1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.cc b/third_party/libwebrtc/api/audio_codecs/audio_format.cc
new file mode 100644
index 0000000000..2a529a49ee
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_format.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2016 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 "api/audio_codecs/audio_format.h"
+
+#include <utility>
+
+#include "absl/strings/match.h"
+
+namespace webrtc {
+
+SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
+SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels)
+    : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               const Parameters& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(param) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               Parameters&& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(std::move(param)) {}
+
+bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
+  return absl::EqualsIgnoreCase(name, o.name) &&
+         clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
+}
+
+SdpAudioFormat::~SdpAudioFormat() = default;
+SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
+SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+
+bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+  return absl::EqualsIgnoreCase(a.name, b.name) &&
+         a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
+         a.parameters == b.parameters;
+}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int bitrate_bps)
+    : AudioCodecInfo(sample_rate_hz,
+                     num_channels,
+                     bitrate_bps,
+                     bitrate_bps,
+                     bitrate_bps) {}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int default_bitrate_bps,
+                               int min_bitrate_bps,
+                               int max_bitrate_bps)
+    : sample_rate_hz(sample_rate_hz),
+      num_channels(num_channels),
+      default_bitrate_bps(default_bitrate_bps),
+      min_bitrate_bps(min_bitrate_bps),
+      max_bitrate_bps(max_bitrate_bps) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_GE(min_bitrate_bps, 0);
+  RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+  RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
+}
+
+}  // namespace webrtc