Index: webrtc/modules/utility/source/coder.cc |
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc |
index f2ae43eb108a939f261ae17646a05c26fd307643..f72d03b887de9356df67d44bb20f7d8963f868c5 100644 |
--- a/webrtc/modules/utility/source/coder.cc |
+++ b/webrtc/modules/utility/source/coder.cc |
@@ -54,7 +54,7 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) { |
return 0; |
} |
-int32_t AudioCoder::Decode(AudioFrame& decoded_audio, |
+int32_t AudioCoder::Decode(AudioFrame* decoded_audio, |
uint32_t samp_freq_hz, |
const int8_t* incoming_payload, |
size_t payload_length) { |
@@ -68,22 +68,22 @@ int32_t AudioCoder::Decode(AudioFrame& decoded_audio, |
} |
bool muted; |
int32_t ret = |
- acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted); |
+ acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted); |
RTC_DCHECK(!muted); |
return ret; |
} |
-int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio, |
- uint16_t& samp_freq_hz) { |
+int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio, |
+ uint16_t samp_freq_hz) { |
bool muted; |
- int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted); |
+ int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted); |
RTC_DCHECK(!muted); |
return ret; |
} |
int32_t AudioCoder::Encode(const AudioFrame& audio, |
int8_t* encoded_data, |
- size_t& encoded_length_in_bytes) { |
+ size_t* encoded_length_in_bytes) { |
// Fake a timestamp in case audio doesn't contain a correct timestamp. |
// Make a local copy of the audio frame since audio is const |
AudioFrame audio_frame; |
@@ -98,7 +98,7 @@ int32_t AudioCoder::Encode(const AudioFrame& audio, |
return -1; |
} |
encoded_data_ = encoded_data; |
- encoded_length_in_bytes = encoded_length_in_bytes_; |
+ *encoded_length_in_bytes = encoded_length_in_bytes_; |
return 0; |
} |