Index: webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc |
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc |
index 9757b4a0100674095353e9e20e43f986a1b3aba8..af164c4bb72b0b25525b3999f3e6887390a0b802 100644 |
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc |
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc |
@@ -16,6 +16,10 @@ namespace webrtc { |
void AudioDecoderPcmU::Reset() {} |
+int AudioDecoderPcmU::SampleRateHz() const { |
+ return 8000; |
+} |
+ |
size_t AudioDecoderPcmU::Channels() const { |
return num_channels_; |
} |
@@ -25,7 +29,7 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded, |
int sample_rate_hz, |
int16_t* decoded, |
SpeechType* speech_type) { |
- RTC_DCHECK_EQ(sample_rate_hz, 8000); |
+ RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); |
int16_t temp_type = 1; // Default is speech. |
size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type); |
*speech_type = ConvertSpeechType(temp_type); |
@@ -40,6 +44,10 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded, |
void AudioDecoderPcmA::Reset() {} |
+int AudioDecoderPcmA::SampleRateHz() const { |
+ return 8000; |
+} |
+ |
size_t AudioDecoderPcmA::Channels() const { |
return num_channels_; |
} |
@@ -49,7 +57,7 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded, |
int sample_rate_hz, |
int16_t* decoded, |
SpeechType* speech_type) { |
- RTC_DCHECK_EQ(sample_rate_hz, 8000); |
+ RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); |
int16_t temp_type = 1; // Default is speech. |
size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type); |
*speech_type = ConvertSpeechType(temp_type); |