Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
 
 #include <cstring>
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 8000;
 
+AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderIlbc::Config config;
+  config.frame_size_ms = codec_inst.pacsize / 8;
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
+
 }  // namespace
 
 // static
 const size_t AudioEncoderIlbc::kMaxSamplesPerPacket;
 
 bool AudioEncoderIlbc::Config::IsOk() const {
   return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
           frame_size_ms == 60) &&
       static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
           kMaxSamplesPerPacket;
 }
 
 AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
-    : payload_type_(config.payload_type),
-      num_10ms_frames_per_packet_(
-          static_cast<size_t>(config.frame_size_ms / 10)),
-      num_10ms_frames_buffered_(0) {
-  CHECK(config.IsOk());
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
-  const int encoder_frame_size_ms = config.frame_size_ms > 30
-                                        ? config.frame_size_ms / 2
-                                        : config.frame_size_ms;
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+    : num_10ms_frames_per_packet_(config.frame_size_ms / 10),

[hlundin-webrtc, 2015/09/07 20:00:01]

Why is cast gone?

[kwiberg-webrtc, 2015/09/08 10:47:45]

It got lost in the rebase. Fixed now.

+      config_(config),
+      encoder_(nullptr) {
+  Reset();
 }
 
+AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst)
+    : AudioEncoderIlbc(CreateConfig(codec_inst)) {}
+
 AudioEncoderIlbc::~AudioEncoderIlbc() {
   CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
+size_t AudioEncoderIlbc::MaxEncodedBytes() const {
+  return RequiredOutputSizeBytes();
+}
+
 int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 int AudioEncoderIlbc::NumChannels() const {
   return 1;
 }
 
-size_t AudioEncoderIlbc::MaxEncodedBytes() const {
-  return RequiredOutputSizeBytes();
-}
-
 size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderIlbc::GetTargetBitrate() const {
   switch (num_10ms_frames_per_packet_) {
@@ skipping 36 matching lines @@
   const int output_len = WebRtcIlbcfix_Encode(
       encoder_,
       input_buffer_,
       kSampleRateHz / 100 * num_10ms_frames_per_packet_,
       encoded);
   CHECK_GE(output_len, 0);
   EncodedInfo info;
   info.encoded_bytes = static_cast<size_t>(output_len);
   DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
   info.encoded_timestamp = first_timestamp_in_buffer_;
-  info.payload_type = payload_type_;
+  info.payload_type = config_.payload_type;
   return info;
 }
 
+void AudioEncoderIlbc::Reset() {
+  if (encoder_)
+    CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  CHECK(config_.IsOk());
+  CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms = config_.frame_size_ms > 30
+                                        ? config_.frame_size_ms / 2
+                                        : config_.frame_size_ms;
+  CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
-namespace {
-AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderIlbc::Config config;
-  config.frame_size_ms = codec_inst.pacsize / 8;
-  config.payload_type = codec_inst.pltype;
-  return config;
-}
-}  // namespace
-
-AudioEncoderMutableIlbc::AudioEncoderMutableIlbc(const CodecInst& codec_inst)
-    : AudioEncoderMutableImpl<AudioEncoderIlbc>(CreateConfig(codec_inst)) {
-}
-
 }  // namespace webrtc