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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 "call/audio_send_stream.h"
#include <stddef.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)
: send_transport(send_transport) {}
AudioSendStream::Config::~Config() = default;
std::string AudioSendStream::Config::ToString() const {
rtc::StringBuilder ss;
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
ss << ", min_bitrate_bps: " << min_bitrate_bps;
ss << ", max_bitrate_bps: " << max_bitrate_bps;
ss << ", has audio_network_adaptor_config: "
<< (audio_network_adaptor_config ? "true" : "false");
ss << ", has_dscp: " << (has_dscp ? "true" : "false");
ss << ", send_codec_spec: "
<< (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
ss << "}";
return ss.Release();
}
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;
if (!rid.empty()) {
ss << ", rid: " << rid;
}
if (!mid.empty()) {
ss << ", mid: " << mid;
}
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 << ", enable_non_sender_rtt: "
<< (enable_non_sender_rtt ? "true" : "false");
ss << ", cng_payload_type: "
<< (cng_payload_type ? rtc::ToString(*cng_payload_type) : "<unset>");
ss << ", red_payload_type: "
<< (red_payload_type ? rtc::ToString(*red_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 &&
enable_non_sender_rtt == rhs.enable_non_sender_rtt &&
cng_payload_type == rhs.cng_payload_type &&
red_payload_type == rhs.red_payload_type &&
payload_type == rhs.payload_type && format == rhs.format &&
target_bitrate_bps == rhs.target_bitrate_bps) {
return true;
}
return false;
}
} // namespace webrtc
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