blob: ca244d550fed05248c5f650d93c09afb36dadc25 [file] [log] [blame]
/*
* 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