1 files changed, 108 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/audio_send_stream.cc b/third_party/libwebrtc/call/audio_send_stream.cc
new file mode 100644
index 0000000000..a36050a9f7
--- /dev/null
+++ b/third_party/libwebrtc/call/audio_send_stream.cc
@@ -0,0 +1,108 @@
+/*
+ *  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