| Index: webrtc/modules/audio_coding/codecs/audio_format.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/audio_format.h b/webrtc/modules/audio_coding/codecs/audio_format.h
|
| deleted file mode 100644
|
| index 6f2c8cfa54e150b32d3f1a597f45f0d7dd98b4ff..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/codecs/audio_format.h
|
| +++ /dev/null
|
| @@ -1,81 +0,0 @@
|
| -/*
|
| - * 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_AUDIO_FORMAT_H_
|
| -#define WEBRTC_MODULES_AUDIO_CODING_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_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
|
|
|