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* 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.
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/call/transport.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioSinkInterface;
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See:
class AudioReceiveStream {
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;
rtc::Optional<int> codec_payload_type;
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;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
double total_output_energy = 0.0;
// See description of "totalSamplesReceived" in the WebRTC stats spec:
uint64_t total_samples_received = 0;
// See description of "totalSamplesDuration" in the WebRTC stats spec:
double total_output_duration = 0.0;
// See description of "concealedSamples" in the WebRTC stats spec:
uint64_t concealed_samples = 0;
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;
float secondary_decoded_rate = 0.0f;
float secondary_discarded_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;
int32_t decoding_muted_output = 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
// 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;
// Decoder specifications for every payload type that we can receive.
std::map<int, SdpAudioFormat> 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;
// TODO(solenberg): Remove, once AudioMonitor is gone.
virtual int GetOutputLevel() 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;
virtual std::vector<RtpSource> GetSources() const = 0;
virtual ~AudioReceiveStream() {}
} // namespace webrtc