mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00

This reverts commit 80c4cca491
.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
107 lines
3.3 KiB
C++
107 lines
3.3 KiB
C++
/*
|
|
* Copyright (c) 2016 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_NETEQ_TOOLS_NETEQ_TEST_H_
|
|
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
|
|
|
|
#include <map>
|
|
#include <memory>
|
|
#include <string>
|
|
#include <utility>
|
|
|
|
#include "modules/audio_coding/neteq/include/neteq.h"
|
|
#include "modules/audio_coding/neteq/tools/audio_sink.h"
|
|
#include "modules/audio_coding/neteq/tools/neteq_input.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
class NetEqTestErrorCallback {
|
|
public:
|
|
virtual ~NetEqTestErrorCallback() = default;
|
|
virtual void OnInsertPacketError(const NetEqInput::PacketData& packet) {}
|
|
virtual void OnGetAudioError() {}
|
|
};
|
|
|
|
class DefaultNetEqTestErrorCallback : public NetEqTestErrorCallback {
|
|
void OnInsertPacketError(const NetEqInput::PacketData& packet) override;
|
|
void OnGetAudioError() override;
|
|
};
|
|
|
|
class NetEqPostInsertPacket {
|
|
public:
|
|
virtual ~NetEqPostInsertPacket() = default;
|
|
virtual void AfterInsertPacket(const NetEqInput::PacketData& packet,
|
|
NetEq* neteq) = 0;
|
|
};
|
|
|
|
class NetEqGetAudioCallback {
|
|
public:
|
|
virtual ~NetEqGetAudioCallback() = default;
|
|
virtual void BeforeGetAudio(NetEq* neteq) = 0;
|
|
virtual void AfterGetAudio(int64_t time_now_ms,
|
|
const AudioFrame& audio_frame,
|
|
bool muted,
|
|
NetEq* neteq) = 0;
|
|
};
|
|
|
|
// Class that provides an input--output test for NetEq. The input (both packets
|
|
// and output events) is provided by a NetEqInput object, while the output is
|
|
// directed to an AudioSink object.
|
|
class NetEqTest {
|
|
public:
|
|
using DecoderMap = std::map<int, std::pair<NetEqDecoder, std::string> >;
|
|
|
|
struct ExternalDecoderInfo {
|
|
AudioDecoder* decoder;
|
|
NetEqDecoder codec;
|
|
std::string codec_name;
|
|
};
|
|
|
|
using ExtDecoderMap = std::map<int, ExternalDecoderInfo>;
|
|
|
|
struct Callbacks {
|
|
NetEqTestErrorCallback* error_callback = nullptr;
|
|
NetEqPostInsertPacket* post_insert_packet = nullptr;
|
|
NetEqGetAudioCallback* get_audio_callback = nullptr;
|
|
};
|
|
|
|
// Sets up the test with given configuration, codec mappings, input, ouput,
|
|
// and callback objects for error reporting.
|
|
NetEqTest(const NetEq::Config& config,
|
|
const DecoderMap& codecs,
|
|
const ExtDecoderMap& ext_codecs,
|
|
std::unique_ptr<NetEqInput> input,
|
|
std::unique_ptr<AudioSink> output,
|
|
Callbacks callbacks);
|
|
|
|
~NetEqTest() = default;
|
|
|
|
// Runs the test. Returns the duration of the produced audio in ms.
|
|
int64_t Run();
|
|
|
|
// Returns the statistics from NetEq.
|
|
NetEqNetworkStatistics SimulationStats();
|
|
NetEqLifetimeStatistics LifetimeStats() const;
|
|
|
|
private:
|
|
void RegisterDecoders(const DecoderMap& codecs);
|
|
void RegisterExternalDecoders(const ExtDecoderMap& codecs);
|
|
|
|
std::unique_ptr<NetEq> neteq_;
|
|
std::unique_ptr<NetEqInput> input_;
|
|
std::unique_ptr<AudioSink> output_;
|
|
Callbacks callbacks_;
|
|
int sample_rate_hz_;
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
|