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65438812ba
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
97 lines
2.8 KiB
C++
97 lines
2.8 KiB
C++
/*
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* Copyright (c) 2014 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_ACM2_ACM_RECEIVE_TEST_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
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#include <stddef.h> // for size_t
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#include <memory>
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#include <string>
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/scoped_refptr.h"
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#include "rtc_base/constructor_magic.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class AudioCodingModule;
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class AudioDecoder;
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namespace test {
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class AudioSink;
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class PacketSource;
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class AcmReceiveTestOldApi {
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public:
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enum NumOutputChannels : size_t {
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kArbitraryChannels = 0,
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kMonoOutput = 1,
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kStereoOutput = 2,
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kQuadOutput = 4
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};
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AcmReceiveTestOldApi(PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz,
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NumOutputChannels exptected_output_channels,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
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virtual ~AcmReceiveTestOldApi();
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// Registers the codecs with default parameters from ACM.
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void RegisterDefaultCodecs();
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// Registers codecs with payload types matching the pre-encoded NetEq test
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// files.
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void RegisterNetEqTestCodecs();
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// Runs the test and returns true if successful.
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void Run();
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AudioCodingModule* get_acm() { return acm_.get(); }
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protected:
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// Method is called after each block of output audio is received from ACM.
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virtual void AfterGetAudio() {}
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SimulatedClock clock_;
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std::unique_ptr<AudioCodingModule> acm_;
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PacketSource* packet_source_;
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AudioSink* audio_sink_;
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int output_freq_hz_;
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NumOutputChannels exptected_output_channels_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
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};
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// This test toggles the output frequency every |toggle_period_ms|. The test
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// starts with |output_freq_hz_1|. Except for the toggling, it does the same
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// thing as AcmReceiveTestOldApi.
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class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
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public:
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AcmReceiveTestToggleOutputFreqOldApi(
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PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz_1,
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int output_freq_hz_2,
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int toggle_period_ms,
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NumOutputChannels exptected_output_channels);
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protected:
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void AfterGetAudio() override;
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const int output_freq_hz_1_;
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const int output_freq_hz_2_;
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const int toggle_period_ms_;
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int64_t last_toggle_time_ms_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
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