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Moving src/webrtc into src/.

Browse Source 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
api/audio_codecs/L16/BUILD.gn Normal file
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# 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_static_library("audio_encoder_L16") {
sources = [
"audio_encoder_L16.cc",
"audio_encoder_L16.h",
]
deps = [
"..:audio_codecs_api",
"../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
]
}
rtc_static_library("audio_decoder_L16") {
sources = [
"audio_decoder_L16.cc",
"audio_decoder_L16.h",
]
deps = [
"..:audio_codecs_api",
"../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
]
}

43
api/audio_codecs/L16/audio_decoder_L16.cc Normal file
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/*
* 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/L16/audio_decoder_L16.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
namespace webrtc {
rtc::Optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
const SdpAudioFormat& format) {
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.num_channels = rtc::checked_cast<int>(format.num_channels);
return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
? rtc::Optional<Config>(config)
: rtc::Optional<Config>();
}
void AudioDecoderL16::AppendSupportedDecoders(
std::vector<AudioCodecSpec>* specs) {
Pcm16BAppendSupportedCodecSpecs(specs);
}
std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
const Config& config) {
return config.IsOk() ? rtc::MakeUnique<AudioDecoderPcm16B>(
config.sample_rate_hz, config.num_channels)
: nullptr;
}
} // namespace webrtc

44
api/audio_codecs/L16/audio_decoder_L16.h Normal file
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/*
* 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.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
namespace webrtc {
// L16 decoder API for use as a template parameter to
// CreateAudioDecoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
struct AudioDecoderL16 {
struct Config {
bool IsOk() const {
return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
num_channels >= 1;
}
int sample_rate_hz = 8000;
int num_channels = 1;
};
static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config);
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_

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api/audio_codecs/L16/audio_encoder_L16.cc Normal file
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/*
* 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/L16/audio_encoder_L16.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
namespace webrtc {
rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
const SdpAudioFormat& format) {
if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
return rtc::Optional<Config>();
}
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
? rtc::Optional<Config>(config)
: rtc::Optional<Config>();
}
void AudioEncoderL16::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
Pcm16BAppendSupportedCodecSpecs(specs);
}
AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
const AudioEncoderL16::Config& config) {
RTC_DCHECK(config.IsOk());
return {config.sample_rate_hz,
rtc::dchecked_cast<size_t>(config.num_channels),
config.sample_rate_hz * config.num_channels * 16};
}
std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
const AudioEncoderL16::Config& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
AudioEncoderPcm16B::Config c;
c.sample_rate_hz = config.sample_rate_hz;
c.num_channels = config.num_channels;
c.frame_size_ms = config.frame_size_ms;
c.payload_type = payload_type;
return rtc::MakeUnique<AudioEncoderPcm16B>(c);
}
} // namespace webrtc

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api/audio_codecs/L16/audio_encoder_L16.h Normal file
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/*
* 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.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
namespace webrtc {
// L16 encoder API for use as a template parameter to
// CreateAudioEncoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
struct AudioEncoderL16 {
struct Config {
bool IsOk() const {
return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
num_channels >= 1 && frame_size_ms > 0 && frame_size_ms <= 120 &&
frame_size_ms % 10 == 0;
}
int sample_rate_hz = 8000;
int num_channels = 1;
int frame_size_ms = 10;
};
static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
int payload_type);
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_