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Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ )

Browse Source 
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.

Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}

TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1883543002

Cr-Commit-Position: refs/heads/master@{#12330}
This commit is contained in:
ossu
2016-04-12 03:58:07 -07:00
committed by Commit bot
parent 5222d315db
commit 164bc4bbd3
23 changed files with 278 additions and 36 deletions
Download Patch File Download Diff File Expand all files Collapse all files

22
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -100,6 +100,16 @@ AudioEncoderOpus::~AudioEncoderOpus() {
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
}
size_t AudioEncoderOpus::MaxEncodedBytes() const {
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
const size_t bytes_per_millisecond =
static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
const size_t approx_encoded_bytes =
Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
}
int AudioEncoderOpus::SampleRateHz() const {
return kSampleRateHz;
}
@ -188,7 +198,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
RTC_CHECK_EQ(input_buffer_.size(),
Num10msFramesPerPacket() * SamplesPer10msFrame());
const size_t max_encoded_bytes = ApproximateEncodedBytes();
const size_t max_encoded_bytes = MaxEncodedBytes();
EncodedInfo info;
info.encoded_bytes =
encoded->AppendData(
@ -221,16 +231,6 @@ size_t AudioEncoderOpus::SamplesPer10msFrame() const {
return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
}
size_t AudioEncoderOpus::ApproximateEncodedBytes() const {
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
const size_t bytes_per_millisecond =
static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
const size_t approx_encoded_bytes =
Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
}
// If the given config is OK, recreate the Opus encoder instance with those
// settings, save the config, and return true. Otherwise, do nothing and return
// false.

4
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -54,6 +54,7 @@ class AudioEncoderOpus final : public AudioEncoder {
explicit AudioEncoderOpus(const CodecInst& codec_inst);
~AudioEncoderOpus() override;
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
@ -78,7 +79,7 @@ class AudioEncoderOpus final : public AudioEncoder {
ApplicationMode application() const { return config_.application; }
bool dtx_enabled() const { return config_.dtx_enabled; }
protected:
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
@ -86,7 +87,6 @@ class AudioEncoderOpus final : public AudioEncoder {
private:
size_t Num10msFramesPerPacket() const;
size_t SamplesPer10msFrame() const;
size_t ApproximateEncodedBytes() const;
bool RecreateEncoderInstance(const Config& config);
Config config_;