webrtc_m130/modules/audio_coding/acm2/acm_receive_test.h

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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
#include <stddef.h> // for size_t
#include <memory>
#include <string>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/scoped_refptr.h"
#include "rtc_base/constructor_magic.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class AudioCodingModule;
class AudioDecoder;
namespace test {
class AudioSink;
class PacketSource;
class AcmReceiveTestOldApi {
public:
enum NumOutputChannels : size_t {
kArbitraryChannels = 0,
kMonoOutput = 1,
Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa. Reason for revert: Order of initialization of global static strings. Original change's description: > Reland of https://webrtc-review.googlesource.com/c/src/+/114883 > > The difference to the original is new bitexactness strings AND > global static file string constants. 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. > > Bug: webrtc:8649 > Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2 > Reviewed-on: https://webrtc-review.googlesource.com/c/123387 > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26774} TBR=aleloi@webrtc.org,ossu@webrtc.org Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8649 Reviewed-on: https://webrtc-review.googlesource.com/c/123580 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26777}
2019-02-20 15:17:39 +00:00
kStereoOutput = 2
};
AcmReceiveTestOldApi(PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
virtual ~AcmReceiveTestOldApi();
// Registers the codecs with default parameters from ACM.
void RegisterDefaultCodecs();
// Registers codecs with payload types matching the pre-encoded NetEq test
// files.
void RegisterNetEqTestCodecs();
// Runs the test and returns true if successful.
void Run();
AudioCodingModule* get_acm() { return acm_.get(); }
protected:
// Method is called after each block of output audio is received from ACM.
virtual void AfterGetAudio() {}
SimulatedClock clock_;
std::unique_ptr<AudioCodingModule> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;
NumOutputChannels exptected_output_channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
};
// This test toggles the output frequency every |toggle_period_ms|. The test
// starts with |output_freq_hz_1|. Except for the toggling, it does the same
// thing as AcmReceiveTestOldApi.
class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
public:
AcmReceiveTestToggleOutputFreqOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz_1,
int output_freq_hz_2,
int toggle_period_ms,
NumOutputChannels exptected_output_channels);
protected:
void AfterGetAudio() override;
const int output_freq_hz_1_;
const int output_freq_hz_2_;
const int toggle_period_ms_;
int64_t last_toggle_time_ms_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_