From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 7 Apr 2024 19:32:43 +0200
Subject: Adding upstream version 1:115.7.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../api/audio_codecs/g722/audio_encoder_g722.cc    | 74 ++++++++++++++++++++++
 1 file changed, 74 insertions(+)
 create mode 100644 third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc

(limited to 'third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc')

diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
new file mode 100644
index 0000000000..56a6c4da6a
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 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/g722/audio_encoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
+      format.clockrate_hz != 8000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderG722::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"G722", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
+    const AudioEncoderG722Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
+    const AudioEncoderG722Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
+}
+
+}  // namespace webrtc
-- 
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