Ожидаемое имя класса C++ с тем же именем класса при компиляции с XCode

Question

Я пытаюсь скомпилировать WebRTC (C++) с XCode. Последний и единственный проект "WebRTC" построить libwebtrc не удалось с ошибкой:

/Volumes/Data/webrtc/webrtc/src/webrtc/audio/audio_receive_stream.h:33:49: Expected class name 
/Volumes/Data/webrtc/webrtc/src/webrtc/audio/audio_receive_stream.h:36:28: No member named 'AudioReceiveStream' in namespace 'webrtc'; did you mean simply 'AudioReceiveStream'? 

Есть 2 класса под названием "AudioReceiveStream"

- Class 1: webrtc::AudioReceiveStream 
- Class 2: webrtc::internal::AudioReceiveStream 


/* 
* Copyright (c) 2015 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 WEBRTC_AUDIO_RECEIVE_STREAM_H_ 
#define WEBRTC_AUDIO_RECEIVE_STREAM_H_ 

#include <map> 
#include <memory> 
#include <string> 
#include <vector> 

#include "webrtc/base/scoped_ref_ptr.h" 
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h" 
#include "webrtc/common_types.h" 
#include "webrtc/config.h" 
#include "webrtc/transport.h" 
#include "webrtc/typedefs.h" 

namespace webrtc { 
class AudioSinkInterface; 

// WORK IN PROGRESS 
// This class is under development and is not yet intended for for use outside 
// of WebRtc/Libjingle. Please use the VoiceEngine API instead. 
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 

class AudioReceiveStream { 
public: 
    struct Stats { 
    uint32_t remote_ssrc = 0; 
    int64_t bytes_rcvd = 0; 
    uint32_t packets_rcvd = 0; 
    uint32_t packets_lost = 0; 
    float fraction_lost = 0.0f; 
    std::string codec_name; 
    uint32_t ext_seqnum = 0; 
    uint32_t jitter_ms = 0; 
    uint32_t jitter_buffer_ms = 0; 
    uint32_t jitter_buffer_preferred_ms = 0; 
    uint32_t delay_estimate_ms = 0; 
    int32_t audio_level = -1; 
    float expand_rate = 0.0f; 
    float speech_expand_rate = 0.0f; 
    float secondary_decoded_rate = 0.0f; 
    float accelerate_rate = 0.0f; 
    float preemptive_expand_rate = 0.0f; 
    int32_t decoding_calls_to_silence_generator = 0; 
    int32_t decoding_calls_to_neteq = 0; 
    int32_t decoding_normal = 0; 
    int32_t decoding_plc = 0; 
    int32_t decoding_cng = 0; 
    int32_t decoding_plc_cng = 0; 
    int64_t capture_start_ntp_time_ms = 0; 
    }; 

    struct Config { 
    std::string ToString() const; 

    // Receive-stream specific RTP settings. 
    struct Rtp { 
     std::string ToString() const; 

     // Synchronization source (stream identifier) to be received. 
     uint32_t remote_ssrc = 0; 

     // Sender SSRC used for sending RTCP (such as receiver reports). 
     uint32_t local_ssrc = 0; 

     // Enable feedback for send side bandwidth estimation. 
     // See 
     // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions 
     // for details. 
     bool transport_cc = false; 

     // See NackConfig for description. 
     NackConfig nack; 

     // RTP header extensions used for the received stream. 
     std::vector<RtpExtension> extensions; 
    } rtp; 

    Transport* rtcp_send_transport = nullptr; 

    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- 
    // level components. 
    // TODO(solenberg): Remove when VoiceEngine channels are created outside 
    // of Call. 
    int voe_channel_id = -1; 

    // Identifier for an A/V synchronization group. Empty string to disable. 
    // TODO(pbos): Synchronize streams in a sync group, not just one video 
    // stream to one audio stream. Tracked by issue webrtc:4762. 
    std::string sync_group; 

    // Decoders for every payload that we can receive. Call owns the 
    // AudioDecoder instances once the Config is submitted to 
    // Call::CreateReceiveStream(). 
    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. 
    std::map<uint8_t, AudioDecoder*> decoder_map; 

    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; 
    }; 

    // Starts stream activity. 
    // When a stream is active, it can receive, process and deliver packets. 
    virtual void Start() = 0; 
    // Stops stream activity. 
    // When a stream is stopped, it can't receive, process or deliver packets. 
    virtual void Stop() = 0; 

    virtual Stats GetStats() const = 0; 

    // Sets an audio sink that receives unmixed audio from the receive stream. 
    // Ownership of the sink is passed to the stream and can be used by the 
    // caller to do lifetime management (i.e. when the sink's dtor is called). 
    // Only one sink can be set and passing a null sink clears an existing one. 
    // NOTE: Audio must still somehow be pulled through AudioTransport for audio 
    // to stream through this sink. In practice, this happens if mixed audio 
    // is being pulled+rendered and/or if audio is being pulled for the purposes 
    // of feeding to the AEC. 
    virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0; 

    // Sets playback gain of the stream, applied when mixing, and thus after it 
    // is potentially forwarded to any attached AudioSinkInterface implementation. 
    virtual void SetGain(float gain) = 0; 

protected: 
    virtual ~AudioReceiveStream() {} 
}; 
} // namespace webrtc 

#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ 

Класс 2:

/* 
* Copyright (c) 2015 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 WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 

#include <memory> 

#include "webrtc/audio_receive_stream.h" 
#include "webrtc/audio_state.h" 
#include "webrtc/base/constructormagic.h" 
#include "webrtc/base/thread_checker.h" 
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 

namespace webrtc { 
class CongestionController; 
class RemoteBitrateEstimator; 
class RtcEventLog; 

namespace voe { 
class ChannelProxy; 
} // namespace voe 

namespace internal { 

class AudioReceiveStream final : public webrtc::AudioReceiveStream { 
public: 
    AudioReceiveStream(CongestionController* congestion_controller, 
        const webrtc::AudioReceiveStream::Config& config, 
        const rtc::scoped_refptr<webrtc::AudioState>& audio_state, 
        webrtc::RtcEventLog* event_log); 
    ~AudioReceiveStream() override; 

    // webrtc::AudioReceiveStream implementation. 
    void Start() override; 
    void Stop() override; 
    webrtc::AudioReceiveStream::Stats GetStats() const override; 
    void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; 
    void SetGain(float gain) override; 

    void SignalNetworkState(NetworkState state); 
    bool DeliverRtcp(const uint8_t* packet, size_t length); 
    bool DeliverRtp(const uint8_t* packet, 
        size_t length, 
        const PacketTime& packet_time); 
    const webrtc::AudioReceiveStream::Config& config() const; 

private: 
    VoiceEngine* voice_engine() const; 

    rtc::ThreadChecker thread_checker_; 
    RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; 
    const webrtc::AudioReceiveStream::Config config_; 
    rtc::scoped_refptr<webrtc::AudioState> audio_state_; 
    std::unique_ptr<RtpHeaderParser> rtp_header_parser_; 
    std::unique_ptr<voe::ChannelProxy> channel_proxy_; 

    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); 
}; 
} // namespace internal 
} // namespace webrtc 

#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 

Заголовок webrtc :: AudioReceiveStream включен, но все еще не удалось.

Если я использую ниндзя для сборки, проблем не возникает. Код не изменяется. установка

XCode как: XCode settings

ThankYou!

Answer 1

В XCode «Построить Настройка», поиск «USE_HEADERMAP» и установите NO