| Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
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| diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
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| index e4d58ae4d40e536fe25051167ea4401655206185..ebe4d6a0e264607c6266ac119fade2ec2e21fb5d 100644
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| --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
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| +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
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| @@ -12,9 +12,15 @@
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|  #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
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|  
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|  #include "webrtc/base/checks.h"
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| +#include "webrtc/base/string_to_number.h"
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|  #include "webrtc/common_types.h"
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|  
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|  namespace webrtc {
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| +namespace {  // NOLINT (not a "regular" header file)
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| +int GetIsacMaxBitrate(int clockrate_hz) {
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| +  return (clockrate_hz == 32000) ? 56000 : 32000;
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| +}
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| +}  // namespace
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|  
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|  template <typename T>
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|  typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
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| @@ -33,6 +39,33 @@ typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
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|  }
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|  
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|  template <typename T>
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| +typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
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| +    int payload_type,
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| +    const SdpAudioFormat& format) {
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| +  typename AudioEncoderIsacT<T>::Config config;
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| +  config.payload_type = payload_type;
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| +  config.sample_rate_hz = format.clockrate_hz;
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| +
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| +  // We only support different frame sizes at 16000 Hz.
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| +  if (config.sample_rate_hz == 16000) {
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| +    auto ptime_iter = format.parameters.find("ptime");
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| +    if (ptime_iter != format.parameters.end()) {
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| +      auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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| +      if (ptime && *ptime >= 60) {
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| +        config.frame_size_ms = 60;
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| +      } else {
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| +        config.frame_size_ms = 30;
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| +      }
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| +    }
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| +  }
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| +
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| +  // Set the default bitrate for ISAC to the maximum bitrate allowed at this
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| +  // clockrate. At this point, adaptive mode is not used by WebRTC.
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| +  config.bit_rate = GetIsacMaxBitrate(format.clockrate_hz);
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| +  return config;
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| +}
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| +
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| +template <typename T>
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|  bool AudioEncoderIsacT<T>::Config::IsOk() const {
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|    if (max_bit_rate < 32000 && max_bit_rate != -1)
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|      return false;
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| @@ -73,6 +106,25 @@ AudioEncoderIsacT<T>::AudioEncoderIsacT(
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|      : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
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|  
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|  template <typename T>
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| +AudioEncoderIsacT<T>::AudioEncoderIsacT(int payload_type,
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| +                                        const SdpAudioFormat& format)
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| +    : AudioEncoderIsacT(CreateIsacConfig<T>(payload_type, format)) {}
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| +
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| +template <typename T>
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| +rtc::Optional<AudioCodecInfo> AudioEncoderIsacT<T>::QueryAudioEncoder(
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| +    const SdpAudioFormat& format) {
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| +  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
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| +    Config config = CreateIsacConfig<T>(0, format);
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| +    if (config.IsOk()) {
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| +      return rtc::Optional<AudioCodecInfo>(
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| +          {config.sample_rate_hz, 1, config.bit_rate, 10000,
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| +           GetIsacMaxBitrate(format.clockrate_hz)});
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| +    }
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| +  }
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| +  return rtc::Optional<AudioCodecInfo>();
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| +}
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| +
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| +template <typename T>
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|  AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
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|    RTC_CHECK_EQ(0, T::Free(isac_state_));
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|  }
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| 
 |