Index: webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
index 5fab599fe1ae42e749fb5aa490612e42433bf36d..d95d6c6710ee5eed8e7d80ed608a035cd9f70b99 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
@@ -13,15 +13,14 @@ |
#include <vector> |
+#include "webrtc/base/checks.h" |
#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h" |
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" |
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
namespace webrtc { |
-// NOTE: This class has neither ThreadChecker, nor locks. The owner of an |
-// AudioEncoderOpus object must ensure that it is not accessed concurrently. |
+struct CodecInst; |
class AudioEncoderOpus final : public AudioEncoder { |
public: |
@@ -31,60 +30,44 @@ class AudioEncoderOpus final : public AudioEncoder { |
}; |
struct Config { |
- Config(); |
bool IsOk() const; |
- int frame_size_ms; |
- int num_channels; |
- int payload_type; |
- ApplicationMode application; |
- int bitrate_bps; |
- bool fec_enabled; |
- int max_playback_rate_hz; |
- int complexity; |
- bool dtx_enabled; |
+ int frame_size_ms = 20; |
+ int num_channels = 1; |
+ int payload_type = 120; |
+ ApplicationMode application = kVoip; |
+ int bitrate_bps = 64000; |
+ bool fec_enabled = false; |
+ int max_playback_rate_hz = 48000; |
+ int complexity = kDefaultComplexity; |
+ bool dtx_enabled = false; |
+ |
+ private: |
+#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; |
+#else |
+ static const int kDefaultComplexity = 9; |
+#endif |
}; |
explicit AudioEncoderOpus(const Config& config); |
+ explicit AudioEncoderOpus(const CodecInst& codec_inst); |
~AudioEncoderOpus() override; |
+ size_t MaxEncodedBytes() const override; |
int SampleRateHz() const override; |
int NumChannels() const override; |
- size_t MaxEncodedBytes() const override; |
size_t Num10MsFramesInNextPacket() const override; |
size_t Max10MsFramesInAPacket() const override; |
int GetTargetBitrate() const override; |
- void SetTargetBitrate(int bits_per_second) override; |
- void SetProjectedPacketLossRate(double fraction) override; |
- |
- double packet_loss_rate() const { return packet_loss_rate_; } |
- ApplicationMode application() const { return application_; } |
- bool dtx_enabled() const { return dtx_enabled_; } |
EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
const int16_t* audio, |
size_t max_encoded_bytes, |
uint8_t* encoded) override; |
- private: |
- const size_t num_10ms_frames_per_packet_; |
- const int num_channels_; |
- const int payload_type_; |
- const ApplicationMode application_; |
- int bitrate_bps_; |
- const bool dtx_enabled_; |
- const size_t samples_per_10ms_frame_; |
- std::vector<int16_t> input_buffer_; |
- OpusEncInst* inst_; |
- uint32_t first_timestamp_in_buffer_; |
- double packet_loss_rate_; |
-}; |
- |
-struct CodecInst; |
- |
-class AudioEncoderMutableOpus |
- : public AudioEncoderMutableImpl<AudioEncoderOpus> { |
- public: |
- explicit AudioEncoderMutableOpus(const CodecInst& codec_inst); |
+ void Reset() override; |
bool SetFec(bool enable) override; |
// Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice |
@@ -94,18 +77,24 @@ class AudioEncoderMutableOpus |
bool SetApplication(Application application) override; |
bool SetMaxPlaybackRate(int frequency_hz) override; |
- AudioEncoderOpus::ApplicationMode application() const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->application(); |
- } |
- double packet_loss_rate() const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->packet_loss_rate(); |
- } |
- bool dtx_enabled() const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->dtx_enabled(); |
- } |
+ 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; } |
+ |
+ private: |
+ int Num10msFramesPerPacket() const; |
+ int SamplesPer10msFrame() 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 |