henrik.lundin@webrtc.org | c1c9291 | 2014-12-16 13:41:36 +0000 | [diff] [blame] | 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" |
Peter Boström | d7b7ae8 | 2015-12-08 13:41:35 +0100 | [diff] [blame] | 12 | |
henrik.lundin@webrtc.org | f45c8ca | 2015-02-05 18:29:39 +0000 | [diff] [blame] | 13 | #include "webrtc/base/checks.h" |
Peter Boström | d7b7ae8 | 2015-12-08 13:41:35 +0100 | [diff] [blame] | 14 | #include "webrtc/base/trace_event.h" |
henrik.lundin@webrtc.org | c1c9291 | 2014-12-16 13:41:36 +0000 | [diff] [blame] | 15 | |
| 16 | namespace webrtc { |
| 17 | |
kwiberg | 12cfc9b | 2015-09-08 05:57:53 -0700 | [diff] [blame] | 18 | AudioEncoder::EncodedInfo::EncodedInfo() = default; |
henrik.lundin@webrtc.org | c1c9291 | 2014-12-16 13:41:36 +0000 | [diff] [blame] | 19 | |
kjellander | 470dd37 | 2016-04-19 03:03:23 -0700 | [diff] [blame] | 20 | AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default; |
| 21 | |
kwiberg | 12cfc9b | 2015-09-08 05:57:53 -0700 | [diff] [blame] | 22 | AudioEncoder::EncodedInfo::~EncodedInfo() = default; |
| 23 | |
| 24 | int AudioEncoder::RtpTimestampRateHz() const { |
| 25 | return SampleRateHz(); |
henrik.lundin@webrtc.org | c1c9291 | 2014-12-16 13:41:36 +0000 | [diff] [blame] | 26 | } |
| 27 | |
kwiberg | 288886b | 2015-11-06 01:21:35 -0800 | [diff] [blame] | 28 | AudioEncoder::EncodedInfo AudioEncoder::Encode( |
| 29 | uint32_t rtp_timestamp, |
| 30 | rtc::ArrayView<const int16_t> audio, |
ossu | 10a029e | 2016-03-01 00:41:31 -0800 | [diff] [blame] | 31 | rtc::Buffer* encoded) { |
| 32 | TRACE_EVENT0("webrtc", "AudioEncoder::Encode"); |
| 33 | RTC_CHECK_EQ(audio.size(), |
| 34 | static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); |
| 35 | |
| 36 | const size_t old_size = encoded->size(); |
ossu | 4f43fcf | 2016-03-04 00:54:32 -0800 | [diff] [blame] | 37 | EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); |
ossu | 10a029e | 2016-03-01 00:41:31 -0800 | [diff] [blame] | 38 | RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); |
| 39 | return info; |
| 40 | } |
| 41 | |
kwiberg | 12cfc9b | 2015-09-08 05:57:53 -0700 | [diff] [blame] | 42 | bool AudioEncoder::SetFec(bool enable) { |
| 43 | return !enable; |
henrik.lundin@webrtc.org | 478cedc | 2015-01-27 18:24:45 +0000 | [diff] [blame] | 44 | } |
| 45 | |
kwiberg | 12cfc9b | 2015-09-08 05:57:53 -0700 | [diff] [blame] | 46 | bool AudioEncoder::SetDtx(bool enable) { |
| 47 | return !enable; |
| 48 | } |
| 49 | |
| 50 | bool AudioEncoder::SetApplication(Application application) { |
| 51 | return false; |
| 52 | } |
| 53 | |
kwiberg | 3f5f1c2 | 2015-09-08 23:15:33 -0700 | [diff] [blame] | 54 | void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} |
kwiberg | 12cfc9b | 2015-09-08 05:57:53 -0700 | [diff] [blame] | 55 | |
| 56 | void AudioEncoder::SetProjectedPacketLossRate(double fraction) {} |
| 57 | |
| 58 | void AudioEncoder::SetTargetBitrate(int target_bps) {} |
| 59 | |
ossu | 2903ba5 | 2016-04-18 06:14:33 -0700 | [diff] [blame] | 60 | size_t AudioEncoder::MaxEncodedBytes() const { |
| 61 | RTC_CHECK(false); |
| 62 | return 0; |
| 63 | } |
| 64 | |
henrik.lundin@webrtc.org | c1c9291 | 2014-12-16 13:41:36 +0000 | [diff] [blame] | 65 | } // namespace webrtc |