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Switch to using AudioEncoderOpus instead of ACMOpus

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This change switches from the old codec wrapper ACMOpus to the new
AudioEncoderOpus wrapped in an ACMGenericCodecWrapper.

BUG=4228
TEST=Please, try the Opus codec extensively.
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33259004

Cr-Commit-Position: refs/heads/master@{#8341}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8341 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-02-11 21:33:34 +00:00
parent 1226e926e6
commit 6c68c85b46
3 changed files with 20 additions and 2 deletions
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17
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -20,6 +20,16 @@ namespace {
const int kMinBitrateBps = 500; const int kMinBitrateBps = 500;
const int kMaxBitrateBps = 512000; const int kMaxBitrateBps = 512000;
// TODO(tlegrand): Remove this code when we have proper APIs to set the
// complexity at a higher level.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
// default, to save encoder complexity.
const int kDefaultComplexity = 5;
#else
const int kDefaultComplexity = 9;
#endif
// We always encode at 48 kHz. // We always encode at 48 kHz.
const int kSampleRateHz = 48000; const int kSampleRateHz = 48000;
@ -42,7 +52,8 @@ AudioEncoderOpus::Config::Config()
application(kVoip), application(kVoip),
bitrate_bps(64000), bitrate_bps(64000),
fec_enabled(false), fec_enabled(false),
max_playback_rate_hz(48000) { max_playback_rate_hz(48000),
complexity(kDefaultComplexity) {
} }
bool AudioEncoderOpus::Config::IsOk() const { bool AudioEncoderOpus::Config::IsOk() const {
@ -52,6 +63,8 @@ bool AudioEncoderOpus::Config::IsOk() const {
return false; return false;
if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps) if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
return false; return false;
if (complexity < 0 || complexity > 10)
return false;
return true; return true;
} }
@ -75,6 +88,8 @@ AudioEncoderOpus::AudioEncoderOpus(const Config& config)
} }
CHECK_EQ(0, CHECK_EQ(0,
WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz)); WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
} }
AudioEncoderOpus::~AudioEncoderOpus() { AudioEncoderOpus::~AudioEncoderOpus() {

1
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
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@ -38,6 +38,7 @@ class AudioEncoderOpus final : public AudioEncoder {
int bitrate_bps; int bitrate_bps;
bool fec_enabled; bool fec_enabled;
int max_playback_rate_hz; int max_playback_rate_hz;
int complexity;
}; };
explicit AudioEncoderOpus(const Config& config); explicit AudioEncoderOpus(const Config& config);

4
webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
View File

@ -695,7 +695,9 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst& codec_inst,
#endif #endif
} else if (!STR_CASE_CMP(codec_inst.plname, "opus")) { } else if (!STR_CASE_CMP(codec_inst.plname, "opus")) {
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
return new ACMOpus(kOpus, enable_red); return new ACMGenericCodecWrapper(codec_inst, cng_pt_nb, cng_pt_wb,
cng_pt_swb, cng_pt_fb, enable_red,
red_payload_type);
#endif #endif
} else if (!STR_CASE_CMP(codec_inst.plname, "speex")) { } else if (!STR_CASE_CMP(codec_inst.plname, "speex")) {
#ifdef WEBRTC_CODEC_SPEEX #ifdef WEBRTC_CODEC_SPEEX