iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs

webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc

Index: webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
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index 0000000000000000000000000000000000000000..500cfd1797bc951ec75bacb751daa3cfecf59c04
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+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
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+/*
+ *  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 "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+      (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
+      format.num_channels == 1) {
+    Config config;
+    config.sample_rate_hz = format.clockrate_hz;
+    if (config.sample_rate_hz == 16000) {
+      // For sample rate 16 kHz, optionally use 60 ms frames, instead of the
+      // default 30 ms.
+      const auto ptime_iter = format.parameters.find("ptime");
+      if (ptime_iter != format.parameters.end()) {
+        const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+        if (ptime && *ptime >= 60) {
+          config.frame_size_ms = 60;
+        }
+      }
+    }
+    return rtc::Optional<Config>(config);
+  } else {
+    return rtc::Optional<Config>();
+  }
+}
+
+void AudioEncoderIsacFloat::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (int sample_rate_hz : {16000, 32000}) {
+    const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1};
+    const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+    specs->push_back({fmt, info});
+  }
+}
+
+AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
+    const AudioEncoderIsacFloat::Config& config) {
+  RTC_DCHECK(config.IsOk());
+  constexpr int min_bitrate = 10000;
+  const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000;
+  const int default_bitrate = max_bitrate;
+  return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
+    const AudioEncoderIsacFloat::Config& config,
+    int payload_type) {
+  RTC_DCHECK(config.IsOk());
+  AudioEncoderIsacFloatImpl::Config c;
+  c.sample_rate_hz = config.sample_rate_hz;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  return rtc::MakeUnique<AudioEncoderIsacFloatImpl>(c);
+}
+
+}  // namespace webrtc