blob: 4b3eb845e03790a27d27deb0634c742b5caa8e5f (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
|
/*
* 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 API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// G711 encoder API for use as a template parameter to
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderG711 {
struct Config {
enum class Type { kPcmU, kPcmA };
bool IsOk() const {
return (type == Type::kPcmU || type == Type::kPcmA) &&
frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
num_channels >= 1 &&
num_channels <= AudioEncoder::kMaxNumberOfChannels;
}
Type type = Type::kPcmU;
int num_channels = 1;
int frame_size_ms = 20;
};
static absl::optional<AudioEncoderG711::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,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
|
