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https://github.com/mollyim/webrtc.git
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4175914f41
This reverts commit 48655cfdbf.
Reason for revert: failing upstream tests
Original change's description:
> Send absolute capture time through audio coding module.
>
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}
TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
109 lines
2.8 KiB
C++
109 lines
2.8 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_ENCODEDECODETEST_H_
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#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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#include <stdio.h>
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#include <string.h>
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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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#include "modules/audio_coding/test/RTPFile.h"
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#include "modules/include/module_common_types.h"
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namespace webrtc {
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#define MAX_INCOMING_PAYLOAD 8096
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// TestPacketization callback which writes the encoded payloads to file
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class TestPacketization : public AudioPacketizationCallback {
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public:
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TestPacketization(RTPStream* rtpStream, uint16_t frequency);
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~TestPacketization();
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int32_t SendData(const AudioFrameType frameType,
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const uint8_t payloadType,
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const uint32_t timeStamp,
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const uint8_t* payloadData,
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const size_t payloadSize) override;
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private:
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static void MakeRTPheader(uint8_t* rtpHeader,
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uint8_t payloadType,
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int16_t seqNo,
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uint32_t timeStamp,
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uint32_t ssrc);
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RTPStream* _rtpStream;
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int32_t _frequency;
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int16_t _seqNo;
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};
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class Sender {
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public:
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Sender();
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void Setup(AudioCodingModule* acm,
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RTPStream* rtpStream,
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std::string in_file_name,
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int in_sample_rate,
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int payload_type,
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SdpAudioFormat format);
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void Teardown();
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void Run();
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bool Add10MsData();
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protected:
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AudioCodingModule* _acm;
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private:
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PCMFile _pcmFile;
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AudioFrame _audioFrame;
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TestPacketization* _packetization;
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};
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class Receiver {
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public:
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Receiver();
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virtual ~Receiver() {}
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void Setup(AudioCodingModule* acm,
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RTPStream* rtpStream,
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std::string out_file_name,
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size_t channels,
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int file_num);
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void Teardown();
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void Run();
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virtual bool IncomingPacket();
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bool PlayoutData();
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private:
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PCMFile _pcmFile;
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int16_t* _playoutBuffer;
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uint16_t _playoutLengthSmpls;
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int32_t _frequency;
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bool _firstTime;
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protected:
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AudioCodingModule* _acm;
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uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
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RTPStream* _rtpStream;
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RTPHeader _rtpHeader;
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size_t _realPayloadSizeBytes;
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size_t _payloadSizeBytes;
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uint32_t _nextTime;
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};
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class EncodeDecodeTest {
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public:
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EncodeDecodeTest();
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void Perform();
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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