20a4b3fb2a
These are the changes made to WebRtcVoiceEngine and surrounding code. It still contains some things that are inelegant, like how AudioCodecSpec and AudioFormatInfo is ferried around in SendCodecSpec. This should probably be resolved before landing. There are also a few test still that are disabled. They should be removed or fixed, as the case may be. I've put this CL up to get a better overview of the changes made and how reviewable they are. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2705093002 Cr-Commit-Position: refs/heads/master@{#17904}
89 lines
2.8 KiB
C++
89 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 "webrtc/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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