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| 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 "webrtc/modules/audio_coding/codecs/audio_encoder.h" | |
| 12 | |
| 13 #include "webrtc/base/checks.h" | |
| 14 #include "webrtc/base/trace_event.h" | |
| 15 | |
| 16 namespace webrtc { | |
| 17 | |
| 18 AudioEncoder::EncodedInfo::EncodedInfo() = default; | |
| 19 AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default; | |
| 20 AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default; | |
| 21 AudioEncoder::EncodedInfo::~EncodedInfo() = default; | |
| 22 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=( | |
| 23 const EncodedInfo&) = default; | |
| 24 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) = | |
| 25 default; | |
| 26 | |
| 27 int AudioEncoder::RtpTimestampRateHz() const { | |
| 28 return SampleRateHz(); | |
| 29 } | |
| 30 | |
| 31 AudioEncoder::EncodedInfo AudioEncoder::Encode( | |
| 32 uint32_t rtp_timestamp, | |
| 33 rtc::ArrayView<const int16_t> audio, | |
| 34 rtc::Buffer* encoded) { | |
| 35 TRACE_EVENT0("webrtc", "AudioEncoder::Encode"); | |
| 36 RTC_CHECK_EQ(audio.size(), | |
| 37 static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); | |
| 38 | |
| 39 const size_t old_size = encoded->size(); | |
| 40 EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); | |
| 41 RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); | |
| 42 return info; | |
| 43 } | |
| 44 | |
| 45 bool AudioEncoder::SetFec(bool enable) { | |
| 46 return !enable; | |
| 47 } | |
| 48 | |
| 49 bool AudioEncoder::SetDtx(bool enable) { | |
| 50 return !enable; | |
| 51 } | |
| 52 | |
| 53 bool AudioEncoder::GetDtx() const { | |
| 54 return false; | |
| 55 } | |
| 56 | |
| 57 bool AudioEncoder::SetApplication(Application application) { | |
| 58 return false; | |
| 59 } | |
| 60 | |
| 61 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} | |
| 62 | |
| 63 void AudioEncoder::SetTargetBitrate(int target_bps) {} | |
| 64 | |
| 65 rtc::ArrayView<std::unique_ptr<AudioEncoder>> | |
| 66 AudioEncoder::ReclaimContainedEncoders() { return nullptr; } | |
| 67 | |
| 68 bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, | |
| 69 RtcEventLog* event_log, | |
| 70 const Clock* clock) { | |
| 71 return false; | |
| 72 } | |
| 73 | |
| 74 void AudioEncoder::DisableAudioNetworkAdaptor() {} | |
| 75 | |
| 76 void AudioEncoder::OnReceivedUplinkPacketLossFraction( | |
| 77 float uplink_packet_loss_fraction) {} | |
| 78 | |
| 79 void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( | |
| 80 float uplink_recoverable_packet_loss_fraction) {} | |
| 81 | |
| 82 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { | |
| 83 OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>()); | |
| 84 } | |
| 85 | |
| 86 void AudioEncoder::OnReceivedUplinkBandwidth( | |
| 87 int target_audio_bitrate_bps, | |
| 88 rtc::Optional<int64_t> probing_interval_ms) {} | |
| 89 | |
| 90 void AudioEncoder::OnReceivedRtt(int rtt_ms) {} | |
| 91 | |
| 92 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} | |
| 93 | |
| 94 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, | |
| 95 int max_frame_length_ms) {} | |
| 96 | |
| 97 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2799033006:
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. (Closed)
