blob: 367c9b90eae5421424683a8f6ece8f919ed02de8 [file] [log] [blame]
* 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 <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
namespace webrtc {
struct CodecInst;
class AudioEncoderOpus final : public AudioEncoder {
enum ApplicationMode {
kVoip = 0,
kAudio = 1,
struct Config {
Config(const Config&);
Config& operator=(const Config&);
bool IsOk() const;
int GetBitrateBps() const;
int frame_size_ms = 20;
size_t num_channels = 1;
int payload_type = 120;
ApplicationMode application = kVoip;
rtc::Optional<int> bitrate_bps; // Unset means to use default value.
bool fec_enabled = false;
int max_playback_rate_hz = 48000;
int complexity = kDefaultComplexity;
bool dtx_enabled = false;
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
// default, to save encoder complexity.
static const int kDefaultComplexity = 5;
static const int kDefaultComplexity = 9;
explicit AudioEncoderOpus(const Config& config);
explicit AudioEncoderOpus(const CodecInst& codec_inst);
~AudioEncoderOpus() override;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void Reset() override;
bool SetFec(bool enable) override;
// Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
// being inactive. During that, it still sends 2 packets (one for content, one
// for signaling) about every 400 ms.
bool SetDtx(bool enable) override;
bool SetApplication(Application application) override;
void SetMaxPlaybackRate(int frequency_hz) override;
void SetProjectedPacketLossRate(double fraction) override;
void SetTargetBitrate(int target_bps) override;
// Getters for testing.
double packet_loss_rate() const { return packet_loss_rate_; }
ApplicationMode application() const { return config_.application; }
bool dtx_enabled() const { return config_.dtx_enabled; }
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
size_t Num10msFramesPerPacket() const;
size_t SamplesPer10msFrame() const;
size_t SufficientOutputBufferSize() const;
bool RecreateEncoderInstance(const Config& config);
Config config_;
double packet_loss_rate_;
std::vector<int16_t> input_buffer_;
OpusEncInst* inst_;
uint32_t first_timestamp_in_buffer_;
} // namespace webrtc