1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "api/audio_codecs/audio_encoder.h"
12
13 #include "rtc_base/checks.h"
14 #include "rtc_base/trace_event.h"
15
16 namespace webrtc {
17
18 ANAStats::ANAStats() = default;
19 ANAStats::~ANAStats() = default;
20 ANAStats::ANAStats(const ANAStats&) = default;
21
22 AudioEncoder::EncodedInfo::EncodedInfo() = default;
23 AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
24 AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
25 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
26 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
27 const EncodedInfo&) = default;
28 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
29 default;
30
RtpTimestampRateHz() const31 int AudioEncoder::RtpTimestampRateHz() const {
32 return SampleRateHz();
33 }
34
Encode(uint32_t rtp_timestamp,rtc::ArrayView<const int16_t> audio,rtc::Buffer * encoded)35 AudioEncoder::EncodedInfo AudioEncoder::Encode(
36 uint32_t rtp_timestamp,
37 rtc::ArrayView<const int16_t> audio,
38 rtc::Buffer* encoded) {
39 TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
40 RTC_CHECK_EQ(audio.size(),
41 static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
42
43 const size_t old_size = encoded->size();
44 EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
45 RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
46 return info;
47 }
48
SetFec(bool enable)49 bool AudioEncoder::SetFec(bool enable) {
50 return !enable;
51 }
52
SetDtx(bool enable)53 bool AudioEncoder::SetDtx(bool enable) {
54 return !enable;
55 }
56
GetDtx() const57 bool AudioEncoder::GetDtx() const {
58 return false;
59 }
60
SetApplication(Application application)61 bool AudioEncoder::SetApplication(Application application) {
62 return false;
63 }
64
SetMaxPlaybackRate(int frequency_hz)65 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
66
SetTargetBitrate(int target_bps)67 void AudioEncoder::SetTargetBitrate(int target_bps) {}
68
69 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
ReclaimContainedEncoders()70 AudioEncoder::ReclaimContainedEncoders() {
71 return nullptr;
72 }
73
EnableAudioNetworkAdaptor(const std::string & config_string,RtcEventLog * event_log)74 bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
75 RtcEventLog* event_log) {
76 return false;
77 }
78
DisableAudioNetworkAdaptor()79 void AudioEncoder::DisableAudioNetworkAdaptor() {}
80
OnReceivedUplinkPacketLossFraction(float uplink_packet_loss_fraction)81 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
82 float uplink_packet_loss_fraction) {}
83
OnReceivedUplinkRecoverablePacketLossFraction(float uplink_recoverable_packet_loss_fraction)84 void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
85 float uplink_recoverable_packet_loss_fraction) {}
86
OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps)87 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
88 OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::nullopt);
89 }
90
OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,rtc::Optional<int64_t> bwe_period_ms)91 void AudioEncoder::OnReceivedUplinkBandwidth(
92 int target_audio_bitrate_bps,
93 rtc::Optional<int64_t> bwe_period_ms) {}
94
OnReceivedRtt(int rtt_ms)95 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
96
OnReceivedOverhead(size_t overhead_bytes_per_packet)97 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
98
SetReceiverFrameLengthRange(int min_frame_length_ms,int max_frame_length_ms)99 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
100 int max_frame_length_ms) {}
101
GetANAStats() const102 ANAStats AudioEncoder::GetANAStats() const {
103 return ANAStats();
104 }
105
106 } // namespace webrtc
107