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62238097c9
This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802}
100 lines
2.7 KiB
C++
100 lines
2.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
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#define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
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#include <math.h>
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#include <memory>
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#define PCMA_AND_PCMU
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namespace webrtc {
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enum StereoMonoMode { kNotSet, kMono, kStereo };
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class TestPackStereo : public AudioPacketizationCallback {
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public:
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TestPackStereo();
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~TestPackStereo();
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void RegisterReceiverACM(AudioCodingModule* acm);
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int32_t SendData(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms) override;
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uint16_t payload_size();
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uint32_t timestamp_diff();
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void reset_payload_size();
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void set_codec_mode(StereoMonoMode mode);
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void set_lost_packet(bool lost);
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private:
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AudioCodingModule* receiver_acm_;
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int16_t seq_no_;
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uint32_t timestamp_diff_;
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uint32_t last_in_timestamp_;
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uint64_t total_bytes_;
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int payload_size_;
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StereoMonoMode codec_mode_;
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// Simulate packet losses
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bool lost_packet_;
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};
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class TestStereo {
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public:
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TestStereo();
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~TestStereo();
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void Perform();
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private:
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// The default value of '-1' indicates that the registration is based only on
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// codec name and a sampling frequncy matching is not required. This is useful
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// for codecs which support several sampling frequency.
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void RegisterSendCodec(char side,
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char* codec_name,
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int32_t samp_freq_hz,
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int rate,
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int pack_size,
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int channels);
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void Run(TestPackStereo* channel,
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int in_channels,
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int out_channels,
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int percent_loss = 0);
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void OpenOutFile(int16_t test_number);
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std::unique_ptr<AudioCodingModule> acm_a_;
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std::unique_ptr<AudioCodingModule> acm_b_;
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TestPackStereo* channel_a2b_;
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PCMFile* in_file_stereo_;
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PCMFile* in_file_mono_;
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PCMFile out_file_;
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int16_t test_cntr_;
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uint16_t pack_size_samp_;
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uint16_t pack_size_bytes_;
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int counter_;
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char* send_codec_name_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
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