Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index d7203b9da3ea8b5adb9f1998d285cd0320fbce4d..85caf07c4ae8fc631a3fd1a94228c355719b47b1 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -94,10 +94,7 @@ int AudioEncoderG722::GetTargetBitrate() const {
 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
-  RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
-
+    rtc::Buffer* encoded) {
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
@@ -117,27 +114,35 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   num_10ms_frames_buffered_ = 0;
   const size_t samples_per_channel = SamplesPerChannel();
   for (size_t i = 0; i < num_channels_; ++i) {
-    const size_t encoded = WebRtcG722_Encode(
+    const size_t bytes_encoded = WebRtcG722_Encode(

[ossu, 2016/02/24 13:24:33]

This used to shadow the input parameter encoded, which didn't seem right to me.

[kwiberg-webrtc, 2016/02/25 00:29:04]

Acknowledged.

         encoders_[i].encoder, encoders_[i].speech_buffer.get(),
         samples_per_channel, encoders_[i].encoded_buffer.data());
-    RTC_CHECK_EQ(encoded, samples_per_channel / 2);
+    RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
   }
 
-  // Interleave the encoded bytes of the different channels. Each separate
-  // channel and the interleaved stream encodes two samples per byte, most
-  // significant half first.
-  for (size_t i = 0; i < samples_per_channel / 2; ++i) {
-    for (size_t j = 0; j < num_channels_; ++j) {
-      uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
-      interleave_buffer_.data()[j] = two_samples >> 4;
-      interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
-    }
-    for (size_t j = 0; j < num_channels_; ++j)
-      encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
-                                       interleave_buffer_.data()[2 * j + 1];
-  }
+  const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
+
+  encoded->AppendData(bytes_to_encode, [&] (rtc::ArrayView<uint8_t> encoded) {
+      // Interleave the encoded bytes of the different channels. Each separate
+      // channel and the interleaved stream encodes two samples per byte, most
+      // significant half first.
+      for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+        for (size_t j = 0; j < num_channels_; ++j) {
+          uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+          interleave_buffer_.data()[j] = two_samples >> 4;
+          interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+        }
+        for (size_t j = 0; j < num_channels_; ++j)
+          encoded[i * num_channels_ + j] =
+              interleave_buffer_.data()[2 * j] << 4 |
+              interleave_buffer_.data()[2 * j + 1];
+      }
+
+      return bytes_to_encode;
+    });
+
   EncodedInfo info;
-  info.encoded_bytes = samples_per_channel / 2 * num_channels_;
+  info.encoded_bytes = bytes_to_encode;

[kwiberg-webrtc, 2016/02/25 00:29:04]

Have the AppendData call here and set .encoded_bytes to its return value?

[ossu, 2016/02/25 10:39:51]

Yeah.

   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   return info;