AudioEncoderOpusImpl: Remove unused static methods

Bug: webrtc:10631
Change-Id: I17583ff04f461a281c4ab0ad9322506431c9cade
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138074
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28070}
This commit is contained in:
Karl Wiberg
2019-05-23 12:57:01 +02:00
committed by Commit Bot
parent 87da109df5
commit e0eb325d0d
2 changed files with 0 additions and 24 deletions

View File

@ -304,25 +304,6 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
return absl::make_unique<AudioEncoderOpusImpl>(config, payload_type);
}
absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (absl::EqualsIgnoreCase(format.name, GetPayloadName()) &&
format.clockrate_hz == kRtpTimestampRateHz && format.num_channels == 2) {
const size_t num_channels = GetChannelCount(format);
const int bitrate =
CalculateBitrate(GetMaxPlaybackRate(format), num_channels,
GetFormatParameter(format, "maxaveragebitrate"));
AudioCodecInfo info(kRtpTimestampRateHz, num_channels, bitrate,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
return info;
}
return absl::nullopt;
}
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||

View File

@ -81,11 +81,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
~AudioEncoderOpusImpl() override;
// Static interface for use by BuiltinAudioEncoderFactory.
static constexpr const char* GetPayloadName() { return "opus"; }
static absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format);
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;