| /* |
| * Copyright (c) 2014 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. |
| */ |
| |
| #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h" |
| |
| #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| // We always encode at 48 kHz. |
| const int kSampleRateHz = 48000; |
| |
| int DivExact(int a, int b) { |
| CHECK_EQ(a % b, 0); |
| return a / b; |
| } |
| |
| int16_t ClampInt16(size_t x) { |
| return static_cast<int16_t>( |
| std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()))); |
| } |
| |
| int16_t CastInt16(size_t x) { |
| DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())); |
| return static_cast<int16_t>(x); |
| } |
| |
| } // namespace |
| |
| AudioEncoderOpus::Config::Config() : frame_size_ms(20), num_channels(1) {} |
| |
| bool AudioEncoderOpus::Config::IsOk() const { |
| if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) |
| return false; |
| if (num_channels <= 0) |
| return false; |
| return true; |
| } |
| |
| AudioEncoderOpus::AudioEncoderOpus(const Config& config) |
| : num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)), |
| num_channels_(config.num_channels), |
| samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) { |
| CHECK(config.IsOk()); |
| input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_); |
| CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_)); |
| } |
| |
| AudioEncoderOpus::~AudioEncoderOpus() { |
| CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_)); |
| } |
| |
| int AudioEncoderOpus::sample_rate_hz() const { |
| return kSampleRateHz; |
| } |
| |
| int AudioEncoderOpus::num_channels() const { |
| return num_channels_; |
| } |
| |
| int AudioEncoderOpus::Num10MsFramesInNextPacket() const { |
| return num_10ms_frames_per_packet_; |
| } |
| |
| bool AudioEncoderOpus::Encode(uint32_t timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| size_t* encoded_bytes, |
| EncodedInfo* info) { |
| if (input_buffer_.empty()) |
| first_timestamp_in_buffer_ = timestamp; |
| input_buffer_.insert(input_buffer_.end(), audio, |
| audio + samples_per_10ms_frame_); |
| if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) * |
| samples_per_10ms_frame_)) { |
| *encoded_bytes = 0; |
| return true; |
| } |
| CHECK_EQ(input_buffer_.size(), |
| static_cast<size_t>(num_10ms_frames_per_packet_) * |
| samples_per_10ms_frame_); |
| int16_t r = WebRtcOpus_Encode( |
| inst_, &input_buffer_[0], |
| DivExact(CastInt16(input_buffer_.size()), num_channels_), |
| ClampInt16(max_encoded_bytes), encoded); |
| input_buffer_.clear(); |
| if (r < 0) |
| return false; |
| *encoded_bytes = r; |
| info->encoded_timestamp = first_timestamp_in_buffer_; |
| return true; |
| } |
| |
| } // namespace webrtc |