c7a8b08a7c
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
73 lines
1.9 KiB
C++
73 lines
1.9 KiB
C++
/*
|
|
* 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 "webrtc/audio/audio_send_stream.h"
|
|
|
|
#include <string>
|
|
|
|
#include "webrtc/base/checks.h"
|
|
#include "webrtc/base/logging.h"
|
|
|
|
namespace webrtc {
|
|
std::string AudioSendStream::Config::Rtp::ToString() const {
|
|
std::stringstream ss;
|
|
ss << "{ssrc: " << ssrc;
|
|
ss << ", extensions: [";
|
|
for (size_t i = 0; i < extensions.size(); ++i) {
|
|
ss << extensions[i].ToString();
|
|
if (i != extensions.size() - 1)
|
|
ss << ", ";
|
|
}
|
|
ss << ']';
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
std::string AudioSendStream::Config::ToString() const {
|
|
std::stringstream ss;
|
|
ss << "{rtp: " << rtp.ToString();
|
|
ss << ", voe_channel_id: " << voe_channel_id;
|
|
// TODO(solenberg): Encoder config.
|
|
ss << ", cng_payload_type: " << cng_payload_type;
|
|
ss << ", red_payload_type: " << red_payload_type;
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
namespace internal {
|
|
AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config)
|
|
: config_(config) {
|
|
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
|
|
RTC_DCHECK(config.voe_channel_id != -1);
|
|
}
|
|
|
|
AudioSendStream::~AudioSendStream() {
|
|
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
|
|
}
|
|
|
|
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
|
|
return webrtc::AudioSendStream::Stats();
|
|
}
|
|
|
|
void AudioSendStream::Start() {
|
|
}
|
|
|
|
void AudioSendStream::Stop() {
|
|
}
|
|
|
|
void AudioSendStream::SignalNetworkState(NetworkState state) {
|
|
}
|
|
|
|
bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
|
|
return false;
|
|
}
|
|
} // namespace internal
|
|
} // namespace webrtc
|