Index: webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc |
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc |
index 201a2d4bb43354401f56d7501833ff0364bbfb8c..195265dba60dbd8c2fb5ce1733986b706361580d 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc |
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc |
@@ -15,13 +15,15 @@ |
namespace webrtc { |
-// Explicit instantiation of AudioEncoderDecoderIsacT<IsacFloat>, a.k.a. |
-// AudioEncoderDecoderIsac. |
-template class AudioEncoderDecoderIsacT<IsacFloat>; |
+// Explicit instantiation: |
+template class AudioEncoderIsacT<IsacFloat>; |
+template class AudioDecoderIsacT<IsacFloat>; |
namespace { |
-AudioEncoderDecoderIsac::Config CreateConfig(const CodecInst& codec_inst) { |
- AudioEncoderDecoderIsac::Config config; |
+AudioEncoderIsac::Config CreateConfig(const CodecInst& codec_inst, |
+ LockedIsacBandwidthInfo* bwinfo) { |
+ AudioEncoderIsac::Config config; |
+ config.bwinfo = bwinfo; |
config.payload_type = codec_inst.pltype; |
config.sample_rate_hz = codec_inst.plfreq; |
config.frame_size_ms = |
@@ -33,111 +35,24 @@ AudioEncoderDecoderIsac::Config CreateConfig(const CodecInst& codec_inst) { |
} |
} // namespace |
-AudioEncoderDecoderMutableIsacFloat::AudioEncoderDecoderMutableIsacFloat( |
- const CodecInst& codec_inst) |
- : AudioEncoderMutableImpl<AudioEncoderDecoderIsac, |
- AudioEncoderDecoderMutableIsac>( |
- CreateConfig(codec_inst)) { |
+AudioEncoderMutableIsacFloat::AudioEncoderMutableIsacFloat( |
+ const CodecInst& codec_inst, |
+ LockedIsacBandwidthInfo* bwinfo) |
+ : AudioEncoderMutableImpl<AudioEncoderIsac>( |
+ CreateConfig(codec_inst, bwinfo)) { |
} |
-void AudioEncoderDecoderMutableIsacFloat::UpdateSettings( |
- const CodecInst& codec_inst) { |
- bool success = Reconstruct(CreateConfig(codec_inst)); |
- DCHECK(success); |
-} |
- |
-void AudioEncoderDecoderMutableIsacFloat::SetMaxPayloadSize( |
+void AudioEncoderMutableIsacFloat::SetMaxPayloadSize( |
int max_payload_size_bytes) { |
auto conf = config(); |
conf.max_payload_size_bytes = max_payload_size_bytes; |
Reconstruct(conf); |
} |
-void AudioEncoderDecoderMutableIsacFloat::SetMaxRate(int max_rate_bps) { |
+void AudioEncoderMutableIsacFloat::SetMaxRate(int max_rate_bps) { |
auto conf = config(); |
conf.max_bit_rate = max_rate_bps; |
Reconstruct(conf); |
} |
-int AudioEncoderDecoderMutableIsacFloat::Decode(const uint8_t* encoded, |
- size_t encoded_len, |
- int sample_rate_hz, |
- size_t max_decoded_bytes, |
- int16_t* decoded, |
- SpeechType* speech_type) { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->Decode(encoded, encoded_len, sample_rate_hz, |
- max_decoded_bytes, decoded, speech_type); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::DecodeRedundant( |
- const uint8_t* encoded, |
- size_t encoded_len, |
- int sample_rate_hz, |
- size_t max_decoded_bytes, |
- int16_t* decoded, |
- SpeechType* speech_type) { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->DecodeRedundant(encoded, encoded_len, sample_rate_hz, |
- max_decoded_bytes, decoded, speech_type); |
-} |
- |
-bool AudioEncoderDecoderMutableIsacFloat::HasDecodePlc() const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->HasDecodePlc(); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::DecodePlc(int num_frames, |
- int16_t* decoded) { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->DecodePlc(num_frames, decoded); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::Init() { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->Init(); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::IncomingPacket( |
- const uint8_t* payload, |
- size_t payload_len, |
- uint16_t rtp_sequence_number, |
- uint32_t rtp_timestamp, |
- uint32_t arrival_timestamp) { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->IncomingPacket(payload, payload_len, rtp_sequence_number, |
- rtp_timestamp, arrival_timestamp); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::ErrorCode() { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->ErrorCode(); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::PacketDuration( |
- const uint8_t* encoded, |
- size_t encoded_len) const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->PacketDuration(encoded, encoded_len); |
-} |
- |
-int AudioEncoderDecoderMutableIsacFloat::PacketDurationRedundant( |
- const uint8_t* encoded, |
- size_t encoded_len) const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->PacketDurationRedundant(encoded, encoded_len); |
-} |
- |
-bool AudioEncoderDecoderMutableIsacFloat::PacketHasFec( |
- const uint8_t* encoded, |
- size_t encoded_len) const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->PacketHasFec(encoded, encoded_len); |
-} |
- |
-size_t AudioEncoderDecoderMutableIsacFloat::Channels() const { |
- CriticalSectionScoped cs(encoder_lock_.get()); |
- return encoder()->Channels(); |
-} |
- |
} // namespace webrtc |