我试图用XCode编译WebRTC（C ++）。最后也是唯一一个项目＆＃34; webrtc＆＃34;构建libwebtrc失败，错误：
/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'?
有两个名为＆＃34; AudioReceiveStream＆＃34;
的班级- 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_
第2课：
/*
* 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_
包含了webrtc :: AudioReceiveStream的标题，但仍然失败了。
如果我使用忍者来构建，则不会出现问题。代码未被修改。
XCode设置为： XCode settings
谢谢！
答案 0 :(得分：0)
在XCode“Build Setting”中，搜索“USE_HEADERMAP”并设置为NO
