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Move AudioDecoder and related stuff to the api/ directory

Browse source 
BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
This commit is contained in:
kwiberg 2017-02-10 08:15:44 -08:00 committed by Commit bot
parent 84a3759825
commit 087bd34d23
78 changed files with 535 additions and 380 deletions
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2
webrtc/api/BUILD.gn
View file

@ -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",
]
}

10
webrtc/api/DEPS
View file

@ -12,9 +12,11 @@ specific_include_rules = {
"+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",
],
}

39
webrtc/api/audio_codecs/BUILD.gn Normal file
View file

@ -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",
]
}

64
webrtc/modules/audio_coding/codecs/audio_decoder.cc → webrtc/api/audio_codecs/audio_decoder.cc
View file

@ -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 sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* speech_type) {
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);
@ -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 sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* 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);
@ -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 {

177
webrtc/api/audio_codecs/audio_decoder.h Normal file
View file

@ -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_

11
webrtc/modules/audio_coding/codecs/audio_decoder_factory.h → webrtc/api/audio_codecs/audio_decoder_factory.h
View file

@ -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_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
#ifndef WEBRTC_API_AUDIO_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_

5
webrtc/modules/audio_coding/codecs/audio_format.cc → webrtc/api/audio_codecs/audio_format.cc
View file

@ -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)
: format(format) {}
AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {}
AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
: format(std::move(format)) {}

6
webrtc/modules/audio_coding/codecs/audio_format.h → webrtc/api/audio_codecs/audio_format.h
View file

@ -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_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <ostream>
@ -78,4 +78,4 @@ struct AudioCodecSpec {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_

21
webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc Normal file
View file

@ -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

25
webrtc/api/audio_codecs/builtin_audio_decoder_factory.h Normal file
View file

@ -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_

4
webrtc/api/peerconnectioninterface.h
View file

@ -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 {

2
webrtc/call/audio_receive_stream.h
View file

@ -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"

2
webrtc/logging/BUILD.gn
View file

@ -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",
]

2
webrtc/media/base/mediaengine.h
View file

@ -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

2
webrtc/media/engine/payload_type_mapper.cc
View file

@ -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 {

2
webrtc/media/engine/payload_type_mapper.h
View file

@ -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 {

5
webrtc/media/engine/webrtcmediaengine.cc
View file

@ -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 {

2
webrtc/media/engine/webrtcmediaengine_unittest.cc
View file

@ -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;

6
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View file

@ -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;

126
webrtc/modules/audio_coding/BUILD.gn
View file

@ -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",
]
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",
"codecs/builtin_audio_decoder_factory_internal.cc",
"codecs/builtin_audio_decoder_factory_internal.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",
":audio_decoder_factory_interface",
":builtin_audio_decoder_factory",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs: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",
":builtin_audio_decoder_factory",
"../../api/audio_codecs:audio_codecs_api",
"../../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",
]
}

2
webrtc/modules/audio_coding/acm2/acm_receive_test.cc
View file

@ -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"

2
webrtc/modules/audio_coding/acm2/acm_receive_test.h
View file

@ -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 {

2
webrtc/modules/audio_coding/acm2/acm_receiver.cc
View file

@ -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"

2
webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
View file

@ -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"

2
webrtc/modules/audio_coding/acm2/audio_coding_module.cc
View file

@ -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"

2
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
View file

@ -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"

4
webrtc/modules/audio_coding/acm2/rent_a_codec.h
View file

@ -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"

169
webrtc/modules/audio_coding/codecs/audio_decoder.h
View file

@ -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 <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#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_

2
webrtc/modules/audio_coding/codecs/audio_format_conversion.h
View file

@ -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 {

17
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
View file

@ -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/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
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

11
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc → webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
View file

@ -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"
@ -176,14 +177,15 @@ 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;
#ifdef WEBRTC_CODEC_OPUS
// clang-format off
AudioCodecSpec opus({"opus", 48000, 2, {
{"minptime", "10"},
{"useinbandfec", "1"}
}});
// clang-format on
opus.allow_comfort_noise = false;
opus.supports_network_adaption = true;
specs.push_back(opus);
@ -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>);
}

24
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h Normal file
View file

@ -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_

2
webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc → webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
View file

@ -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 {

2
webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
View file

@ -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 {

2
webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
View file

@ -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;

2
webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
View file

@ -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;

2
webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
View file

@ -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 {

2
webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
View file

@ -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 {

4
webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h
View file

@ -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 {

2
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
View file

@ -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 {

2
webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
View file

@ -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 {

2
webrtc/modules/audio_coding/include/audio_coding_module.h
View file

@ -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"

7
webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
View file

@ -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 {

2
webrtc/modules/audio_coding/neteq/comfort_noise.cc
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/decoder_database.cc
View file

@ -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 {

4
webrtc/modules/audio_coding/neteq/decoder_database.h
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
View file

@ -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"

3
webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
View file

@ -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 {

3
webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
View file

@ -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"

6
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View file

@ -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"

7
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View file

@ -10,14 +10,12 @@
#include <memory>
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.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"

2
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/normal.cc
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/packet.h
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/packet_buffer.cc
View file

@ -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"

5
webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
View file

@ -10,11 +10,10 @@
// Unit tests for PacketBuffer class.
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#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_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"

2
webrtc/modules/audio_coding/neteq/post_decode_vad.h
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
View file

@ -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"

5
webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
View file

@ -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/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_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"

2
webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
View file

@ -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 {

2
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
View file

@ -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 {

2
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
View file

@ -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"

3
webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
View file

@ -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"

2
webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
View file

@ -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 {

2
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
View file

@ -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"

4
webrtc/modules/utility/BUILD.gn
View file

@ -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",
]
}

2
webrtc/pc/peerconnectionfactory.cc
View file

@ -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"

2
webrtc/pc/peerconnectioninterface_unittest.cc
View file

@ -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"

2
webrtc/test/call_test.cc
View file

@ -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"

2
webrtc/test/fuzzers/audio_decoder_fuzzer.cc
View file

@ -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 {

2
webrtc/tools/BUILD.gn
View file

@ -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",

6
webrtc/voice_engine/BUILD.gn
View file

@ -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",

2
webrtc/voice_engine/coder.cc
View file

@ -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 {

4
webrtc/voice_engine/include/voe_base.h
View file

@ -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 {

2
webrtc/voice_engine/voe_base_impl.cc
View file

@ -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"