Add aecdump support to audioproc_f
Originally landed here: https://codereview.webrtc.org/1409943002/
The transient suppression fix landed here: https://codereview.webrtc.org/1411423010/
TBR=mflodman
Review URL: https://codereview.webrtc.org/1432843002
Cr-Commit-Position: refs/heads/master@{#10722}
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index 8dae7d6..ac11bcd 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -13,6 +13,7 @@
#include <algorithm>
#include <cstdio>
#include <limits>
+#include <sstream>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
@@ -37,9 +38,17 @@
FILE* file_;
};
+std::string WavFile::FormatAsString() const {
+ std::ostringstream s;
+ s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
+ << ", Duration: "
+ << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
+ return s.str();
+}
+
WavReader::WavReader(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "rb")) {
- RTC_CHECK(file_handle_ && "Could not open wav file for reading.");
+ RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
ReadableWavFile readable(file_handle_);
WavFormat format;
@@ -96,7 +105,7 @@
num_channels_(num_channels),
num_samples_(0),
file_handle_(fopen(filename.c_str(), "wb")) {
- RTC_CHECK(file_handle_ && "Could not open wav file for writing.");
+ RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_));
diff --git a/webrtc/common_audio/wav_file.h b/webrtc/common_audio/wav_file.h
index 2eadd3f..42b0618 100644
--- a/webrtc/common_audio/wav_file.h
+++ b/webrtc/common_audio/wav_file.h
@@ -29,6 +29,9 @@
virtual int sample_rate() const = 0;
virtual int num_channels() const = 0;
virtual uint32_t num_samples() const = 0;
+
+ // Returns a human-readable string containing the audio format.
+ std::string FormatAsString() const;
};
// Simple C++ class for writing 16-bit PCM WAV files. All error handling is
diff --git a/webrtc/modules/audio_processing/audio_processing_tests.gypi b/webrtc/modules/audio_processing/audio_processing_tests.gypi
index 0314c69..523602b 100644
--- a/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -128,7 +128,11 @@
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
- 'sources': [ 'test/audioproc_float.cc', ],
+ 'sources': [
+ 'test/audio_file_processor.cc',
+ 'test/audio_file_processor.h',
+ 'test/audioproc_float.cc',
+ ],
},
{
'target_name': 'unpack_aecdump',
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
new file mode 100644
index 0000000..ca244d5
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -0,0 +1,177 @@
+/*
+ * Copyright (c) 2015 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 "webrtc/modules/audio_processing/test/audio_file_processor.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+using rtc::scoped_ptr;
+using rtc::CheckedDivExact;
+using std::vector;
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace webrtc {
+namespace {
+
+// Returns a StreamConfig corresponding to file.
+StreamConfig GetStreamConfig(const WavFile& file) {
+ return StreamConfig(file.sample_rate(), file.num_channels());
+}
+
+// Returns a ChannelBuffer corresponding to file.
+ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
+ return ChannelBuffer<float>(
+ CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond),
+ file.num_channels());
+}
+
+} // namespace
+
+WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
+ scoped_ptr<WavReader> in_file,
+ scoped_ptr<WavWriter> out_file)
+ : ap_(ap.Pass()),
+ in_buf_(GetChannelBuffer(*in_file)),
+ out_buf_(GetChannelBuffer(*out_file)),
+ input_config_(GetStreamConfig(*in_file)),
+ output_config_(GetStreamConfig(*out_file)),
+ buffer_reader_(in_file.Pass()),
+ buffer_writer_(out_file.Pass()) {}
+
+bool WavFileProcessor::ProcessChunk() {
+ if (!buffer_reader_.Read(&in_buf_)) {
+ return false;
+ }
+ {
+ const auto st = ScopedTimer(mutable_proc_time());
+ RTC_CHECK_EQ(kNoErr,
+ ap_->ProcessStream(in_buf_.channels(), input_config_,
+ output_config_, out_buf_.channels()));
+ }
+ buffer_writer_.Write(out_buf_);
+ return true;
+}
+
+AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
+ FILE* dump_file,
+ scoped_ptr<WavWriter> out_file)
+ : ap_(ap.Pass()),
+ dump_file_(dump_file),
+ out_buf_(GetChannelBuffer(*out_file)),
+ output_config_(GetStreamConfig(*out_file)),
+ buffer_writer_(out_file.Pass()) {
+ RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
+}
+
+AecDumpFileProcessor::~AecDumpFileProcessor() {
+ fclose(dump_file_);
+}
+
+bool AecDumpFileProcessor::ProcessChunk() {
+ Event event_msg;
+
+ // Continue until we process our first Stream message.
+ do {
+ if (!ReadMessageFromFile(dump_file_, &event_msg)) {
+ return false;
+ }
+
+ if (event_msg.type() == Event::INIT) {
+ RTC_CHECK(event_msg.has_init());
+ HandleMessage(event_msg.init());
+
+ } else if (event_msg.type() == Event::STREAM) {
+ RTC_CHECK(event_msg.has_stream());
+ HandleMessage(event_msg.stream());
+
+ } else if (event_msg.type() == Event::REVERSE_STREAM) {
+ RTC_CHECK(event_msg.has_reverse_stream());
+ HandleMessage(event_msg.reverse_stream());
+ }
+ } while (event_msg.type() != Event::STREAM);
+
+ return true;
+}
+
+void AecDumpFileProcessor::HandleMessage(const Init& msg) {
+ RTC_CHECK(msg.has_sample_rate());
+ RTC_CHECK(msg.has_num_input_channels());
+ RTC_CHECK(msg.has_num_reverse_channels());
+
+ in_buf_.reset(new ChannelBuffer<float>(
+ CheckedDivExact(msg.sample_rate(), kChunksPerSecond),
+ msg.num_input_channels()));
+ const int reverse_sample_rate = msg.has_reverse_sample_rate()
+ ? msg.reverse_sample_rate()
+ : msg.sample_rate();
+ reverse_buf_.reset(new ChannelBuffer<float>(
+ CheckedDivExact(reverse_sample_rate, kChunksPerSecond),
+ msg.num_reverse_channels()));
+ input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+ reverse_config_ =
+ StreamConfig(reverse_sample_rate, msg.num_reverse_channels());
+
+ const ProcessingConfig config = {
+ {input_config_, output_config_, reverse_config_, reverse_config_}};
+ RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
+}
+
+void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
+ RTC_CHECK(!msg.has_input_data());
+ RTC_CHECK_EQ(in_buf_->num_channels(), msg.input_channel_size());
+
+ for (int i = 0; i < msg.input_channel_size(); ++i) {
+ RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+ msg.input_channel(i).size());
+ std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+ msg.input_channel(i).size());
+ }
+ {
+ const auto st = ScopedTimer(mutable_proc_time());
+ RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
+ ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+ if (msg.has_keypress()) {
+ ap_->set_stream_key_pressed(msg.keypress());
+ }
+ RTC_CHECK_EQ(kNoErr,
+ ap_->ProcessStream(in_buf_->channels(), input_config_,
+ output_config_, out_buf_.channels()));
+ }
+
+ buffer_writer_.Write(out_buf_);
+}
+
+void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
+ RTC_CHECK(!msg.has_data());
+ RTC_CHECK_EQ(reverse_buf_->num_channels(), msg.channel_size());
+
+ for (int i = 0; i < msg.channel_size(); ++i) {
+ RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+ msg.channel(i).size());
+ std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
+ msg.channel(i).size());
+ }
+ {
+ const auto st = ScopedTimer(mutable_proc_time());
+ // TODO(ajm): This currently discards the processed output, which is needed
+ // for e.g. intelligibility enhancement.
+ RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
+ reverse_buf_->channels(), reverse_config_,
+ reverse_config_, reverse_buf_->channels()));
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.h b/webrtc/modules/audio_processing/test/audio_file_processor.h
new file mode 100644
index 0000000..a3153b2
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.h
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+
+// Holds a few statistics about a series of TickIntervals.
+struct TickIntervalStats {
+ TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
+ TickInterval sum;
+ TickInterval max;
+ TickInterval min;
+};
+
+// Interface for processing an input file with an AudioProcessing instance and
+// dumping the results to an output file.
+class AudioFileProcessor {
+ public:
+ static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
+
+ virtual ~AudioFileProcessor() {}
+
+ // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
+ // writes to the output file.
+ virtual bool ProcessChunk() = 0;
+
+ // Returns the execution time of all AudioProcessing calls.
+ const TickIntervalStats& proc_time() const { return proc_time_; }
+
+ protected:
+ // RAII class for execution time measurement. Updates the provided
+ // TickIntervalStats based on the time between ScopedTimer creation and
+ // leaving the enclosing scope.
+ class ScopedTimer {
+ public:
+ explicit ScopedTimer(TickIntervalStats* proc_time)
+ : proc_time_(proc_time), start_time_(TickTime::Now()) {}
+
+ ~ScopedTimer() {
+ TickInterval interval = TickTime::Now() - start_time_;
+ proc_time_->sum += interval;
+ proc_time_->max = std::max(proc_time_->max, interval);
+ proc_time_->min = std::min(proc_time_->min, interval);
+ }
+
+ private:
+ TickIntervalStats* const proc_time_;
+ TickTime start_time_;
+ };
+
+ TickIntervalStats* mutable_proc_time() { return &proc_time_; }
+
+ private:
+ TickIntervalStats proc_time_;
+};
+
+// Used to read from and write to WavFile objects.
+class WavFileProcessor final : public AudioFileProcessor {
+ public:
+ // Takes ownership of all parameters.
+ WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+ rtc::scoped_ptr<WavReader> in_file,
+ rtc::scoped_ptr<WavWriter> out_file);
+ virtual ~WavFileProcessor() {}
+
+ // Processes one chunk from the WAV input and writes to the WAV output.
+ bool ProcessChunk() override;
+
+ private:
+ rtc::scoped_ptr<AudioProcessing> ap_;
+
+ ChannelBuffer<float> in_buf_;
+ ChannelBuffer<float> out_buf_;
+ const StreamConfig input_config_;
+ const StreamConfig output_config_;
+ ChannelBufferWavReader buffer_reader_;
+ ChannelBufferWavWriter buffer_writer_;
+};
+
+// Used to read from an aecdump file and write to a WavWriter.
+class AecDumpFileProcessor final : public AudioFileProcessor {
+ public:
+ // Takes ownership of all parameters.
+ AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+ FILE* dump_file,
+ rtc::scoped_ptr<WavWriter> out_file);
+
+ virtual ~AecDumpFileProcessor();
+
+ // Processes messages from the aecdump file until the first Stream message is
+ // completed. Passes other data from the aecdump messages as appropriate.
+ bool ProcessChunk() override;
+
+ private:
+ void HandleMessage(const webrtc::audioproc::Init& msg);
+ void HandleMessage(const webrtc::audioproc::Stream& msg);
+ void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+
+ rtc::scoped_ptr<AudioProcessing> ap_;
+ FILE* dump_file_;
+
+ rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
+ rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
+ ChannelBuffer<float> out_buf_;
+ StreamConfig input_config_;
+ StreamConfig reverse_config_;
+ const StreamConfig output_config_;
+ ChannelBufferWavWriter buffer_writer_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 2697e51..3f1dc37 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -9,6 +9,7 @@
*/
#include <stdio.h>
+#include <iostream>
#include <sstream>
#include <string>
@@ -18,26 +19,28 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/trace_to_stderr.h"
-DEFINE_string(dump, "", "The name of the debug dump file to read from.");
-DEFINE_string(i, "", "The name of the input file to read from.");
-DEFINE_string(i_rev, "", "The name of the reverse input file to read from.");
-DEFINE_string(o, "out.wav", "Name of the output file to write to.");
-DEFINE_string(o_rev,
- "out_rev.wav",
- "Name of the reverse output file to write to.");
-DEFINE_int32(out_channels, 0, "Number of output channels. Defaults to input.");
-DEFINE_int32(out_sample_rate, 0,
- "Output sample rate in Hz. Defaults to input.");
+DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
+DEFINE_string(i, "", "Name of the capture input stream file to read from.");
+DEFINE_string(
+ o,
+ "out.wav",
+ "Name of the output file to write the processed capture stream to.");
+DEFINE_int32(out_channels, 1, "Number of output channels.");
+DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
DEFINE_string(mic_positions, "",
"Space delimited cartesian coordinates of microphones in meters. "
"The coordinates of each point are contiguous. "
"For a two element array: \"x1 y1 z1 x2 y2 z2\"");
-DEFINE_double(target_angle_degrees, 90, "The azimuth of the target in radians");
+DEFINE_double(
+ target_angle_degrees,
+ 90,
+ "The azimuth of the target in degrees. Only applies to beamforming.");
DEFINE_bool(aec, false, "Enable echo cancellation.");
DEFINE_bool(agc, false, "Enable automatic gain control.");
@@ -64,15 +67,6 @@
"All components are disabled by default. If any bi-directional components\n"
"are enabled, only debug dump files are permitted.";
-// Returns a StreamConfig corresponding to wav_file if it's non-nullptr.
-// Otherwise returns a default initialized StreamConfig.
-StreamConfig MakeStreamConfig(const WavFile* wav_file) {
- if (wav_file) {
- return {wav_file->sample_rate(), wav_file->num_channels()};
- }
- return {};
-}
-
} // namespace
int main(int argc, char* argv[]) {
@@ -84,48 +78,34 @@
"An input file must be specified with either -i or -dump.\n");
return 1;
}
- if (!FLAGS_dump.empty()) {
- fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
+ if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) {
+ fprintf(stderr, "-aec and -ie require a -dump file.\n");
+ return 1;
+ }
+ if (FLAGS_ie) {
+ fprintf(stderr,
+ "FIXME(ajm): The intelligibility enhancer output is not dumped.\n");
return 1;
}
test::TraceToStderr trace_to_stderr(true);
- WavReader in_file(FLAGS_i);
- // If the output format is uninitialized, use the input format.
- const int out_channels =
- FLAGS_out_channels ? FLAGS_out_channels : in_file.num_channels();
- const int out_sample_rate =
- FLAGS_out_sample_rate ? FLAGS_out_sample_rate : in_file.sample_rate();
- WavWriter out_file(FLAGS_o, out_sample_rate, out_channels);
-
Config config;
- config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
- config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
-
if (FLAGS_bf || FLAGS_all) {
- const size_t num_mics = in_file.num_channels();
- const std::vector<Point> array_geometry =
- ParseArrayGeometry(FLAGS_mic_positions, num_mics);
- RTC_CHECK_EQ(array_geometry.size(), num_mics);
-
+ if (FLAGS_mic_positions.empty()) {
+ fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
+ return 1;
+ }
config.Set<Beamforming>(new Beamforming(
- true, array_geometry,
+ true, ParseArrayGeometry(FLAGS_mic_positions),
SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
1.f)));
}
+ config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+ config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
- if (!FLAGS_dump.empty()) {
- RTC_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;
- }
- bool process_reverse = !FLAGS_i_rev.empty();
+ RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
- RTC_CHECK_EQ(kNoErr,
- ap->gain_control()->set_mode(GainControl::kFixedDigital));
RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
if (FLAGS_ns_level != -1) {
@@ -135,109 +115,38 @@
}
ap->set_stream_key_pressed(FLAGS_ts);
- printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
- FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
- printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
- FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
+ rtc::scoped_ptr<AudioFileProcessor> processor;
+ auto out_file = rtc_make_scoped_ptr(
+ new WavWriter(FLAGS_o, FLAGS_out_sample_rate, FLAGS_out_channels));
+ std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
+ if (FLAGS_dump.empty()) {
+ auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
+ std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
+ processor.reset(
+ new WavFileProcessor(ap.Pass(), in_file.Pass(), out_file.Pass()));
- ChannelBuffer<float> in_buf(
- rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
- in_file.num_channels());
- ChannelBuffer<float> out_buf(
- rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
- out_file.num_channels());
-
- std::vector<float> in_interleaved(in_buf.size());
- std::vector<float> out_interleaved(out_buf.size());
-
- rtc::scoped_ptr<WavReader> in_rev_file;
- rtc::scoped_ptr<WavWriter> out_rev_file;
- rtc::scoped_ptr<ChannelBuffer<float>> in_rev_buf;
- rtc::scoped_ptr<ChannelBuffer<float>> out_rev_buf;
- std::vector<float> in_rev_interleaved;
- std::vector<float> out_rev_interleaved;
- if (process_reverse) {
- in_rev_file.reset(new WavReader(FLAGS_i_rev));
- out_rev_file.reset(new WavWriter(FLAGS_o_rev, in_rev_file->sample_rate(),
- in_rev_file->num_channels()));
- printf("In rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
- FLAGS_i_rev.c_str(), in_rev_file->num_channels(),
- in_rev_file->sample_rate());
- printf("Out rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
- FLAGS_o_rev.c_str(), out_rev_file->num_channels(),
- out_rev_file->sample_rate());
- in_rev_buf.reset(new ChannelBuffer<float>(
- rtc::CheckedDivExact(in_rev_file->sample_rate(), kChunksPerSecond),
- in_rev_file->num_channels()));
- in_rev_interleaved.resize(in_rev_buf->size());
- out_rev_buf.reset(new ChannelBuffer<float>(
- rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond),
- out_rev_file->num_channels()));
- out_rev_interleaved.resize(out_rev_buf->size());
+ } else {
+ processor.reset(new AecDumpFileProcessor(
+ ap.Pass(), fopen(FLAGS_dump.c_str(), "rb"), out_file.Pass()));
}
- TickTime processing_start_time;
- TickInterval accumulated_time;
int num_chunks = 0;
-
- const auto input_config = MakeStreamConfig(&in_file);
- const auto output_config = MakeStreamConfig(&out_file);
- const auto reverse_input_config = MakeStreamConfig(in_rev_file.get());
- const auto reverse_output_config = MakeStreamConfig(out_rev_file.get());
-
- while (in_file.ReadSamples(in_interleaved.size(),
- &in_interleaved[0]) == in_interleaved.size()) {
- // Have logs display the file time rather than wallclock time.
+ while (processor->ProcessChunk()) {
trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
- FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(),
- &in_interleaved[0]);
- Deinterleave(&in_interleaved[0], in_buf.num_frames(),
- in_buf.num_channels(), in_buf.channels());
- if (process_reverse) {
- in_rev_file->ReadSamples(in_rev_interleaved.size(),
- in_rev_interleaved.data());
- FloatS16ToFloat(in_rev_interleaved.data(), in_rev_interleaved.size(),
- in_rev_interleaved.data());
- Deinterleave(in_rev_interleaved.data(), in_rev_buf->num_frames(),
- in_rev_buf->num_channels(), in_rev_buf->channels());
- }
-
- if (FLAGS_perf) {
- processing_start_time = TickTime::Now();
- }
- RTC_CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
- output_config, out_buf.channels()));
- if (process_reverse) {
- RTC_CHECK_EQ(kNoErr, ap->ProcessReverseStream(
- in_rev_buf->channels(), reverse_input_config,
- reverse_output_config, out_rev_buf->channels()));
- }
- if (FLAGS_perf) {
- accumulated_time += TickTime::Now() - processing_start_time;
- }
-
- Interleave(out_buf.channels(), out_buf.num_frames(),
- out_buf.num_channels(), &out_interleaved[0]);
- FloatToFloatS16(&out_interleaved[0], out_interleaved.size(),
- &out_interleaved[0]);
- out_file.WriteSamples(&out_interleaved[0], out_interleaved.size());
- if (process_reverse) {
- Interleave(out_rev_buf->channels(), out_rev_buf->num_frames(),
- out_rev_buf->num_channels(), out_rev_interleaved.data());
- FloatToFloatS16(out_rev_interleaved.data(), out_rev_interleaved.size(),
- out_rev_interleaved.data());
- out_rev_file->WriteSamples(out_rev_interleaved.data(),
- out_rev_interleaved.size());
- }
- num_chunks++;
+ ++num_chunks;
}
+
if (FLAGS_perf) {
- int64_t execution_time_ms = accumulated_time.Milliseconds();
- printf("\nExecution time: %.3f s\nFile time: %.2f s\n"
- "Time per chunk: %.3f ms\n",
- execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond,
- execution_time_ms * 1.f / num_chunks);
+ const auto& proc_time = processor->proc_time();
+ int64_t exec_time_us = proc_time.sum.Microseconds();
+ printf(
+ "\nExecution time: %.3f s, File time: %.2f s\n"
+ "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
+ exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
+ exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
+ 1.f * proc_time.min.Microseconds());
}
+
return 0;
}
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index d07db92..ae6b4dc 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -636,8 +636,8 @@
}
if (!raw_output) {
- // The WAV file needs to be reset every time, because it cant change
- // it's sample rate or number of channels.
+ // The WAV file needs to be reset every time, because it can't change
+ // its sample rate or number of channels.
output_wav_file.reset(new WavWriter(out_filename + ".wav",
output_sample_rate,
msg.num_output_channels()));
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 1b9ac3c..47bd314 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -31,6 +31,35 @@
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
+ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+ : file_(file.Pass()) {}
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+ RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+ interleaved_.resize(buffer->size());
+ if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+ interleaved_.size()) {
+ return false;
+ }
+
+ FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+ Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+ buffer->channels());
+ return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+ : file_(file.Pass()) {}
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+ RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+ interleaved_.resize(buffer.size());
+ Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+ &interleaved_[0]);
+ FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+ file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
@@ -92,28 +121,32 @@
case 2:
return AudioProcessing::kStereo;
default:
- assert(false);
+ RTC_CHECK(false);
return AudioProcessing::kMono;
}
}
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
- size_t num_mics) {
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
const std::vector<float> values = ParseList<float>(mic_positions);
- RTC_CHECK_EQ(values.size(), 3 * num_mics)
- << "Could not parse mic_positions or incorrect number of points.";
+ const size_t num_mics =
+ rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
+ RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
std::vector<Point> result;
result.reserve(num_mics);
for (size_t i = 0; i < values.size(); i += 3) {
- double x = values[i + 0];
- double y = values[i + 1];
- double z = values[i + 2];
- result.push_back(Point(x, y, z));
+ result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
}
return result;
}
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+ size_t num_mics) {
+ std::vector<Point> result = ParseArrayGeometry(mic_positions);
+ RTC_CHECK_EQ(result.size(), num_mics)
+ << "Could not parse mic_positions or incorrect number of points.";
+ return result;
+}
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index 75e4239..93a0138 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -43,6 +43,35 @@
RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
};
+// Reads ChannelBuffers from a provided WavReader.
+class ChannelBufferWavReader final {
+ public:
+ explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
+
+ // Reads data from the file according to the |buffer| format. Returns false if
+ // a full buffer can't be read from the file.
+ bool Read(ChannelBuffer<float>* buffer);
+
+ private:
+ rtc::scoped_ptr<WavReader> file_;
+ std::vector<float> interleaved_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
+};
+
+// Writes ChannelBuffers to a provided WavWriter.
+class ChannelBufferWavWriter final {
+ public:
+ explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
+ void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+ rtc::scoped_ptr<WavWriter> file_;
+ std::vector<float> interleaved_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
+};
+
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
@@ -118,6 +147,9 @@
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
size_t num_mics);
+// Same as above, but without the num_mics check for when it isn't available.
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/webrtc/system_wrappers/include/tick_util.h b/webrtc/system_wrappers/include/tick_util.h
index 6e3b05e..f8a5ef7 100644
--- a/webrtc/system_wrappers/include/tick_util.h
+++ b/webrtc/system_wrappers/include/tick_util.h
@@ -85,6 +85,7 @@
class TickInterval {
public:
TickInterval();
+ explicit TickInterval(int64_t interval);
int64_t Milliseconds() const;
int64_t Microseconds() const;
@@ -105,8 +106,6 @@
friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
private:
- explicit TickInterval(int64_t interval);
-
friend class TickTime;
friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
