4f08faae82
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files. TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change. Bug: webrtc:9719 Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28016}
103 lines
3.4 KiB
C++
103 lines
3.4 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 "call/audio_send_stream.h"
|
|
|
|
#include <stddef.h>
|
|
|
|
#include "rtc_base/string_encode.h"
|
|
#include "rtc_base/strings/audio_format_to_string.h"
|
|
#include "rtc_base/strings/string_builder.h"
|
|
|
|
namespace webrtc {
|
|
|
|
AudioSendStream::Stats::Stats() = default;
|
|
AudioSendStream::Stats::~Stats() = default;
|
|
|
|
AudioSendStream::Config::Config(
|
|
Transport* send_transport,
|
|
const MediaTransportConfig& media_transport_config)
|
|
: send_transport(send_transport),
|
|
media_transport_config(media_transport_config) {}
|
|
|
|
AudioSendStream::Config::Config(Transport* send_transport)
|
|
: Config(send_transport, MediaTransportConfig()) {}
|
|
|
|
AudioSendStream::Config::~Config() = default;
|
|
|
|
std::string AudioSendStream::Config::ToString() const {
|
|
char buf[1024];
|
|
rtc::SimpleStringBuilder ss(buf);
|
|
ss << "{rtp: " << rtp.ToString();
|
|
ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
|
|
ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
|
|
ss << ", media_transport_config: " << media_transport_config.DebugString();
|
|
ss << ", min_bitrate_bps: " << min_bitrate_bps;
|
|
ss << ", max_bitrate_bps: " << max_bitrate_bps;
|
|
ss << ", send_codec_spec: "
|
|
<< (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
AudioSendStream::Config::Rtp::Rtp() = default;
|
|
|
|
AudioSendStream::Config::Rtp::~Rtp() = default;
|
|
|
|
std::string AudioSendStream::Config::Rtp::ToString() const {
|
|
char buf[1024];
|
|
rtc::SimpleStringBuilder ss(buf);
|
|
ss << "{ssrc: " << ssrc;
|
|
ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false");
|
|
ss << ", extensions: [";
|
|
for (size_t i = 0; i < extensions.size(); ++i) {
|
|
ss << extensions[i].ToString();
|
|
if (i != extensions.size() - 1) {
|
|
ss << ", ";
|
|
}
|
|
}
|
|
ss << ']';
|
|
ss << ", c_name: " << c_name;
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
|
|
int payload_type,
|
|
const SdpAudioFormat& format)
|
|
: payload_type(payload_type), format(format) {}
|
|
AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
|
|
|
|
std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
|
|
char buf[1024];
|
|
rtc::SimpleStringBuilder ss(buf);
|
|
ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
|
|
ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
|
|
ss << ", cng_payload_type: "
|
|
<< (cng_payload_type ? rtc::ToString(*cng_payload_type) : "<unset>");
|
|
ss << ", payload_type: " << payload_type;
|
|
ss << ", format: " << rtc::ToString(format);
|
|
ss << '}';
|
|
return ss.str();
|
|
}
|
|
|
|
bool AudioSendStream::Config::SendCodecSpec::operator==(
|
|
const AudioSendStream::Config::SendCodecSpec& rhs) const {
|
|
if (nack_enabled == rhs.nack_enabled &&
|
|
transport_cc_enabled == rhs.transport_cc_enabled &&
|
|
cng_payload_type == rhs.cng_payload_type &&
|
|
payload_type == rhs.payload_type && format == rhs.format &&
|
|
target_bitrate_bps == rhs.target_bitrate_bps) {
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
} // namespace webrtc
|