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; |
} |