g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]>

It's a win for red, and a toss-up for g722 since it never resizes its
buffer.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45219005

Cr-Commit-Position: refs/heads/master@{#9067}

4 files changed, 17 insertions, 21 deletions

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 68e8e084a1..a9fa25f0ef 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -38,14 +38,14 @@ AudioEncoderG722::AudioEncoderG722(const Config& config)
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
-      interleave_buffer_(new uint8_t[2 * num_channels_]) {
+      interleave_buffer_(2 * num_channels_) {
   CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   const int samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
   for (int i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
-    encoders_[i].encoded_buffer.reset(new uint8_t[samples_per_channel / 2]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
 }
 
@@ -105,7 +105,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   for (int i = 0; i < num_channels_; ++i) {
     const int encoded = WebRtcG722_Encode(
         encoders_[i].encoder, encoders_[i].speech_buffer.get(),
-        samples_per_channel, encoders_[i].encoded_buffer.get());
+        samples_per_channel, encoders_[i].encoded_buffer.data<uint8_t>());
     CHECK_GE(encoded, 0);
     CHECK_EQ(encoded, samples_per_channel / 2);
   }
@@ -115,13 +115,13 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // significant half first.
   for (int i = 0; i < samples_per_channel / 2; ++i) {
     for (int j = 0; j < num_channels_; ++j) {
-      uint8_t two_samples = encoders_[j].encoded_buffer[i];
-      interleave_buffer_[j] = two_samples >> 4;
-      interleave_buffer_[num_channels_ + j] = two_samples & 0xf;
+      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 (int j = 0; j < num_channels_; ++j)
-      encoded[i * num_channels_ + j] =
-          interleave_buffer_[2 * j] << 4 | interleave_buffer_[2 * j + 1];
+      encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
+                                       interleave_buffer_.data()[2 * j + 1];
   }
   EncodedInfo info;
   info.encoded_bytes = samples_per_channel / 2 * num_channels_;
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
index b1be6b952e..ba6e5dde67 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
@@ -48,7 +49,7 @@ class AudioEncoderG722 : public AudioEncoder {
   struct EncoderState {
     G722EncInst* encoder;
     rtc::scoped_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
-    rtc::scoped_ptr<uint8_t[]> encoded_buffer;  // Already encoded.
+    rtc::Buffer encoded_buffer;                 // Already encoded.
     EncoderState();
     ~EncoderState();
   };
@@ -61,7 +62,7 @@ class AudioEncoderG722 : public AudioEncoder {
   int num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const rtc::scoped_ptr<EncoderState[]> encoders_;
-  const rtc::scoped_ptr<uint8_t[]> interleave_buffer_;
+  rtc::Buffer interleave_buffer_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 86f1158d9d..cc657a6838 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -18,8 +18,7 @@ namespace webrtc {
 
 AudioEncoderCopyRed::AudioEncoderCopyRed(const Config& config)
     : speech_encoder_(config.speech_encoder),
-      red_payload_type_(config.payload_type),
-      secondary_allocated_(0) {
+      red_payload_type_(config.payload_type) {
   CHECK(speech_encoder_) << "Speech encoder not provided.";
 }
 
@@ -79,18 +78,14 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal(
     info.redundant.push_back(info);
     DCHECK_EQ(info.redundant.size(), 1u);
     if (secondary_info_.encoded_bytes > 0) {
-      memcpy(&encoded[info.encoded_bytes], secondary_encoded_.get(),
+      memcpy(&encoded[info.encoded_bytes], secondary_encoded_.data(),
              secondary_info_.encoded_bytes);
       info.redundant.push_back(secondary_info_);
       DCHECK_EQ(info.redundant.size(), 2u);
     }
     // Save primary to secondary.
-    if (secondary_allocated_ < info.encoded_bytes) {
-      secondary_encoded_.reset(new uint8_t[info.encoded_bytes]);
-      secondary_allocated_ = info.encoded_bytes;
-    }
-    CHECK(secondary_encoded_);
-    memcpy(secondary_encoded_.get(), encoded, info.encoded_bytes);
+    secondary_encoded_.SetSize(info.encoded_bytes);
+    memcpy(secondary_encoded_.data(), encoded, info.encoded_bytes);
     secondary_info_ = info;
     DCHECK_EQ(info.speech, info.redundant[0].speech);
   }
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index fd92d52457..99e5ee2a3e 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
@@ -53,8 +54,7 @@ class AudioEncoderCopyRed : public AudioEncoder {
  private:
   AudioEncoder* speech_encoder_;
   int red_payload_type_;
-  rtc::scoped_ptr<uint8_t[]> secondary_encoded_;
-  size_t secondary_allocated_;
+  rtc::Buffer secondary_encoded_;
   EncodedInfoLeaf secondary_info_;
 };