Move AudioDecoder and related stuff to the api/ directory

BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}

webrtc/api/BUILD.gn

@@ -29,8 +29,8 @@ rtc_source_set("call_api") {
     ":transport_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
-    "../modules/audio_coding:audio_decoder_factory_interface",
     "../modules/audio_coding:audio_encoder_interface",
+    "audio_codecs:audio_codecs_api",
   ]
 }
 

webrtc/api/DEPS

@@ -12,9 +12,11 @@ specific_include_rules = {
     "+webrtc/voice_engine",
   ],
 
-  # TODO(kwiberg): Remove this exception when audio_decoder_factory.h
+  # We allow .cc files in webrtc/api/ to #include a bunch of stuff
-  # has moved to api/.
+  # that's off-limits for the .h files. That's because .h files leak
-  "peerconnectioninterface\.h": [
+  # their #includes to whoever's #including them, but .cc files do not
-    "+webrtc/modules/audio_coding/codecs/audio_decoder_factory.h",
+  # since no one #includes them.
+  ".*\.cc": [
+    "+webrtc/modules/audio_coding",
   ],
 }

webrtc/api/audio_codecs/BUILD.gn

@@ -0,0 +1,39 @@
+# Copyright (c) 2017 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.
+
+import("../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("audio_codecs_api") {
+  sources = [
+    "audio_decoder.cc",
+    "audio_decoder.h",
+    "audio_decoder_factory.h",
+    "audio_format.cc",
+    "audio_format.h",
+  ]
+  deps = [
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
+  ]
+}
+
+rtc_static_library("builtin_audio_decoder_factory") {
+  sources = [
+    "builtin_audio_decoder_factory.cc",
+    "builtin_audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "../../base:rtc_base_approved",
+    "../../modules/audio_coding:builtin_audio_decoder_factory_internal",
+  ]
+}

webrtc/api/audio_codecs/audio_decoder.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 
 #include <assert.h>
 #include <memory>
@@ -18,10 +18,39 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/sanitizer.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
+namespace {
+
+class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
+      : decoder_(decoder), payload_(std::move(payload)) {}
+
+  size_t Duration() const override {
+    const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    return ret < 0 ? 0 : static_cast<size_t>(ret);
+  }
+
+  rtc::Optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    auto speech_type = AudioDecoder::kSpeech;
+    const int ret = decoder_->Decode(
+        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    return ret < 0 ? rtc::Optional<DecodeResult>()
+                   : rtc::Optional<DecodeResult>(
+                         {static_cast<size_t>(ret), speech_type});
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace
+
 AudioDecoder::ParseResult::ParseResult() = default;
 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
@@ -41,14 +70,17 @@ std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
     uint32_t timestamp) {
   std::vector<ParseResult> results;
   std::unique_ptr<EncodedAudioFrame> frame(
-      new LegacyEncodedAudioFrame(this, std::move(payload)));
+      new OldStyleEncodedFrame(this, std::move(payload)));
   results.emplace_back(timestamp, 0, std::move(frame));
   return results;
 }
 
-int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
+int AudioDecoder::Decode(const uint8_t* encoded,
-                         int sample_rate_hz, size_t max_decoded_bytes,
+                         size_t encoded_len,
-                         int16_t* decoded, SpeechType* speech_type) {
+                         int sample_rate_hz,
+                         size_t max_decoded_bytes,
+                         int16_t* decoded,
+                         SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
   rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
   int duration = PacketDuration(encoded, encoded_len);
@@ -60,9 +92,12 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
                         speech_type);
 }
 
-int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
+int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
-                                  int sample_rate_hz, size_t max_decoded_bytes,
+                                  size_t encoded_len,
-                                  int16_t* decoded, SpeechType* speech_type) {
+                                  int sample_rate_hz,
+                                  size_t max_decoded_bytes,
+                                  int16_t* decoded,
+                                  SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
   rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
   int duration = PacketDurationRedundant(encoded, encoded_len);
@@ -76,13 +111,16 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
 
 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
                                           size_t encoded_len,
-                                          int sample_rate_hz, int16_t* decoded,
+                                          int sample_rate_hz,
+                                          int16_t* decoded,
                                           SpeechType* speech_type) {
   return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
                         speech_type);
 }
 
-bool AudioDecoder::HasDecodePlc() const { return false; }
+bool AudioDecoder::HasDecodePlc() const {
+  return false;
+}
 
 size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
   return 0;
@@ -96,7 +134,9 @@ int AudioDecoder::IncomingPacket(const uint8_t* payload,
   return 0;
 }
 
-int AudioDecoder::ErrorCode() { return 0; }
+int AudioDecoder::ErrorCode() {
+  return 0;
+}
 
 int AudioDecoder::PacketDuration(const uint8_t* encoded,
                                  size_t encoded_len) const {

webrtc/api/audio_codecs/audio_decoder.h

@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDecoder {
+ public:
+  enum SpeechType {
+    kSpeech = 1,
+    kComfortNoise = 2,
+  };
+
+  // Used by PacketDuration below. Save the value -1 for errors.
+  enum { kNotImplemented = -2 };
+
+  AudioDecoder() = default;
+  virtual ~AudioDecoder() = default;
+
+  class EncodedAudioFrame {
+   public:
+    struct DecodeResult {
+      size_t num_decoded_samples;
+      SpeechType speech_type;
+    };
+
+    virtual ~EncodedAudioFrame() = default;
+
+    // Returns the duration in samples-per-channel of this audio frame.
+    // If no duration can be ascertained, returns zero.
+    virtual size_t Duration() const = 0;
+
+    // Decodes this frame of audio and writes the result in |decoded|.
+    // |decoded| must be large enough to store as many samples as indicated by a
+    // call to Duration() . On success, returns an rtc::Optional containing the
+    // total number of samples across all channels, as well as whether the
+    // decoder produced comfort noise or speech. On failure, returns an empty
+    // rtc::Optional. Decode may be called at most once per frame object.
+    virtual rtc::Optional<DecodeResult> Decode(
+        rtc::ArrayView<int16_t> decoded) const = 0;
+  };
+
+  struct ParseResult {
+    ParseResult();
+    ParseResult(uint32_t timestamp,
+                int priority,
+                std::unique_ptr<EncodedAudioFrame> frame);
+    ParseResult(ParseResult&& b);
+    ~ParseResult();
+
+    ParseResult& operator=(ParseResult&& b);
+
+    // The timestamp of the frame is in samples per channel.
+    uint32_t timestamp;
+    // The relative priority of the frame compared to other frames of the same
+    // payload and the same timeframe. A higher value means a lower priority.
+    // The highest priority is zero - negative values are not allowed.
+    int priority;
+    std::unique_ptr<EncodedAudioFrame> frame;
+  };
+
+  // Let the decoder parse this payload and prepare zero or more decodable
+  // frames. Each frame must be between 10 ms and 120 ms long. The caller must
+  // ensure that the AudioDecoder object outlives any frame objects returned by
+  // this call. The decoder is free to swap or move the data from the |payload|
+  // buffer. |timestamp| is the input timestamp, in samples, corresponding to
+  // the start of the payload.
+  virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                                uint32_t timestamp);
+
+  // Decodes |encode_len| bytes from |encoded| and writes the result in
+  // |decoded|. The maximum bytes allowed to be written into |decoded| is
+  // |max_decoded_bytes|. Returns the total number of samples across all
+  // channels. If the decoder produced comfort noise, |speech_type|
+  // is set to kComfortNoise, otherwise it is kSpeech. The desired output
+  // sample rate is provided in |sample_rate_hz|, which must be valid for the
+  // codec at hand.
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
+
+  // Same as Decode(), but interfaces to the decoders redundant decode function.
+  // The default implementation simply calls the regular Decode() method.
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
+
+  // Indicates if the decoder implements the DecodePlc method.
+  virtual bool HasDecodePlc() const;
+
+  // Calls the packet-loss concealment of the decoder to update the state after
+  // one or several lost packets. The caller has to make sure that the
+  // memory allocated in |decoded| should accommodate |num_frames| frames.
+  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
+
+  // Resets the decoder state (empty buffers etc.).
+  virtual void Reset() = 0;
+
+  // Notifies the decoder of an incoming packet to NetEQ.
+  virtual int IncomingPacket(const uint8_t* payload,
+                             size_t payload_len,
+                             uint16_t rtp_sequence_number,
+                             uint32_t rtp_timestamp,
+                             uint32_t arrival_timestamp);
+
+  // Returns the last error code from the decoder.
+  virtual int ErrorCode();
+
+  // Returns the duration in samples-per-channel of the payload in |encoded|
+  // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
+  // estimate is available, or -1 in case of an error.
+  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the duration in samples-per-channel of the redandant payload in
+  // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
+  // duration estimate is available, or -1 in case of an error.
+  virtual int PacketDurationRedundant(const uint8_t* encoded,
+                                      size_t encoded_len) const;
+
+  // Detects whether a packet has forward error correction. The packet is
+  // comprised of the samples in |encoded| which is |encoded_len| bytes long.
+  // Returns true if the packet has FEC and false otherwise.
+  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the actual sample rate of the decoder's output. This value may not
+  // change during the lifetime of the decoder.
+  virtual int SampleRateHz() const = 0;
+
+  // The number of channels in the decoder's output. This value may not change
+  // during the lifetime of the decoder.
+  virtual size_t Channels() const = 0;
+
+ protected:
+  static SpeechType ConvertSpeechType(int16_t type);
+
+  virtual int DecodeInternal(const uint8_t* encoded,
+                             size_t encoded_len,
+                             int sample_rate_hz,
+                             int16_t* decoded,
+                             SpeechType* speech_type) = 0;
+
+  virtual int DecodeRedundantInternal(const uint8_t* encoded,
+                                      size_t encoded_len,
+                                      int sample_rate_hz,
+                                      int16_t* decoded,
+                                      SpeechType* speech_type);
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_

webrtc/api/audio_codecs/audio_decoder_factory.h

@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
+#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
 
 #include <memory>
 #include <vector>
 
-#include "webrtc/base/atomicops.h"
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
 namespace webrtc {
 
@@ -35,4 +34,4 @@ class AudioDecoderFactory : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
+#endif  // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_

webrtc/api/audio_codecs/audio_format.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
 
 #include "webrtc/common_types.h"
 
@@ -77,8 +77,7 @@ std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
   return os;
 }
 
-AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format)
+AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {}
-    : format(format) {}
 
 AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
     : format(std::move(format)) {}

webrtc/api/audio_codecs/audio_format.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
+#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
 
 #include <map>
 #include <ostream>
@@ -70,12 +70,12 @@ struct AudioCodecSpec {
   ~AudioCodecSpec() = default;
 
   SdpAudioFormat format;
-  bool allow_comfort_noise = true;    // This codec can be used with an external
+  bool allow_comfort_noise = true;  // This codec can be used with an external
-                                      // comfort noise generator.
+                                    // comfort noise generator.
   bool supports_network_adaption = false;  // This codec can adapt to varying
                                            // network conditions.
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
+#endif  // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_

webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc

@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2017 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/api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
+  return CreateBuiltinAudioDecoderFactoryInternal();
+}
+
+}  // namespace webrtc

webrtc/api/audio_codecs/builtin_audio_decoder_factory.h

@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio decoders.
+// NOTE: This function is still under development and may change without notice.
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

webrtc/api/peerconnectioninterface.h

@@ -73,13 +73,14 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/datachannelinterface.h"
 #include "webrtc/api/dtmfsenderinterface.h"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
+#include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/api/umametrics.h"
 #include "webrtc/base/fileutils.h"
@@ -89,7 +90,6 @@
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/media/base/mediachannel.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {

webrtc/call/audio_receive_stream.h

@@ -16,10 +16,10 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/typedefs.h"

webrtc/logging/BUILD.gn

@@ -139,7 +139,7 @@ if (rtc_enable_protobuf) {
         "../base:rtc_base_approved",
 
         # TODO(kwiberg): Remove this dependency.
-        "../modules/audio_coding:audio_format",
+        "../api/audio_codecs:audio_codecs_api",
         "../modules/rtp_rtcp:rtp_rtcp",
         "//third_party/gflags",
       ]

webrtc/media/base/mediaengine.h

@@ -18,6 +18,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/sigslotrepeater.h"
@@ -25,7 +26,6 @@
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videocommon.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 
 #if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
 #define DISABLE_MEDIA_ENGINE_FACTORY

webrtc/media/engine/payload_type_mapper.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/media/engine/payload_type_mapper.h"
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/common_types.h"
 #include "webrtc/media/base/mediaconstants.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
 namespace cricket {
 

webrtc/media/engine/payload_type_mapper.h

@@ -14,9 +14,9 @@
 #include <map>
 #include <set>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/media/base/codec.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
 namespace cricket {
 

webrtc/media/engine/webrtcmediaengine.cc

@@ -9,16 +9,17 @@
  */
 
 #include "webrtc/media/engine/webrtcmediaengine.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 
 #include <algorithm>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/media/engine/webrtcvoiceengine.h"
+
 #ifdef HAVE_WEBRTC_VIDEO
 #include "webrtc/media/engine/webrtcvideoengine2.h"
 #else
 #include "webrtc/media/engine/nullwebrtcvideoengine.h"
 #endif
-#include "webrtc/media/engine/webrtcvoiceengine.h"
 
 namespace cricket {
 

webrtc/media/engine/webrtcmediaengine_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 
 using webrtc::RtpExtension;

webrtc/media/engine/webrtcvoiceengine_unittest.cc

@@ -10,12 +10,11 @@
 
 #include <memory>
 
-#include "webrtc/pc/channel.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/call/call.h"
-#include "webrtc/test/field_trial.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/fakenetworkinterface.h"
@@ -24,10 +23,11 @@
 #include "webrtc/media/engine/fakewebrtccall.h"
 #include "webrtc/media/engine/fakewebrtcvoiceengine.h"
 #include "webrtc/media/engine/webrtcvoiceengine.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
+#include "webrtc/pc/channel.h"
+#include "webrtc/test/field_trial.h"
 
 using testing::Return;
 using testing::StrictMock;

webrtc/modules/audio_coding/BUILD.gn

@@ -39,52 +39,27 @@ audio_coding_deps = audio_codec_deps + [
                       "../../system_wrappers",
                     ]
 
-rtc_static_library("audio_format") {
-  sources = [
-    "codecs/audio_format.cc",
-    "codecs/audio_format.h",
-  ]
-  deps = [
-    "../..:webrtc_common",
-  ]
-}
-
 rtc_static_library("audio_format_conversion") {
   sources = [
     "codecs/audio_format_conversion.cc",
     "codecs/audio_format_conversion.h",
   ]
   deps = [
-    ":audio_format",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
   ]
 }
 
-rtc_source_set("audio_decoder_factory_interface") {
+rtc_static_library("builtin_audio_decoder_factory_internal") {
   sources = [
-    "codecs/audio_decoder_factory.h",
+    "codecs/builtin_audio_decoder_factory_internal.cc",
-  ]
+    "codecs/builtin_audio_decoder_factory_internal.h",
-  deps = [
-    ":audio_decoder_interface",
-    ":audio_format",
-
-    # TODO(charujain): Clean this dependency when downstream projects are
-    # updated to properly depend on audio_format_conversion target.
-    ":audio_format_conversion",
-    "../../base:rtc_base_approved",
-  ]
-}
-
-rtc_static_library("builtin_audio_decoder_factory") {
-  sources = [
-    "codecs/builtin_audio_decoder_factory.cc",
-    "codecs/builtin_audio_decoder_factory.h",
   ]
   deps = [
            "../..:webrtc_common",
            "../../base:rtc_base_approved",
-           ":audio_decoder_factory_interface",
+           "../../api/audio_codecs:audio_codecs_api",
          ] + audio_codec_deps
   defines = audio_codec_defines
 }
@@ -101,7 +76,7 @@ rtc_static_library("rent_a_codec") {
     "acm2/rent_a_codec.h",
   ]
   deps = [
-           ":audio_decoder_interface",
+           "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
            "../../base:rtc_base_approved",
          ] + audio_codec_deps
@@ -149,9 +124,8 @@ rtc_static_library("audio_coding") {
   }
 
   deps = audio_coding_deps + [
-           ":audio_decoder_interface",
+           "../../api/audio_codecs:audio_codecs_api",
-           ":audio_decoder_factory_interface",
+           "../../api/audio_codecs:builtin_audio_decoder_factory",
-           ":builtin_audio_decoder_factory",
            ":neteq",
            ":rent_a_codec",
            "../../base:rtc_base_approved",
@@ -160,15 +134,13 @@ rtc_static_library("audio_coding") {
   defines = audio_coding_defines
 }
 
-rtc_static_library("audio_decoder_interface") {
+rtc_static_library("legacy_encoded_audio_frame") {
   sources = [
-    "codecs/audio_decoder.cc",
-    "codecs/audio_decoder.h",
     "codecs/legacy_encoded_audio_frame.cc",
     "codecs/legacy_encoded_audio_frame.h",
   ]
   deps = [
-    "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
   ]
 }
@@ -246,9 +218,10 @@ rtc_static_library("g711") {
   public_configs = [ ":g711_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":legacy_encoded_audio_frame",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
   ]
   public_deps = [
@@ -287,9 +260,10 @@ rtc_static_library("g722") {
   public_configs = [ ":g722_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":legacy_encoded_audio_frame",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
   ]
   public_deps = [
@@ -329,9 +303,10 @@ rtc_static_library("ilbc") {
   public_configs = [ ":ilbc_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":legacy_encoded_audio_frame",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
     "../../common_audio",
   ]
@@ -487,9 +462,9 @@ rtc_source_set("ilbc_c") {
   public_configs = [ ":ilbc_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
     "../../common_audio",
   ]
@@ -525,9 +500,9 @@ rtc_static_library("isac") {
   ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":isac_common",
+    "../../api/audio_codecs:audio_codecs_api",
   ]
   public_deps = [
     ":isac_c",
@@ -619,9 +594,9 @@ rtc_static_library("isac_fix") {
   public_configs = [ ":isac_fix_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":isac_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../system_wrappers",
   ]
@@ -695,10 +670,10 @@ rtc_source_set("isac_fix_c") {
   public_configs = [ ":isac_fix_config" ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":isac_common",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
     "../../common_audio",
     "../../system_wrappers",
@@ -799,10 +774,11 @@ rtc_static_library("pcm16b") {
   ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":g711",
+    ":legacy_encoded_audio_frame",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
   ]
   public_deps = [
@@ -837,10 +813,10 @@ rtc_static_library("webrtc_opus") {
   ]
 
   deps = [
-    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":audio_network_adaptor",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:rtc_base_approved",
     "../../base:rtc_numerics",
     "../../common_audio",
@@ -1031,16 +1007,13 @@ rtc_static_library("neteq") {
   ]
 
   deps = [
-    ":audio_decoder_factory_interface",
-    ":audio_decoder_interface",
-    ":audio_format",
-    ":builtin_audio_decoder_factory",
     ":cng",
     ":g711",
     ":isac_fix",
     ":pcm16b",
     ":rent_a_codec",
     "../..:webrtc_common",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:gtest_prod",
     "../../base:rtc_base_approved",
     "../../common_audio",
@@ -1095,9 +1068,9 @@ rtc_source_set("neteq_test_minimal") {
 
   deps = [
     ":audio_encoder_interface",
-    ":builtin_audio_decoder_factory",
     ":neteq",
     "../..:webrtc_common",
+    "../../api/audio_codecs:builtin_audio_decoder_factory",
     "../../base:rtc_base_approved",
   ]
 }
@@ -1161,9 +1134,9 @@ if (rtc_include_tests) {
     deps = [
       ":audio_coding",
       ":audio_format_conversion",
-      ":builtin_audio_decoder_factory",
       ":pcm16b_c",
       "../..:webrtc_common",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
       "../../test:fileutils",
@@ -1215,8 +1188,8 @@ if (rtc_include_tests) {
     deps = audio_coding_deps + [
              ":audio_coding",
              ":audio_format_conversion",
-             ":audio_decoder_factory_interface",
+             "../../api/audio_codecs:audio_codecs_api",
-             ":builtin_audio_decoder_factory",
+             "../../api/audio_codecs:builtin_audio_decoder_factory",
              ":neteq_unittest_tools",
              "../../base:rtc_base_approved",
              "../../test:test_support",
@@ -1235,7 +1208,7 @@ if (rtc_include_tests) {
 
     deps = audio_coding_deps + [
              ":audio_coding",
-             ":audio_decoder_interface",
+             "../../api/audio_codecs:audio_codecs_api",
              ":audio_encoder_interface",
              ":neteq_unittest_tools",
              "../../base:rtc_base_approved",
@@ -1324,12 +1297,12 @@ if (rtc_include_tests) {
 
     deps += audio_coding_deps
     deps += [
-      ":audio_decoder_interface",
       ":ilbc",
       ":isac",
       ":isac_fix",
       ":neteq",
       ":neteq_unittest_tools",
+      "../../api/audio_codecs:audio_codecs_api",
       "../../common_audio",
       "../../test:test_main",
       "//testing/gtest",
@@ -1460,12 +1433,12 @@ if (rtc_include_tests) {
     }
 
     deps = [
-      ":audio_decoder_interface",
-      ":builtin_audio_decoder_factory",
       ":neteq",
       ":neteq_unittest_tools",
       ":pcm16b",
       "../..:webrtc_common",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../base:rtc_base_approved",
       "../../system_wrappers",
       "../../test:test_support",
@@ -1486,10 +1459,10 @@ if (rtc_include_tests) {
     }
 
     deps = [
-      ":builtin_audio_decoder_factory",
       ":neteq",
       ":neteq_unittest_tools",
       "../..:webrtc_common",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../base:rtc_base_approved",
       "../../test:test_support",
       "//testing/gtest",
@@ -1540,10 +1513,10 @@ if (rtc_include_tests) {
     }
 
     deps = [
-      ":audio_decoder_interface",
       ":audio_encoder_interface",
       ":pcm16b",
       "../..:webrtc_common",
+      "../../api/audio_codecs:audio_codecs_api",
       "../../base:rtc_base_approved",
       "../../common_audio",
       "../../test:rtp_test_utils",
@@ -1999,7 +1972,7 @@ if (rtc_include_tests) {
       "audio_network_adaptor/frame_length_controller_unittest.cc",
       "audio_network_adaptor/mock/mock_controller.h",
       "audio_network_adaptor/mock/mock_controller_manager.h",
-      "codecs/audio_decoder_factory_unittest.cc",
+      "codecs/builtin_audio_decoder_factory_unittest.cc",
       "codecs/cng/audio_encoder_cng_unittest.cc",
       "codecs/cng/cng_unittest.cc",
       "codecs/ilbc/ilbc_unittest.cc",
@@ -2063,17 +2036,16 @@ if (rtc_include_tests) {
       ":acm_receive_test",
       ":acm_send_test",
       ":audio_coding",
-      ":audio_decoder_interface",
       ":audio_encoder_interface",
       ":audio_format_conversion",
       ":audio_network_adaptor",
-      ":builtin_audio_decoder_factory",
       ":cng",
       ":g711",
       ":ilbc",
       ":isac",
       ":isac_c",
       ":isac_fix",
+      ":legacy_encoded_audio_frame",
       ":neteq",
       ":neteq_test_support",
       ":neteq_unittest_tools",
@@ -2082,6 +2054,8 @@ if (rtc_include_tests) {
       ":rent_a_codec",
       ":webrtc_opus",
       "../..:webrtc_common",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../base:rtc_base",
       "../../base:rtc_base_approved",
       "../../base:rtc_base_tests_utils",
@@ -2115,3 +2089,27 @@ if (rtc_include_tests) {
     }
   }
 }
+
+# For backwards compatibility only! Use
+# webrtc/api/audio_codecs:audio_codecs_api instead.
+# TODO(kwiberg): Remove this.
+rtc_source_set("audio_decoder_interface") {
+  sources = [
+    "codecs/audio_decoder.h",
+  ]
+  deps = [
+    "../../api/audio_codecs:audio_codecs_api",
+  ]
+}
+
+# For backwards compatibility only! Use
+# webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
+# TODO(kwiberg): Remove this.
+rtc_source_set("builtin_audio_decoder_factory") {
+  sources = [
+    "codecs/builtin_audio_decoder_factory.h",
+  ]
+  deps = [
+    "../../api/audio_codecs:builtin_audio_decoder_factory",
+  ]
+}

webrtc/modules/audio_coding/acm2/acm_receive_test.cc

@@ -15,8 +15,8 @@
 
 #include <memory>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"

webrtc/modules/audio_coding/acm2/acm_receive_test.h

@@ -14,9 +14,9 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/acm2/acm_receiver.cc

@@ -15,13 +15,13 @@
 #include <algorithm>  // sort
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"

webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc

@@ -13,9 +13,9 @@
 #include <algorithm>  // std::min
 #include <memory>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/system_wrappers/include/clock.h"

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

@@ -10,13 +10,13 @@
 
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/trace.h"
 

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

@@ -13,13 +13,13 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receive_test.h"
 #include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"

webrtc/modules/audio_coding/acm2/rent_a_codec.h

@@ -15,12 +15,12 @@
 #include <map>
 #include <memory>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/typedefs.h"

webrtc/modules/audio_coding/codecs/audio_decoder.h

@@ -8,172 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_decoder.h instead!
+// TODO(kwiberg): Remove it.
+
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 
-#include <memory>
+#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include <vector>
 
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/optional.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// This is the interface class for decoders in NetEQ. Each codec type will have
-// and implementation of this class.
-class AudioDecoder {
- public:
-  enum SpeechType {
-    kSpeech = 1,
-    kComfortNoise = 2
-  };
-
-  // Used by PacketDuration below. Save the value -1 for errors.
-  enum { kNotImplemented = -2 };
-
-  AudioDecoder() = default;
-  virtual ~AudioDecoder() = default;
-
-  class EncodedAudioFrame {
-   public:
-    struct DecodeResult {
-      size_t num_decoded_samples;
-      SpeechType speech_type;
-    };
-
-    virtual ~EncodedAudioFrame() = default;
-
-    // Returns the duration in samples-per-channel of this audio frame.
-    // If no duration can be ascertained, returns zero.
-    virtual size_t Duration() const = 0;
-
-    // Decodes this frame of audio and writes the result in |decoded|.
-    // |decoded| must be large enough to store as many samples as indicated by a
-    // call to Duration() . On success, returns an rtc::Optional containing the
-    // total number of samples across all channels, as well as whether the
-    // decoder produced comfort noise or speech. On failure, returns an empty
-    // rtc::Optional. Decode may be called at most once per frame object.
-    virtual rtc::Optional<DecodeResult> Decode(
-        rtc::ArrayView<int16_t> decoded) const = 0;
-  };
-
-  struct ParseResult {
-    ParseResult();
-    ParseResult(uint32_t timestamp,
-                int priority,
-                std::unique_ptr<EncodedAudioFrame> frame);
-    ParseResult(ParseResult&& b);
-    ~ParseResult();
-
-    ParseResult& operator=(ParseResult&& b);
-
-    // The timestamp of the frame is in samples per channel.
-    uint32_t timestamp;
-    // The relative priority of the frame compared to other frames of the same
-    // payload and the same timeframe. A higher value means a lower priority.
-    // The highest priority is zero - negative values are not allowed.
-    int priority;
-    std::unique_ptr<EncodedAudioFrame> frame;
-  };
-
-  // Let the decoder parse this payload and prepare zero or more decodable
-  // frames. Each frame must be between 10 ms and 120 ms long. The caller must
-  // ensure that the AudioDecoder object outlives any frame objects returned by
-  // this call. The decoder is free to swap or move the data from the |payload|
-  // buffer. |timestamp| is the input timestamp, in samples, corresponding to
-  // the start of the payload.
-  virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
-                                                uint32_t timestamp);
-
-  // Decodes |encode_len| bytes from |encoded| and writes the result in
-  // |decoded|. The maximum bytes allowed to be written into |decoded| is
-  // |max_decoded_bytes|. Returns the total number of samples across all
-  // channels. If the decoder produced comfort noise, |speech_type|
-  // is set to kComfortNoise, otherwise it is kSpeech. The desired output
-  // sample rate is provided in |sample_rate_hz|, which must be valid for the
-  // codec at hand.
-  int Decode(const uint8_t* encoded,
-             size_t encoded_len,
-             int sample_rate_hz,
-             size_t max_decoded_bytes,
-             int16_t* decoded,
-             SpeechType* speech_type);
-
-  // Same as Decode(), but interfaces to the decoders redundant decode function.
-  // The default implementation simply calls the regular Decode() method.
-  int DecodeRedundant(const uint8_t* encoded,
-                      size_t encoded_len,
-                      int sample_rate_hz,
-                      size_t max_decoded_bytes,
-                      int16_t* decoded,
-                      SpeechType* speech_type);
-
-  // Indicates if the decoder implements the DecodePlc method.
-  virtual bool HasDecodePlc() const;
-
-  // Calls the packet-loss concealment of the decoder to update the state after
-  // one or several lost packets. The caller has to make sure that the
-  // memory allocated in |decoded| should accommodate |num_frames| frames.
-  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
-
-  // Resets the decoder state (empty buffers etc.).
-  virtual void Reset() = 0;
-
-  // Notifies the decoder of an incoming packet to NetEQ.
-  virtual int IncomingPacket(const uint8_t* payload,
-                             size_t payload_len,
-                             uint16_t rtp_sequence_number,
-                             uint32_t rtp_timestamp,
-                             uint32_t arrival_timestamp);
-
-  // Returns the last error code from the decoder.
-  virtual int ErrorCode();
-
-  // Returns the duration in samples-per-channel of the payload in |encoded|
-  // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
-  // estimate is available, or -1 in case of an error.
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-
-  // Returns the duration in samples-per-channel of the redandant payload in
-  // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
-  // duration estimate is available, or -1 in case of an error.
-  virtual int PacketDurationRedundant(const uint8_t* encoded,
-                                      size_t encoded_len) const;
-
-  // Detects whether a packet has forward error correction. The packet is
-  // comprised of the samples in |encoded| which is |encoded_len| bytes long.
-  // Returns true if the packet has FEC and false otherwise.
-  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
-
-  // Returns the actual sample rate of the decoder's output. This value may not
-  // change during the lifetime of the decoder.
-  virtual int SampleRateHz() const = 0;
-
-  // The number of channels in the decoder's output. This value may not change
-  // during the lifetime of the decoder.
-  virtual size_t Channels() const = 0;
-
- protected:
-  static SpeechType ConvertSpeechType(int16_t type);
-
-  virtual int DecodeInternal(const uint8_t* encoded,
-                             size_t encoded_len,
-                             int sample_rate_hz,
-                             int16_t* decoded,
-                             SpeechType* speech_type) = 0;
-
-  virtual int DecodeRedundantInternal(const uint8_t* encoded,
-                                      size_t encoded_len,
-                                      int sample_rate_hz,
-                                      int16_t* decoded,
-                                      SpeechType* speech_type);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
-};
-
-}  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_

webrtc/modules/audio_coding/codecs/audio_format_conversion.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h

@@ -8,20 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
+// TODO(kwiberg): Remove it.
+
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
 
-#include <memory>
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
-
-namespace webrtc {
-
-// Creates a new factory that can create the built-in types of audio decoders.
-// NOTE: This function is still under development and may change without notice.
-rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
-
-}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc

@@ -8,8 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h"
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/checks.h"
@@ -176,34 +177,35 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
   std::vector<AudioCodecSpec> GetSupportedDecoders() override {
     // Although this looks a bit strange, it means specs need only be initalized
     // once, and that that initialization is thread-safe.
-    static std::vector<AudioCodecSpec> specs =
+    static std::vector<AudioCodecSpec> specs = [] {
-        []{
+      std::vector<AudioCodecSpec> specs;
-          std::vector<AudioCodecSpec> specs;
 #ifdef WEBRTC_CODEC_OPUS
-          AudioCodecSpec opus({"opus", 48000, 2, {
+      // clang-format off
-                                 {"minptime", "10"},
+      AudioCodecSpec opus({"opus", 48000, 2, {
-                                 {"useinbandfec", "1"}
+                             {"minptime", "10"},
-                               }});
+                             {"useinbandfec", "1"}
-          opus.allow_comfort_noise = false;
+                           }});
-          opus.supports_network_adaption = true;
+      // clang-format on
-          specs.push_back(opus);
+      opus.allow_comfort_noise = false;
+      opus.supports_network_adaption = true;
+      specs.push_back(opus);
 #endif
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-          specs.push_back(AudioCodecSpec({"isac", 16000, 1}));
+      specs.push_back(AudioCodecSpec({"isac", 16000, 1}));
 #endif
 #if (defined(WEBRTC_CODEC_ISAC))
-          specs.push_back(AudioCodecSpec({"isac", 32000, 1}));
+      specs.push_back(AudioCodecSpec({"isac", 32000, 1}));
 #endif
 #ifdef WEBRTC_CODEC_G722
-          specs.push_back(AudioCodecSpec({"G722", 8000, 1}));
+      specs.push_back(AudioCodecSpec({"G722", 8000, 1}));
 #endif
 #ifdef WEBRTC_CODEC_ILBC
-          specs.push_back(AudioCodecSpec({"iLBC", 8000, 1}));
+      specs.push_back(AudioCodecSpec({"iLBC", 8000, 1}));
 #endif
-          specs.push_back(AudioCodecSpec({"PCMU", 8000, 1}));
+      specs.push_back(AudioCodecSpec({"PCMU", 8000, 1}));
-          specs.push_back(AudioCodecSpec({"PCMA", 8000, 1}));
+      specs.push_back(AudioCodecSpec({"PCMA", 8000, 1}));
-          return specs;
+      return specs;
-        }();
+    }();
     return specs;
   }
 
@@ -239,7 +241,8 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
 
 }  // namespace
 
-rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
+rtc::scoped_refptr<AudioDecoderFactory>
+CreateBuiltinAudioDecoderFactoryInternal() {
   return rtc::scoped_refptr<AudioDecoderFactory>(
       new rtc::RefCountedObject<BuiltinAudioDecoderFactory>);
 }

webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h

@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
+
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDecoderFactory>
+CreateBuiltinAudioDecoderFactoryInternal();
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_

webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h

@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 typedef struct WebRtcG722DecInst G722DecInst;
 

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
 

webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h

@@ -13,10 +13,10 @@
 
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h

@@ -13,8 +13,8 @@
 
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/array_view.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h

@@ -13,9 +13,9 @@
 
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/include/audio_coding_module.h

@@ -15,11 +15,11 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/base/deprecation.h"
 #include "webrtc/base/function_view.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module.h"

webrtc/modules/audio_coding/neteq/audio_decoder_impl.h

@@ -13,13 +13,14 @@
 
 #include <assert.h>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/typedefs.h"
+
 #ifdef WEBRTC_CODEC_G722
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/typedefs.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/neteq/comfort_noise.cc

@@ -12,8 +12,8 @@
 
 #include <assert.h>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"

webrtc/modules/audio_coding/neteq/decoder_database.cc

@@ -12,9 +12,9 @@
 
 #include <utility>  // pair
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
 

webrtc/modules/audio_coding/neteq/decoder_database.h

@@ -15,10 +15,10 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_types.h"  // NULL
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"

webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc

@@ -15,7 +15,7 @@
 
 #include <string>
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/test/gmock.h"

webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h

@@ -11,8 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_
 
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_decoder.h"
-
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h

@@ -11,8 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
 
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_decoder.h"
-
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/gmock.h"

webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc

@@ -12,7 +12,7 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"

webrtc/modules/audio_coding/neteq/neteq_impl.cc

@@ -17,13 +17,13 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/sanitizer.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
@@ -39,11 +39,11 @@
 #include "webrtc/modules/audio_coding/neteq/merge.h"
 #include "webrtc/modules/audio_coding/neteq/nack_tracker.h"
 #include "webrtc/modules/audio_coding/neteq/normal.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
+#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
+#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -10,14 +10,12 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
-
 #include "webrtc/base/safe_conversions.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
@@ -27,6 +25,7 @@
 #include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
+#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"

webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc

@@ -15,7 +15,7 @@
 #include <string>
 #include <list>
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

@@ -21,10 +21,10 @@
 #include <vector>
 
 #include "gflags/gflags.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/sha1digest.h"
 #include "webrtc/base/stringencode.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"

webrtc/modules/audio_coding/neteq/normal.cc

@@ -14,9 +14,9 @@
 
 #include <algorithm>  // min
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"

webrtc/modules/audio_coding/neteq/packet.h

@@ -14,8 +14,8 @@
 #include <list>
 #include <memory>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/typedefs.h"
 

webrtc/modules/audio_coding/neteq/packet_buffer.cc

@@ -16,8 +16,8 @@
 
 #include <algorithm>  // find_if()
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 

webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc

@@ -10,11 +10,10 @@
 
 // Unit tests for PacketBuffer class.
 
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"

webrtc/modules/audio_coding/neteq/post_decode_vad.h

@@ -13,10 +13,10 @@
 
 #include <string>  // size_t
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/common_types.h"  // NULL
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/typedefs.h"

webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc

@@ -17,7 +17,7 @@
 #include <memory>
 #include <utility>  // pair
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"

webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc

@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h

@@ -13,9 +13,9 @@
 
 #include <memory>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/optional.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc

@@ -11,8 +11,8 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h

@@ -14,7 +14,7 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module_common_types.h"
 

webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc

@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc

@@ -10,8 +10,9 @@
 
 #include <math.h>
 #include <stdio.h>
+
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"

webrtc/modules/audio_coding/neteq/tools/neteq_test.cc

@@ -12,7 +12,7 @@
 
 #include <iostream>
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 
 namespace webrtc {
 namespace test {

webrtc/modules/audio_coding/test/TwoWayCommunication.cc

@@ -20,9 +20,9 @@
 #include <Windows.h>
 #endif
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"

webrtc/modules/utility/BUILD.gn

@@ -35,10 +35,6 @@ rtc_static_library("utility") {
     "../../base:rtc_task_queue",
     "../../common_audio",
     "../../system_wrappers",
-    "../audio_coding",
-    "../audio_coding:audio_format_conversion",
-    "../audio_coding:builtin_audio_decoder_factory",
-    "../audio_coding:rent_a_codec",
     "../media_file",
   ]
 }

webrtc/pc/peerconnectionfactory.cc

@@ -12,6 +12,7 @@
 
 #include <utility>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/mediastreamproxy.h"
 #include "webrtc/api/mediastreamtrackproxy.h"
@@ -23,7 +24,6 @@
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvideodecoderfactory.h"
 #include "webrtc/media/engine/webrtcvideoencoderfactory.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/client/basicportallocator.h"

webrtc/pc/peerconnectioninterface_unittest.cc

@@ -13,6 +13,7 @@
 #include <string>
 #include <utility>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
@@ -26,7 +27,6 @@
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/sctp/sctptransportinternal.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/mediasession.h"

webrtc/test/call_test.cc

@@ -12,9 +12,9 @@
 
 #include <algorithm>
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"

webrtc/test/fuzzers/audio_decoder_fuzzer.cc

@@ -12,9 +12,9 @@
 
 #include <limits>
 
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/optional.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {

webrtc/tools/BUILD.gn

@@ -208,7 +208,7 @@ if (rtc_enable_protobuf) {
       "../modules/audio_coding:ana_debug_dump_proto",
 
       # TODO(kwiberg): Remove this dependency.
-      "../modules/audio_coding:audio_format",
+      "../api/audio_codecs:audio_codecs_api",
       "../modules/congestion_controller",
       "../modules/rtp_rtcp",
       "../system_wrappers:system_wrappers_default",

webrtc/voice_engine/BUILD.gn

@@ -15,9 +15,9 @@ rtc_static_library("audio_coder") {
   ]
   deps = [
     "..:webrtc_common",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
     "../modules/audio_coding",
     "../modules/audio_coding:audio_format_conversion",
-    "../modules/audio_coding:builtin_audio_decoder_factory",
     "../modules/audio_coding:rent_a_codec",
   ]
 
@@ -152,13 +152,13 @@ rtc_static_library("voice_engine") {
     "../api:audio_mixer_api",
     "../api:call_api",
     "../api:transport_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
     "../audio/utility:audio_frame_operations",
     "../base:rtc_base_approved",
     "../common_audio",
     "../logging:rtc_event_log_api",
-    "../modules/audio_coding:audio_decoder_factory_interface",
     "../modules/audio_coding:audio_format_conversion",
-    "../modules/audio_coding:builtin_audio_decoder_factory",
     "../modules/audio_coding:rent_a_codec",
     "../modules/audio_conference_mixer",
     "../modules/audio_device",

webrtc/voice_engine/coder.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/voice_engine/coder.h"
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {

webrtc/voice_engine/include/voe_base.h

@@ -34,10 +34,10 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_H
 #define WEBRTC_VOICE_ENGINE_VOE_BASE_H
 
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 
 namespace webrtc {
 

webrtc/voice_engine/voe_base_impl.cc

@@ -10,10 +10,10 @@
 
 #include "webrtc/voice_engine/voe_base_impl.h"
 
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"