Revert "AudioEncoderDecoderIsac: Merge the two config structs"
Reason for revert - breaks Hangouts
This reverts commit 7c324cac50ac38122b3f3b26455bc55ad834bfc0.
BUG=chromium:478161
Review URL: https://webrtc-codereview.appspot.com/43209004
Cr-Commit-Position: refs/heads/master@{#9030}
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index ad53166..cda5548 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -25,7 +25,8 @@
template <typename T>
class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
public:
- // Allowed combinations of sample rate, frame size, and bit rate are
+ // For constructing an encoder in instantaneous mode. Allowed combinations
+ // are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
@@ -33,24 +34,34 @@
struct Config {
Config();
bool IsOk() const;
-
int payload_type;
int sample_rate_hz;
int frame_size_ms;
- int bit_rate; // Limit on the short-term average bit rate, in bits/s.
- int max_payload_size_bytes;
+ int bit_rate; // Limit on the short-term average bit rate, in bits/second.
int max_bit_rate;
+ int max_payload_size_bytes;
+ };
- // If true, the encoder will dynamically adjust frame size and bit rate;
- // the configured values are then merely the starting point.
- bool adaptive_mode;
-
- // In adaptive mode, prevent adaptive changes to the frame size. (Not used
- // in nonadaptive mode.)
- bool enforce_frame_size;
+ // For constructing an encoder in channel-adaptive mode. Allowed combinations
+ // are
+ // - 16000 Hz, 30 ms, 10000-32000 bps
+ // - 16000 Hz, 60 ms, 10000-32000 bps
+ // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
+ // - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
+ struct ConfigAdaptive {
+ ConfigAdaptive();
+ bool IsOk() const;
+ int payload_type;
+ int sample_rate_hz;
+ int initial_frame_size_ms;
+ int initial_bit_rate;
+ int max_bit_rate;
+ bool enforce_frame_size; // Prevent adaptive changes to the frame size?
+ int max_payload_size_bytes;
};
explicit AudioEncoderDecoderIsacT(const Config& config);
+ explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
~AudioEncoderDecoderIsacT() override;
// AudioEncoder public methods.
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 186d318..e81686a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -22,17 +22,16 @@
namespace webrtc {
const int kIsacPayloadType = 103;
+const int kDefaultBitRate = 32000;
template <typename T>
AudioEncoderDecoderIsacT<T>::Config::Config()
: payload_type(kIsacPayloadType),
sample_rate_hz(16000),
frame_size_ms(30),
- bit_rate(32000),
- max_payload_size_bytes(-1),
+ bit_rate(kDefaultBitRate),
max_bit_rate(-1),
- adaptive_mode(false),
- enforce_frame_size(false) {
+ max_payload_size_bytes(-1) {
}
template <typename T>
@@ -48,7 +47,7 @@
if (max_payload_size_bytes > 400)
return false;
return (frame_size_ms == 30 || frame_size_ms == 60) &&
- (bit_rate >= 10000 && bit_rate <= 32000);
+ ((bit_rate >= 10000 && bit_rate <= 32000) || bit_rate == 0);
case 32000:
case 48000:
if (max_bit_rate > 160000)
@@ -56,7 +55,47 @@
if (max_payload_size_bytes > 600)
return false;
return T::has_swb &&
- (frame_size_ms == 30 && bit_rate >= 10000 && bit_rate <= 56000);
+ (frame_size_ms == 30 &&
+ ((bit_rate >= 10000 && bit_rate <= 56000) || bit_rate == 0));
+ default:
+ return false;
+ }
+}
+
+template <typename T>
+AudioEncoderDecoderIsacT<T>::ConfigAdaptive::ConfigAdaptive()
+ : payload_type(kIsacPayloadType),
+ sample_rate_hz(16000),
+ initial_frame_size_ms(30),
+ initial_bit_rate(kDefaultBitRate),
+ max_bit_rate(-1),
+ enforce_frame_size(false),
+ max_payload_size_bytes(-1) {
+}
+
+template <typename T>
+bool AudioEncoderDecoderIsacT<T>::ConfigAdaptive::IsOk() const {
+ if (max_bit_rate < 32000 && max_bit_rate != -1)
+ return false;
+ if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
+ return false;
+ switch (sample_rate_hz) {
+ case 16000:
+ if (max_bit_rate > 53400)
+ return false;
+ if (max_payload_size_bytes > 400)
+ return false;
+ return (initial_frame_size_ms == 30 || initial_frame_size_ms == 60) &&
+ initial_bit_rate >= 10000 && initial_bit_rate <= 32000;
+ case 32000:
+ case 48000:
+ if (max_bit_rate > 160000)
+ return false;
+ if (max_payload_size_bytes > 600)
+ return false;
+ return T::has_swb &&
+ (initial_frame_size_ms == 30 && initial_bit_rate >= 10000 &&
+ initial_bit_rate <= 56000);
default:
return false;
}
@@ -71,15 +110,11 @@
packet_in_progress_(false) {
CHECK(config.IsOk());
CHECK_EQ(0, T::Create(&isac_state_));
- CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
+ CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
- if (config.adaptive_mode) {
- CHECK_EQ(0, T::ControlBwe(isac_state_, config.bit_rate,
- config.frame_size_ms, config.enforce_frame_size));
-
- } else {
- CHECK_EQ(0, T::Control(isac_state_, config.bit_rate, config.frame_size_ms));
- }
+ CHECK_EQ(0, T::Control(isac_state_, config.bit_rate == 0 ? kDefaultBitRate
+ : config.bit_rate,
+ config.frame_size_ms));
// When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is
// still set to 32000 Hz, since there is no full-band mode in the decoder.
CHECK_EQ(0, T::SetDecSampRate(isac_state_,
@@ -89,7 +124,29 @@
T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
if (config.max_bit_rate != -1)
CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
- CHECK_EQ(0, T::DecoderInit(isac_state_));
+}
+
+template <typename T>
+AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(
+ const ConfigAdaptive& config)
+ : payload_type_(config.payload_type),
+ state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ decoder_sample_rate_hz_(0),
+ lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ packet_in_progress_(false) {
+ CHECK(config.IsOk());
+ CHECK_EQ(0, T::Create(&isac_state_));
+ CHECK_EQ(0, T::EncoderInit(isac_state_, 0));
+ CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
+ CHECK_EQ(0, T::ControlBwe(isac_state_, config.initial_bit_rate,
+ config.initial_frame_size_ms,
+ config.enforce_frame_size));
+ CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
+ if (config.max_payload_size_bytes != -1)
+ CHECK_EQ(0,
+ T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
+ if (config.max_bit_rate != -1)
+ CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
}
template <typename T>
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
index c852d8b..6b04081 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -293,34 +293,51 @@
#endif
#ifdef WEBRTC_CODEC_ISACFX
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
+ DCHECK_EQ(codec_inst.plfreq, 16000);
is_isac_ = true;
- AudioEncoderDecoderIsacFix::Config config;
- config.payload_type = codec_inst.pltype;
- config.sample_rate_hz = codec_inst.plfreq;
- config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 16);
- if (codec_inst.rate != -1)
+ AudioEncoderDecoderIsacFix* enc_dec;
+ if (codec_inst.rate == -1) {
+ // Adaptive mode.
+ AudioEncoderDecoderIsacFix::ConfigAdaptive config;
+ config.payload_type = codec_inst.pltype;
+ enc_dec = new AudioEncoderDecoderIsacFix(config);
+ } else {
+ // Channel independent mode.
+ AudioEncoderDecoderIsacFix::Config config;
config.bit_rate = codec_inst.rate;
- config.max_payload_size_bytes = max_payload_size_bytes_;
- config.max_bit_rate = max_rate_bps_;
- config.adaptive_mode = (codec_inst.rate == -1);
- auto* enc_dec = new AudioEncoderDecoderIsacFix(config);
+ config.frame_size_ms = codec_inst.pacsize / 16;
+ config.payload_type = codec_inst.pltype;
+ enc_dec = new AudioEncoderDecoderIsacFix(config);
+ }
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif
#ifdef WEBRTC_CODEC_ISAC
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
is_isac_ = true;
- AudioEncoderDecoderIsac::Config config;
- config.payload_type = codec_inst.pltype;
- config.sample_rate_hz = codec_inst.plfreq;
- config.frame_size_ms =
- rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
- if (codec_inst.rate != -1)
+ AudioEncoderDecoderIsac* enc_dec;
+ if (codec_inst.rate == -1) {
+ // Adaptive mode.
+ AudioEncoderDecoderIsac::ConfigAdaptive config;
+ config.sample_rate_hz = codec_inst.plfreq;
+ config.initial_frame_size_ms = rtc::CheckedDivExact(
+ 1000 * codec_inst.pacsize, config.sample_rate_hz);
+ config.max_payload_size_bytes = max_payload_size_bytes_;
+ config.max_bit_rate = max_rate_bps_;
+ config.payload_type = codec_inst.pltype;
+ enc_dec = new AudioEncoderDecoderIsac(config);
+ } else {
+ // Channel independent mode.
+ AudioEncoderDecoderIsac::Config config;
+ config.sample_rate_hz = codec_inst.plfreq;
config.bit_rate = codec_inst.rate;
- config.max_payload_size_bytes = max_payload_size_bytes_;
- config.max_bit_rate = max_rate_bps_;
- config.adaptive_mode = (codec_inst.rate == -1);
- auto* enc_dec = new AudioEncoderDecoderIsac(config);
+ config.frame_size_ms = rtc::CheckedDivExact(1000 * codec_inst.pacsize,
+ config.sample_rate_hz);
+ config.max_payload_size_bytes = max_payload_size_bytes_;
+ config.max_bit_rate = max_rate_bps_;
+ config.payload_type = codec_inst.pltype;
+ enc_dec = new AudioEncoderDecoderIsac(config);
+ }
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 334a2fc..728caef 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -361,7 +361,6 @@
AudioEncoderDecoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
- config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
@@ -381,7 +380,6 @@
AudioEncoderDecoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
- config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
@@ -401,7 +399,6 @@
AudioEncoderDecoderIsacFix::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
- config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
