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

webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc

+/*
+ *  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"
+
+namespace webrtc {
+
+rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(

[ossu, 2017/08/15 14:04:19]

This one should use (or at least be equivalent to) CreateIsacConfig, so that it
picks up ptime properly (and possibly the ibitrate and maxbitrate parameters,
should we add in support for them).

[kwiberg-webrtc, 2017/08/17 13:32:40]

Done.

+    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;
+    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