diff options
Diffstat (limited to 'webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h')
| -rw-r--r-- | webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h | 67 |
1 files changed, 39 insertions, 28 deletions
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h index b15ad94..d99e9c8 100644 --- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h @@ -8,18 +8,21 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ -#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ +#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ +#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ +#include <utility> #include <vector> -#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" -#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h" +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/scoped_refptr.h" +#include "api/units/time_delta.h" +#include "rtc_base/constructor_magic.h" +#include "system_wrappers/include/field_trial.h" namespace webrtc { -struct CodecInst; - template <typename T> class AudioEncoderIsacT final : public AudioEncoder { public: @@ -29,9 +32,6 @@ class AudioEncoderIsacT final : public AudioEncoder { // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) struct Config { bool IsOk() const; - - LockedIsacBandwidthInfo* bwinfo = nullptr; - int payload_type = 103; int sample_rate_hz = 16000; int frame_size_ms = 30; @@ -39,46 +39,48 @@ class AudioEncoderIsacT final : public AudioEncoder { // rate, in bits/s. int max_payload_size_bytes = -1; int max_bit_rate = -1; - - // If true, the encoder will dynamically adjust frame size and bit rate; - // the configured values are then merely the starting point. - bool adaptive_mode = false; - - // In adaptive mode, prevent adaptive changes to the frame size. (Not used - // in nonadaptive mode.) - bool enforce_frame_size = false; }; explicit AudioEncoderIsacT(const Config& config); - explicit AudioEncoderIsacT(const CodecInst& codec_inst, - LockedIsacBandwidthInfo* bwinfo); ~AudioEncoderIsacT() override; - size_t MaxEncodedBytes() const override; int SampleRateHz() const override; - int NumChannels() const override; + size_t NumChannels() const override; size_t Num10MsFramesInNextPacket() const override; size_t Max10MsFramesInAPacket() const override; int GetTargetBitrate() const override; - EncodedInfo EncodeInternal(uint32_t rtp_timestamp, - const int16_t* audio, - size_t max_encoded_bytes, - uint8_t* encoded) override; + void SetTargetBitrate(int target_bps) override; + void OnReceivedTargetAudioBitrate(int target_bps) override; + void OnReceivedUplinkBandwidth( + int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms) override; + void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override; + void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) override; void Reset() override; + absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange() + const override; private: // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and // STREAM_MAXW16_60MS for iSAC fix (60 ms). static const size_t kSufficientEncodeBufferSizeBytes = 400; - static const int kDefaultBitRate = 32000; + static constexpr int kDefaultBitRate = 32000; + static constexpr int kMinBitrateBps = 10000; + static constexpr int MaxBitrateBps(int sample_rate_hz) { + return sample_rate_hz == 32000 ? 56000 : 32000; + } + + void SetTargetBitrate(int target_bps, bool subtract_per_packet_overhead); // Recreate the iSAC encoder instance with the given settings, and save them. void RecreateEncoderInstance(const Config& config); Config config_; typename T::instance_type* isac_state_ = nullptr; - LockedIsacBandwidthInfo* bwinfo_ = nullptr; // Have we accepted input but not yet emitted it in a packet? bool packet_in_progress_ = false; @@ -89,9 +91,18 @@ class AudioEncoderIsacT final : public AudioEncoder { // Timestamp of the previously encoded packet. uint32_t last_encoded_timestamp_; + // Cache the value of the "WebRTC-SendSideBwe-WithOverhead" field trial. + const bool send_side_bwe_with_overhead_ = + field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead"); + + // When we send a packet, expect this many bytes of headers to be added to it. + // Start out with a reasonable default that we can use until we receive a real + // value. + DataSize overhead_per_packet_ = DataSize::Bytes(28); + RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ +#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |