| Index: webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
|
| index 7676e90d9e4f9420f101aeb69b96daabe475680b..379293b748b043985488e180934c2dd5c9baf060 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
|
| @@ -35,7 +35,7 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
|
| int sample_rate_hz,
|
| int16_t* decoded,
|
| SpeechType* speech_type) {
|
| - RTC_DCHECK_EQ(sample_rate_hz, 16000);
|
| + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
|
| int16_t temp_type = 1; // Default is speech.
|
| size_t ret =
|
| WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
|
| @@ -53,6 +53,10 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
|
| return static_cast<int>(2 * encoded_len / Channels());
|
| }
|
|
|
| +int AudioDecoderG722::SampleRateHz() const {
|
| + return 16000;
|
| +}
|
| +
|
| size_t AudioDecoderG722::Channels() const {
|
| return 1;
|
| }
|
| @@ -74,7 +78,7 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
|
| int sample_rate_hz,
|
| int16_t* decoded,
|
| SpeechType* speech_type) {
|
| - RTC_DCHECK_EQ(sample_rate_hz, 16000);
|
| + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
|
| int16_t temp_type = 1; // Default is speech.
|
| // De-interleave the bit-stream into two separate payloads.
|
| uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
|
| @@ -100,6 +104,10 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
|
| return static_cast<int>(ret);
|
| }
|
|
|
| +int AudioDecoderG722Stereo::SampleRateHz() const {
|
| + return 16000;
|
| +}
|
| +
|
| size_t AudioDecoderG722Stereo::Channels() const {
|
| return 2;
|
| }
|
|
|