#ifndef AUDIO_CHANNEL_RECEIVE_H_
#define AUDIO_CHANNEL_RECEIVE_H_
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/environment/environment.h"
#include "api/frame_transformer_interface.h"
#include "api/neteq/neteq_factory.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/rtp_rtcp/source/source_tracker.h"
namespace rtc {
class TimestampWrapAroundHandler;
}
namespace webrtc {
class AudioDeviceModule;
class FrameDecryptorInterface;
class PacketRouter;
class RateLimiter;
class ReceiveStatistics;
class RtpPacketReceived;
class RtpRtcp;
struct CallReceiveStatistics { … };
namespace voe {
class ChannelSendInterface;
class ChannelReceiveInterface : public RtpPacketSinkInterface { … };
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
const Environment& env,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
Transport* rtcp_send_transport,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
}
}
#endif