reset(new WavFile(FLAGS_input_wav_file,
+ input_sample_rate,
+ num_input_channels));
+ output_wav_file.reset(new WavFile(FLAGS_output_wav_file,
+ output_sample_rate,
+ num_output_channels));
+ }
}
}
