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87e2d785a0
This cl deprecates the FrameType enum, and adds aliases AudioFrameType and VideoFrameType. After downstream usage is updated, the enums will be separated and be moved out of common_types.h. Bug: webrtc:6883 Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27011}
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(const AudioFrameType frame_type,
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const uint8_t payload_type,
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const uint32_t timestamp,
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const uint8_t* payload_data,
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const size_t payload_size,
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const RTPFragmentationHeader* fragmentation) 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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