Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #include "webrtc/call/audio_send_stream.h"
      12             : 
      13             : #include <string>
      14             : 
      15             : namespace {
      16             : 
      17           0 : std::string ToString(const webrtc::CodecInst& codec_inst) {
      18           0 :   std::stringstream ss;
      19           0 :   ss << "{pltype: " << codec_inst.pltype;
      20           0 :   ss << ", plname: \"" << codec_inst.plname << "\"";
      21           0 :   ss << ", plfreq: " << codec_inst.plfreq;
      22           0 :   ss << ", pacsize: " << codec_inst.pacsize;
      23           0 :   ss << ", channels: " << codec_inst.channels;
      24           0 :   ss << ", rate: " << codec_inst.rate;
      25           0 :   ss << '}';
      26           0 :   return ss.str();
      27             : }
      28             : }  // namespace
      29             : 
      30             : namespace webrtc {
      31             : 
      32             : AudioSendStream::Stats::Stats() = default;
      33             : AudioSendStream::Stats::~Stats() = default;
      34             : 
      35           0 : AudioSendStream::Config::Config(Transport* send_transport)
      36           0 :     : send_transport(send_transport) {}
      37             : 
      38             : AudioSendStream::Config::~Config() = default;
      39             : 
      40           0 : std::string AudioSendStream::Config::ToString() const {
      41           0 :   std::stringstream ss;
      42           0 :   ss << "{rtp: " << rtp.ToString();
      43           0 :   ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
      44           0 :   ss << ", voe_channel_id: " << voe_channel_id;
      45           0 :   ss << ", min_bitrate_bps: " << min_bitrate_bps;
      46           0 :   ss << ", max_bitrate_bps: " << max_bitrate_bps;
      47           0 :   ss << ", send_codec_spec: " << send_codec_spec.ToString();
      48           0 :   ss << '}';
      49           0 :   return ss.str();
      50             : }
      51             : 
      52             : AudioSendStream::Config::Rtp::Rtp() = default;
      53             : 
      54             : AudioSendStream::Config::Rtp::~Rtp() = default;
      55             : 
      56           0 : std::string AudioSendStream::Config::Rtp::ToString() const {
      57           0 :   std::stringstream ss;
      58           0 :   ss << "{ssrc: " << ssrc;
      59           0 :   ss << ", extensions: [";
      60           0 :   for (size_t i = 0; i < extensions.size(); ++i) {
      61           0 :     ss << extensions[i].ToString();
      62           0 :     if (i != extensions.size() - 1) {
      63           0 :       ss << ", ";
      64             :     }
      65             :   }
      66           0 :   ss << ']';
      67           0 :   ss << ", nack: " << nack.ToString();
      68           0 :   ss << ", c_name: " << c_name;
      69           0 :   ss << '}';
      70           0 :   return ss.str();
      71             : }
      72             : 
      73           0 : AudioSendStream::Config::SendCodecSpec::SendCodecSpec() {
      74           0 :   webrtc::CodecInst empty_inst = {0};
      75           0 :   codec_inst = empty_inst;
      76           0 :   codec_inst.pltype = -1;
      77           0 : }
      78             : 
      79           0 : std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
      80           0 :   std::stringstream ss;
      81           0 :   ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
      82           0 :   ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
      83           0 :   ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
      84           0 :   ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
      85           0 :   ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
      86           0 :   ss << ", cng_payload_type: " << cng_payload_type;
      87           0 :   ss << ", cng_plfreq: " << cng_plfreq;
      88           0 :   ss << ", min_ptime: " << min_ptime_ms;
      89           0 :   ss << ", max_ptime: " << max_ptime_ms;
      90           0 :   ss << ", codec_inst: " << ::ToString(codec_inst);
      91           0 :   ss << '}';
      92           0 :   return ss.str();
      93             : }
      94             : 
      95           0 : bool AudioSendStream::Config::SendCodecSpec::operator==(
      96             :     const AudioSendStream::Config::SendCodecSpec& rhs) const {
      97           0 :   if (nack_enabled == rhs.nack_enabled &&
      98           0 :       transport_cc_enabled == rhs.transport_cc_enabled &&
      99           0 :       enable_codec_fec == rhs.enable_codec_fec &&
     100           0 :       enable_opus_dtx == rhs.enable_opus_dtx &&
     101           0 :       opus_max_playback_rate == rhs.opus_max_playback_rate &&
     102           0 :       cng_payload_type == rhs.cng_payload_type &&
     103           0 :       cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
     104           0 :       min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
     105           0 :     return true;
     106             :   }
     107           0 :   return false;
     108             : }
     109             : }  // namespace webrtc