10cbb4648f
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed. BUG=webrtc:6670 Review-Url: https://codereview.webrtc.org/2247213005 Cr-Commit-Position: refs/heads/master@{#14955}
109 lines
3.5 KiB
C++
109 lines
3.5 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 "webrtc/api/call/audio_send_stream.h"
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#include <string>
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namespace {
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std::string ToString(const webrtc::CodecInst& codec_inst) {
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std::stringstream ss;
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ss << "{pltype: " << codec_inst.pltype;
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ss << ", plname: \"" << codec_inst.plname << "\"";
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ss << ", plfreq: " << codec_inst.plfreq;
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ss << ", pacsize: " << codec_inst.pacsize;
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ss << ", channels: " << codec_inst.channels;
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ss << ", rate: " << codec_inst.rate;
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ss << '}';
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return ss.str();
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}
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} // namespace
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namespace webrtc {
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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)" : "nullptr");
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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: " << send_codec_spec.ToString();
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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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webrtc::CodecInst empty_inst = {0};
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codec_inst = empty_inst;
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codec_inst.pltype = -1;
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}
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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 << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
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ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
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ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
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ss << ", cng_payload_type: " << cng_payload_type;
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ss << ", cng_plfreq: " << cng_plfreq;
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ss << ", min_ptime: " << min_ptime_ms;
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ss << ", max_ptime: " << max_ptime_ms;
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ss << ", codec_inst: " << ::ToString(codec_inst);
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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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enable_codec_fec == rhs.enable_codec_fec &&
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enable_opus_dtx == rhs.enable_opus_dtx &&
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opus_max_playback_rate == rhs.opus_max_playback_rate &&
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cng_payload_type == rhs.cng_payload_type &&
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cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
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min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
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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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