Line data    Source code

       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           0 : int AudioEncoder::RtpTimestampRateHz() const {
      28           0 :   return SampleRateHz();
      29             : }
      30             : 
      31           0 : AudioEncoder::EncodedInfo AudioEncoder::Encode(
      32             :     uint32_t rtp_timestamp,
      33             :     rtc::ArrayView<const int16_t> audio,
      34             :     rtc::Buffer* encoded) {
      35           0 :   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
      36           0 :   RTC_CHECK_EQ(audio.size(),
      37           0 :                static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
      38             : 
      39           0 :   const size_t old_size = encoded->size();
      40           0 :   EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
      41           0 :   RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
      42           0 :   return info;
      43             : }
      44             : 
      45           0 : bool AudioEncoder::SetFec(bool enable) {
      46           0 :   return !enable;
      47             : }
      48             : 
      49           0 : bool AudioEncoder::SetDtx(bool enable) {
      50           0 :   return !enable;
      51             : }
      52             : 
      53           0 : bool AudioEncoder::GetDtx() const {
      54           0 :   return false;
      55             : }
      56             : 
      57           0 : bool AudioEncoder::SetApplication(Application application) {
      58           0 :   return false;
      59             : }
      60             : 
      61           0 : void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
      62             : 
      63           0 : void AudioEncoder::SetTargetBitrate(int target_bps) {}
      64             : 
      65             : rtc::ArrayView<std::unique_ptr<AudioEncoder>>
      66           0 : AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
      67             : 
      68           0 : bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
      69             :                                              RtcEventLog* event_log,
      70             :                                              const Clock* clock) {
      71           0 :   return false;
      72             : }
      73             : 
      74           0 : void AudioEncoder::DisableAudioNetworkAdaptor() {}
      75             : 
      76           0 : void AudioEncoder::OnReceivedUplinkPacketLossFraction(
      77           0 :     float uplink_packet_loss_fraction) {}
      78             : 
      79           0 : void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
      80           0 :   OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
      81           0 : }
      82             : 
      83           0 : void AudioEncoder::OnReceivedUplinkBandwidth(
      84             :     int target_audio_bitrate_bps,
      85           0 :     rtc::Optional<int64_t> probing_interval_ms) {}
      86             : 
      87           0 : void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
      88             : 
      89           0 : void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
      90             : 
      91           0 : void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
      92           0 :                                                int max_frame_length_ms) {}
      93             : 
      94             : }  // namespace webrtc