Pass audio to AudioEncoder::Encode() in an ArrayView

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 6d763005acc4dd3b7b72d45a320ec4b29ae3b049..388b0ff61c6d2b9d02ccebccd7779a329a52bb69 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -21,13 +21,13 @@ int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
-AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
-                                               const int16_t* audio,
-                                               size_t num_samples_per_channel,
-                                               size_t max_encoded_bytes,
-                                               uint8_t* encoded) {
-  RTC_CHECK_EQ(num_samples_per_channel,
-               static_cast<size_t>(SampleRateHz() / 100));
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
   EncodedInfo info =
       EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
   RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);