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"
#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