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Unified Diff: webrtc/modules/audio_processing/test/audio_file_processor.h

Issue 1432843002: Add aecdump support to audioproc_f (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add used includes Created 5 years, 1 month ago
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Index: webrtc/modules/audio_processing/test/audio_file_processor.h
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 0000000000000000000000000000000000000000..a3153b2244cb57b6edc67ad233ebc55501d135be
--- /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_

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