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

webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc

Index: webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c07957d9841d0a31e383a78625b914189554f42c
--- /dev/null
+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
@@ -0,0 +1,61 @@
+/*
+ *  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_fix.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&

[ossu, 2017/08/18 10:19:59]

Does this mean we'll eventually be removing CreateIsacConfig() in the
implementation? Don't like that we have the same functionality implemented twice
(or thrice, with the floating point variant as well). 

[kwiberg-webrtc, 2017/08/18 10:47:21]

Yes, because the code over there will no longer need to be able to parse
SdpAudioFormat.

+      format.clockrate_hz == 16000 && format.num_channels == 1) {
+    Config config;
+    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 AudioEncoderIsacFix::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ISAC", 16000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
+    AudioEncoderIsacFix::Config config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, 1, 32000, 10000, 32000};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
+    AudioEncoderIsacFix::Config config,
+    int payload_type) {
+  RTC_DCHECK(config.IsOk());
+  AudioEncoderIsacFixImpl::Config c;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  return rtc::MakeUnique<AudioEncoderIsacFixImpl>(c);
+}
+
+}  // namespace webrtc