webrtc/api/audio_codecs/audio_format.h
Danil Chapovalov 0bc58cf876 Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
2018-06-21 12:50:03 +00:00

137 lines
4.8 KiB
C++

/*
* 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 API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <string>
#include <utility>
#include "absl/types/optional.h"
#include "rtc_base/checks.h"
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, size_t num_channels);
SdpAudioFormat(const std::string& name,
int clockrate_hz,
size_t num_channels);
SdpAudioFormat(const char* name,
int clockrate_hz,
size_t num_channels,
const Parameters& param);
SdpAudioFormat(const std::string& name,
int clockrate_hz,
size_t num_channels,
const Parameters& param);
~SdpAudioFormat();
// Returns true if this format is compatible with |o|. In SDP terminology:
// would it represent the same codec between an offer and an answer? As
// opposed to operator==, this method disregards codec parameters.
bool Matches(const SdpAudioFormat& o) const;
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;
size_t num_channels;
Parameters parameters;
};
void swap(SdpAudioFormat& a, SdpAudioFormat& b);
// Information about how an audio format is treated by the codec implementation.
// Contains basic information, such as sample rate and number of channels, which
// isn't uniformly presented by SDP. Also contains flags indicating support for
// integrating with other parts of WebRTC, like external VAD and comfort noise
// level calculation.
//
// To avoid API breakage, and make the code clearer, AudioCodecInfo 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:
// AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true};
// than
// AudioCodecInfo info(16000, 1, 32000);
// info.allow_comfort_noise = true;
// info.future_flag_b = true;
// info.future_flag_c = true;
struct AudioCodecInfo {
AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps);
AudioCodecInfo(int sample_rate_hz,
size_t num_channels,
int default_bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps);
AudioCodecInfo(const AudioCodecInfo& b) = default;
~AudioCodecInfo() = default;
bool operator==(const AudioCodecInfo& b) const {
return sample_rate_hz == b.sample_rate_hz &&
num_channels == b.num_channels &&
default_bitrate_bps == b.default_bitrate_bps &&
min_bitrate_bps == b.min_bitrate_bps &&
max_bitrate_bps == b.max_bitrate_bps &&
allow_comfort_noise == b.allow_comfort_noise &&
supports_network_adaption == b.supports_network_adaption;
}
bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); }
bool HasFixedBitrate() const {
RTC_DCHECK_GE(min_bitrate_bps, 0);
RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
return min_bitrate_bps == max_bitrate_bps;
}
int sample_rate_hz;
size_t num_channels;
int default_bitrate_bps;
int min_bitrate_bps;
int max_bitrate_bps;
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.
};
// AudioCodecSpec ties an audio format to specific information about the codec
// and its implementation.
struct AudioCodecSpec {
bool operator==(const AudioCodecSpec& b) const {
return format == b.format && info == b.info;
}
bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); }
SdpAudioFormat format;
AudioCodecInfo info;
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_FORMAT_H_