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+/*
+ * Copyright (c) 2012 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 MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+
+#include <math.h>
+
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+#define PCMA_AND_PCMU
+
+namespace webrtc {
+
+enum StereoMonoMode { kNotSet, kMono, kStereo };
+
+class TestPackStereo : public AudioPacketizationCallback {
+ public:
+ TestPackStereo();
+ ~TestPackStereo();
+
+ void RegisterReceiverACM(AudioCodingModule* acm);
+
+ int32_t SendData(AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size,
+ int64_t absolute_capture_timestamp_ms) override;
+
+ uint16_t payload_size();
+ uint32_t timestamp_diff();
+ void reset_payload_size();
+ void set_codec_mode(StereoMonoMode mode);
+ void set_lost_packet(bool lost);
+
+ private:
+ AudioCodingModule* receiver_acm_;
+ int16_t seq_no_;
+ uint32_t timestamp_diff_;
+ uint32_t last_in_timestamp_;
+ uint64_t total_bytes_;
+ int payload_size_;
+ StereoMonoMode codec_mode_;
+ // Simulate packet losses
+ bool lost_packet_;
+};
+
+class TestStereo {
+ public:
+ TestStereo();
+ ~TestStereo();
+
+ void Perform();
+
+ private:
+ // The default value of '-1' indicates that the registration is based only on
+ // codec name and a sampling frequncy matching is not required. This is useful
+ // for codecs which support several sampling frequency.
+ void RegisterSendCodec(char side,
+ char* codec_name,
+ int32_t samp_freq_hz,
+ int rate,
+ int pack_size,
+ int channels);
+
+ void Run(TestPackStereo* channel,
+ int in_channels,
+ int out_channels,
+ int percent_loss = 0);
+ void OpenOutFile(int16_t test_number);
+
+ std::unique_ptr<AudioCodingModule> acm_a_;
+ std::unique_ptr<AudioCodingModule> acm_b_;
+
+ TestPackStereo* channel_a2b_;
+
+ PCMFile* in_file_stereo_;
+ PCMFile* in_file_mono_;
+ PCMFile out_file_;
+ int16_t test_cntr_;
+ uint16_t pack_size_samp_;
+ uint16_t pack_size_bytes_;
+ int counter_;
+ char* send_codec_name_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_