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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
88 lines
2.8 KiB
C++
88 lines
2.8 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/audio_send_stream.h"
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#include <string>
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namespace webrtc {
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AudioSendStream::Stats::Stats() = default;
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AudioSendStream::Stats::~Stats() = default;
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AudioSendStream::Config::Config(Transport* send_transport)
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: send_transport(send_transport) {}
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AudioSendStream::Config::~Config() = default;
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std::string AudioSendStream::Config::ToString() const {
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std::stringstream ss;
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ss << "{rtp: " << rtp.ToString();
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ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
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ss << ", voe_channel_id: " << voe_channel_id;
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ss << ", min_bitrate_bps: " << min_bitrate_bps;
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ss << ", max_bitrate_bps: " << max_bitrate_bps;
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ss << ", send_codec_spec: "
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<< (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
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ss << '}';
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return ss.str();
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}
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AudioSendStream::Config::Rtp::Rtp() = default;
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AudioSendStream::Config::Rtp::~Rtp() = default;
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std::string AudioSendStream::Config::Rtp::ToString() const {
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std::stringstream ss;
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ss << "{ssrc: " << ssrc;
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ss << ", extensions: [";
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for (size_t i = 0; i < extensions.size(); ++i) {
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ss << extensions[i].ToString();
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if (i != extensions.size() - 1) {
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ss << ", ";
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}
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}
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ss << ']';
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ss << ", nack: " << nack.ToString();
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ss << ", c_name: " << c_name;
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ss << '}';
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return ss.str();
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}
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AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
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int payload_type,
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const SdpAudioFormat& format)
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: payload_type(payload_type), format(format) {}
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AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
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std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
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std::stringstream ss;
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ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
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ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
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ss << ", cng_payload_type: "
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<< (cng_payload_type ? std::to_string(*cng_payload_type) : "<unset>");
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ss << ", payload_type: " << payload_type;
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ss << ", format: " << format;
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ss << '}';
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return ss.str();
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}
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bool AudioSendStream::Config::SendCodecSpec::operator==(
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const AudioSendStream::Config::SendCodecSpec& rhs) const {
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if (nack_enabled == rhs.nack_enabled &&
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transport_cc_enabled == rhs.transport_cc_enabled &&
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cng_payload_type == rhs.cng_payload_type &&
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payload_type == rhs.payload_type && format == rhs.format &&
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target_bitrate_bps == rhs.target_bitrate_bps) {
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return true;
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}
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return false;
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}
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} // namespace webrtc
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