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We changed Encode() and EncodeInternal() return type from bool to void in this issue:

Browse Source 
https://webrtc-codereview.appspot.com/38279004/
Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43839004

Cr-Commit-Position: refs/heads/master@{#8749}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jmarusic@webrtc.org
2015-03-17 12:12:17 +00:00
parent 73d763e71f
commit 0cb612b43b
21 changed files with 222 additions and 237 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -183,19 +183,18 @@ void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
}
}
void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
if (input_buffer_.empty())
first_timestamp_in_buffer_ = rtp_timestamp;
input_buffer_.insert(input_buffer_.end(), audio,
audio + samples_per_10ms_frame_);
if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
samples_per_10ms_frame_)) {
info->encoded_bytes = 0;
return;
return kZeroEncodedBytes;
}
CHECK_EQ(input_buffer_.size(),
static_cast<size_t>(num_10ms_frames_per_packet_) *
@ -207,12 +206,13 @@ void AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp,
ClampInt16(max_encoded_bytes), encoded);
CHECK_GE(r, 0); // Fails only if fed invalid data.
input_buffer_.clear();
info->encoded_bytes = r;
info->encoded_timestamp = first_timestamp_in_buffer_;
info->payload_type = payload_type_;
// Allows Opus to send empty packets.
info->send_even_if_empty = true;
info->speech = r > 0;
EncodedInfo info;
info.encoded_bytes = r;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.send_even_if_empty = true; // Allows Opus to send empty packets.
info.speech = r > 0;
return info;
}
} // namespace webrtc

9
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
View File

@ -58,11 +58,10 @@ class AudioEncoderOpus final : public AudioEncoder {
bool dtx_enabled() const { return dtx_enabled_; }
protected:
void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
private:
const int num_10ms_frames_per_packet_;