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Unified Diff: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

Issue 1184313002: Add AudioEncoder::GetTargetBitrate (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing GN compile Created 5 years, 6 months ago
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Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index befb3558beb85ff8eb304fae5317bbe6f19cde52..d2b20e3b941c932cf736489b17cf4455154a9d32 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -70,16 +70,18 @@ AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
decoder_sample_rate_hz_(0),
lock_(CriticalSectionWrapper::CreateCriticalSection()),
- packet_in_progress_(false) {
+ packet_in_progress_(false),
+ target_bitrate_bps_(config.adaptive_mode ? -1 : (config.bit_rate == 0
+ ? kDefaultBitRate
+ : config.bit_rate)) {
CHECK(config.IsOk());
CHECK_EQ(0, T::Create(&isac_state_));
CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
if (config.adaptive_mode) {
- CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate,
- config.frame_size_ms, config.enforce_frame_size));
-
+ CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
+ config.enforce_frame_size));
} else {
CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
}
@@ -130,6 +132,11 @@ int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
}
template <typename T>
+int AudioEncoderDecoderIsacT<T>::GetTargetBitrate() const {
+ return target_bitrate_bps_;
+}
+
+template <typename T>
AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
uint32_t rtp_timestamp,
const int16_t* audio,

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