AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate

A later change will allow them to differ.

Bug: webrtc:10631
Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27991}
This commit is contained in:
Karl Wiberg 2019-05-16 15:14:01 +02:00 committed by Commit Bot
parent 94079f8452
commit eb16697259
2 changed files with 15 additions and 6 deletions

View File

@ -51,6 +51,7 @@ constexpr int kOpusBitrateWbBps = 20000;
constexpr int kOpusBitrateFbBps = 32000;
constexpr int kSampleRateHz = 48000;
constexpr int kRtpTimestampRateHz = 48000;
constexpr int kDefaultMaxPlaybackRate = 48000;
// These two lists must be sorted from low to high
@ -276,8 +277,10 @@ float AudioEncoderOpusImpl::NewPacketLossRateOptimizer::OptimizePacketLossRate(
void AudioEncoderOpusImpl::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {
"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}};
const SdpAudioFormat fmt = {"opus",
kRtpTimestampRateHz,
2,
{{"minptime", "10"}, {"useinbandfec", "1"}}};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
@ -285,7 +288,8 @@ void AudioEncoderOpusImpl::AppendSupportedEncoders(
AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
AudioCodecInfo info(48000, config.num_channels, *config.bitrate_bps,
AudioCodecInfo info(kRtpTimestampRateHz, config.num_channels,
*config.bitrate_bps,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
@ -303,12 +307,12 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (absl::EqualsIgnoreCase(format.name, GetPayloadName()) &&
format.clockrate_hz == 48000 && format.num_channels == 2) {
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(48000, num_channels, bitrate,
AudioCodecInfo info(kRtpTimestampRateHz, num_channels, bitrate,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
@ -322,7 +326,7 @@ absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
format.clockrate_hz != kRtpTimestampRateHz || format.num_channels != 2) {
return absl::nullopt;
}
@ -482,6 +486,10 @@ size_t AudioEncoderOpusImpl::NumChannels() const {
return config_.num_channels;
}
int AudioEncoderOpusImpl::RtpTimestampRateHz() const {
return kRtpTimestampRateHz;
}
size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
return Num10msFramesPerPacket();
}

View File

@ -88,6 +88,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;