blob: c12d734f7c13782dda4b2ef2c516b4e1ecbe0d93 [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/api/audio_codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
namespace webrtc {
struct CodecInst;
template <typename T>
class AudioEncoderIsacT final : public AudioEncoder {
// Allowed combinations of sample rate, frame size, and bit rate are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct Config {
bool IsOk() const;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
int payload_type = 103;
int sample_rate_hz = 16000;
int frame_size_ms = 30;
int bit_rate = kDefaultBitRate; // Limit on the short-term average bit
// 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,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
~AudioEncoderIsacT() override;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
void Reset() override;
// 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;
// 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;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
// Have we accepted input but not yet emitted it in a packet?
bool packet_in_progress_ = false;
// Timestamp of the first input of the currently in-progress packet.
uint32_t packet_timestamp_;
// Timestamp of the previously encoded packet.
uint32_t last_encoded_timestamp_;
} // namespace webrtc