0946a56023
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1271006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.2 KiB
C++
119 lines
3.2 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TEST_STEREO_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TEST_STEREO_H_
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#include <math.h>
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#include "ACMTest.h"
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#include "Channel.h"
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#include "PCMFile.h"
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namespace webrtc {
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enum StereoMonoMode {
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kNotSet,
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kMono,
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kStereo
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};
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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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virtual int32_t SendData(const FrameType 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 uint16_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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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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uint16_t 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 : public ACMTest {
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public:
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TestStereo(int test_mode);
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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, char* codec_name,
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int32_t samp_freq_hz, int rate, int pack_size,
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int channels, int payload_type);
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void Run(TestPackStereo* channel, int in_channels, 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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void DisplaySendReceiveCodec();
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int32_t SendData(const FrameType 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 uint16_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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int test_mode_;
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AudioCodingModule* acm_a_;
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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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// Payload types for stereo codecs and CNG
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int g722_pltype_;
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int l16_8khz_pltype_;
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int l16_16khz_pltype_;
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int l16_32khz_pltype_;
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int pcma_pltype_;
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int pcmu_pltype_;
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int celt_pltype_;
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int opus_pltype_;
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int cn_8khz_pltype_;
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int cn_16khz_pltype_;
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int cn_32khz_pltype_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TEST_STEREO_H_
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