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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}
82 lines
2.3 KiB
C++
82 lines
2.3 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_TESTALLCODECS_H_
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#define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_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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namespace webrtc {
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class TestPack : public AudioPacketizationCallback {
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public:
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TestPack();
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~TestPack();
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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) override;
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size_t payload_size();
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uint32_t timestamp_diff();
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void reset_payload_size();
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private:
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AudioCodingModule* receiver_acm_;
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uint16_t sequence_number_;
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uint8_t payload_data_[60 * 32 * 2 * 2];
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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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size_t payload_size_;
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};
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class TestAllCodecs {
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public:
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TestAllCodecs();
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~TestAllCodecs();
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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 frequency matching is not required.
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// This is useful for codecs which support several sampling frequency.
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// Note! Only mono mode is tested in this test.
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void RegisterSendCodec(char side,
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char* codec_name,
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int32_t sampling_freq_hz,
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int rate,
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int packet_size,
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size_t extra_byte);
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void Run(TestPack* channel);
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void OpenOutFile(int 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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TestPack* channel_a_to_b_;
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PCMFile infile_a_;
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PCMFile outfile_b_;
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int test_count_;
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int packet_size_samples_;
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size_t packet_size_bytes_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
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