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Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()

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Old review at:
https://webrtc-codereview.appspot.com/43839004/

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45769004

Cr-Commit-Position: refs/heads/master@{#8788}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jmarusic@webrtc.org
2015-03-19 08:50:26 +00:00
parent d21406d333
commit 9afaee74ab
22 changed files with 219 additions and 236 deletions
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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 EncodedInfo();
}
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_;