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}
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 8c443cb..2e3d5fe 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -29,8 +29,8 @@
":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",
]
}
diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS
index 4543a44..0b3778b 100644
--- a/webrtc/api/DEPS
+++ b/webrtc/api/DEPS
@@ -12,9 +12,11 @@
"+webrtc/voice_engine",
],
- # TODO(kwiberg): Remove this exception when audio_decoder_factory.h
- # has moved to api/.
- "peerconnectioninterface\.h": [
- "+webrtc/modules/audio_coding/codecs/audio_decoder_factory.h",
+ # We allow .cc files in webrtc/api/ to #include a bunch of stuff
+ # that's off-limits for the .h files. That's because .h files leak
+ # their #includes to whoever's #including them, but .cc files do not
+ # since no one #includes them.
+ ".*\.cc": [
+ "+webrtc/modules/audio_coding",
],
}
diff --git a/webrtc/api/audio_codecs/BUILD.gn b/webrtc/api/audio_codecs/BUILD.gn
new file mode 100644
index 0000000..c003df8
--- /dev/null
+++ b/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",
+ ]
+}
diff --git a/webrtc/api/audio_codecs/audio_decoder.cc b/webrtc/api/audio_codecs/audio_decoder.cc
new file mode 100644
index 0000000..90342a8
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_decoder.cc
@@ -0,0 +1,169 @@
+/*
+ * 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.
+ */
+
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+
+#include <assert.h>
+#include <memory>
+#include <utility>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/sanitizer.h"
+#include "webrtc/base/trace_event.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,
+ int priority,
+ std::unique_ptr<EncodedAudioFrame> frame)
+ : timestamp(timestamp), priority(priority), frame(std::move(frame)) {
+ RTC_DCHECK_GE(priority, 0);
+}
+
+AudioDecoder::ParseResult::~ParseResult() = default;
+
+AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
+ ParseResult&& b) = default;
+
+std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
+ rtc::Buffer&& payload,
+ uint32_t timestamp) {
+ std::vector<ParseResult> results;
+ std::unique_ptr<EncodedAudioFrame> frame(
+ 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 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);
+ if (duration >= 0 &&
+ duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+ return -1;
+ }
+ return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+ speech_type);
+}
+
+int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
+ size_t encoded_len,
+ 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);
+ if (duration >= 0 &&
+ duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+ return -1;
+ }
+ return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
+ speech_type);
+}
+
+int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ 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;
+}
+
+size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
+ return 0;
+}
+
+int AudioDecoder::IncomingPacket(const uint8_t* payload,
+ size_t payload_len,
+ uint16_t rtp_sequence_number,
+ uint32_t rtp_timestamp,
+ uint32_t arrival_timestamp) {
+ return 0;
+}
+
+int AudioDecoder::ErrorCode() {
+ return 0;
+}
+
+int AudioDecoder::PacketDuration(const uint8_t* encoded,
+ size_t encoded_len) const {
+ return kNotImplemented;
+}
+
+int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
+ size_t encoded_len) const {
+ return kNotImplemented;
+}
+
+bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
+ size_t encoded_len) const {
+ return false;
+}
+
+AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
+ switch (type) {
+ case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
+ case 1:
+ return kSpeech;
+ case 2:
+ return kComfortNoise;
+ default:
+ assert(false);
+ return kSpeech;
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/api/audio_codecs/audio_decoder.h b/webrtc/api/audio_codecs/audio_decoder.h
new file mode 100644
index 0000000..dab7d3b
--- /dev/null
+++ b/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_
diff --git a/webrtc/api/audio_codecs/audio_decoder_factory.h b/webrtc/api/audio_codecs/audio_decoder_factory.h
new file mode 100644
index 0000000..3479a4e
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_decoder_factory.h
@@ -0,0 +1,37 @@
+/*
+ * 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_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/base/refcount.h"
+
+namespace webrtc {
+
+// A factory that creates AudioDecoders.
+// NOTE: This class is still under development and may change without notice.
+class AudioDecoderFactory : public rtc::RefCountInterface {
+ public:
+ virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0;
+
+ virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0;
+
+ virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+ const SdpAudioFormat& format) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
diff --git a/webrtc/api/audio_codecs/audio_format.cc b/webrtc/api/audio_codecs/audio_format.cc
new file mode 100644
index 0000000..b0a86e2
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_format.cc
@@ -0,0 +1,85 @@
+/*
+ * 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.
+ */
+
+#include "webrtc/api/audio_codecs/audio_format.h"
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
+SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
+
+SdpAudioFormat::SdpAudioFormat(const char* name,
+ int clockrate_hz,
+ int num_channels)
+ : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(const std::string& name,
+ int clockrate_hz,
+ int num_channels)
+ : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(const char* name,
+ int clockrate_hz,
+ int num_channels,
+ const Parameters& param)
+ : name(name),
+ clockrate_hz(clockrate_hz),
+ num_channels(num_channels),
+ parameters(param) {}
+
+SdpAudioFormat::SdpAudioFormat(const std::string& name,
+ int clockrate_hz,
+ int num_channels,
+ const Parameters& param)
+ : name(name),
+ clockrate_hz(clockrate_hz),
+ num_channels(num_channels),
+ parameters(param) {}
+
+SdpAudioFormat::~SdpAudioFormat() = default;
+SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
+SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+
+bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+ return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
+ a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
+ a.parameters == b.parameters;
+}
+
+void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
+ using std::swap;
+ swap(a.name, b.name);
+ swap(a.clockrate_hz, b.clockrate_hz);
+ swap(a.num_channels, b.num_channels);
+ swap(a.parameters, b.parameters);
+}
+
+std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
+ os << "{name: " << saf.name;
+ os << ", clockrate_hz: " << saf.clockrate_hz;
+ os << ", num_channels: " << saf.num_channels;
+ os << ", parameters: {";
+ const char* sep = "";
+ for (const auto& kv : saf.parameters) {
+ os << sep << kv.first << ": " << kv.second;
+ sep = ", ";
+ }
+ os << "}}";
+ return os;
+}
+
+AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {}
+
+AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
+ : format(std::move(format)) {}
+
+} // namespace webrtc
diff --git a/webrtc/api/audio_codecs/audio_format.h b/webrtc/api/audio_codecs/audio_format.h
new file mode 100644
index 0000000..db3990f
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_format.h
@@ -0,0 +1,81 @@
+/*
+ * 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_AUDIO_FORMAT_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
+
+#include <map>
+#include <ostream>
+#include <string>
+#include <utility>
+
+namespace webrtc {
+
+// SDP specification for a single audio codec.
+// NOTE: This class is still under development and may change without notice.
+struct SdpAudioFormat {
+ using Parameters = std::map<std::string, std::string>;
+
+ SdpAudioFormat(const SdpAudioFormat&);
+ SdpAudioFormat(SdpAudioFormat&&);
+ SdpAudioFormat(const char* name, int clockrate_hz, int num_channels);
+ SdpAudioFormat(const std::string& name, int clockrate_hz, int num_channels);
+ SdpAudioFormat(const char* name,
+ int clockrate_hz,
+ int num_channels,
+ const Parameters& param);
+ SdpAudioFormat(const std::string& name,
+ int clockrate_hz,
+ int num_channels,
+ const Parameters& param);
+ ~SdpAudioFormat();
+
+ SdpAudioFormat& operator=(const SdpAudioFormat&);
+ SdpAudioFormat& operator=(SdpAudioFormat&&);
+
+ friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
+ friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+ return !(a == b);
+ }
+
+ std::string name;
+ int clockrate_hz;
+ int num_channels;
+ Parameters parameters;
+};
+
+void swap(SdpAudioFormat& a, SdpAudioFormat& b);
+std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
+
+// To avoid API breakage, and make the code clearer, AudioCodecSpec should not
+// be directly initializable with any flags indicating optional support. If it
+// were, these initializers would break any time a new flag was added. It's also
+// more difficult to understand:
+// AudioCodecSpec spec{{"format", 8000, 1}, true, false, false, true, true};
+// than
+// AudioCodecSpec spec({"format", 8000, 1});
+// spec.allow_comfort_noise = true;
+// spec.future_flag_b = true;
+// spec.future_flag_c = true;
+struct AudioCodecSpec {
+ explicit AudioCodecSpec(const SdpAudioFormat& format);
+ explicit AudioCodecSpec(SdpAudioFormat&& format);
+ ~AudioCodecSpec() = default;
+
+ SdpAudioFormat format;
+ bool allow_comfort_noise = true; // This codec can be used with an external
+ // comfort noise generator.
+ bool supports_network_adaption = false; // This codec can adapt to varying
+ // network conditions.
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
diff --git a/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc b/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
new file mode 100644
index 0000000..9bd049b
--- /dev/null
+++ b/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
diff --git a/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h b/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h
new file mode 100644
index 0000000..f7452f4
--- /dev/null
+++ b/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_
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index ea94f50..2a22258 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/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 {