Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
index 0e50fdc82b6d3a15dec05c41dbce6573a6a1960c..a599e291d479b0ba3926e0d8d7e14e5dbaf5f75c 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
@@ -100,6 +100,16 @@ |
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_)); |
} |
+size_t AudioEncoderOpus::MaxEncodedBytes() const { |
+ // Calculate the number of bytes we expect the encoder to produce, |
+ // then multiply by two to give a wide margin for error. |
+ const size_t bytes_per_millisecond = |
+ static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); |
+ const size_t approx_encoded_bytes = |
+ Num10msFramesPerPacket() * 10 * bytes_per_millisecond; |
+ return 2 * approx_encoded_bytes; |
+} |
+ |
int AudioEncoderOpus::SampleRateHz() const { |
return kSampleRateHz; |
} |
@@ -188,7 +198,7 @@ |
RTC_CHECK_EQ(input_buffer_.size(), |
Num10msFramesPerPacket() * SamplesPer10msFrame()); |
- const size_t max_encoded_bytes = ApproximateEncodedBytes(); |
+ const size_t max_encoded_bytes = MaxEncodedBytes(); |
EncodedInfo info; |
info.encoded_bytes = |
encoded->AppendData( |
@@ -219,16 +229,6 @@ |
size_t AudioEncoderOpus::SamplesPer10msFrame() const { |
return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels; |
-} |
- |
-size_t AudioEncoderOpus::ApproximateEncodedBytes() const { |
- // Calculate the number of bytes we expect the encoder to produce, |
- // then multiply by two to give a wide margin for error. |
- const size_t bytes_per_millisecond = |
- static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); |
- const size_t approx_encoded_bytes = |
- Num10msFramesPerPacket() * 10 * bytes_per_millisecond; |
- return 2 * approx_encoded_bytes; |
} |
// If the given config is OK, recreate the Opus encoder instance with those |