Enable 8 bit and float pcm record formats for AudioFlinger

Update sampling rate handling as well.

Bug: 19570772
Change-Id: I872248e64c0578b2e48869a68fee0d51bd0640c3

2 files changed, 55 insertions, 14 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f3206cb..5002099 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -45,6 +45,8 @@
 #include "AudioFlinger.h"
 #include "ServiceUtilities.h"
 
+#include <media/AudioResamplerPublic.h>
+
 #include <media/EffectsFactoryApi.h>
 #include <audio_effects/effect_visualizer.h>
 #include <audio_effects/effect_ns.h>
@@ -1140,19 +1142,46 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form
     if (ret != NO_ERROR) {
         return 0;
     }
+    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
+        return 0;
+    }
 
     AutoMutex lock(mHardwareLock);
     mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
-    audio_config_t config;
-    memset(&config, 0, sizeof(config));
-    config.sample_rate = sampleRate;
-    config.channel_mask = channelMask;
-    config.format = format;
+    audio_config_t config, proposed;
+    memset(&proposed, 0, sizeof(proposed));
+    proposed.sample_rate = sampleRate;
+    proposed.channel_mask = channelMask;
+    proposed.format = format;
 
     audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
-    size_t size = dev->get_input_buffer_size(dev, &config);
+    size_t frames;
+    for (;;) {
+        // Note: config is currently a const parameter for get_input_buffer_size()
+        // but we use a copy from proposed in case config changes from the call.
+        config = proposed;
+        frames = dev->get_input_buffer_size(dev, &config);
+        if (frames != 0) {
+            break; // hal success, config is the result
+        }
+        // change one parameter of the configuration each iteration to a more "common" value
+        // to see if the device will support it.
+        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
+            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
+        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
+            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
+        } else {
+            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
+                    "format %#x, channelMask 0x%X",
+                    sampleRate, format, channelMask);
+            break; // retries failed, break out of loop with frames == 0.
+        }
+    }
     mHardwareStatus = AUDIO_HW_IDLE;
-    return size;
+    if (frames > 0 && config.sample_rate != sampleRate) {
+        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
+    }
+    return frames; // may be converted to bytes at the Java level.
 }
 
 uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
@@ -1419,9 +1448,8 @@ sp<IAudioRecord> AudioFlinger::openRecord(
         goto Exit;
     }
 
-    // we don't yet support anything other than 16-bit PCM
-    if (!(audio_is_valid_format(format) &&
-            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
+    // we don't yet support anything other than linear PCM
+    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
         ALOGE("openRecord() invalid format %#x", format);
         lStatus = BAD_VALUE;
         goto Exit;
@@ -2002,11 +2030,11 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m
             status, address.string());
 
     // If the input could not be opened with the requested parameters and we can handle the
-    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
-    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
+    // conversion internally, try to open again with the proposed parameters.
     if (status == BAD_VALUE &&
-            config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT &&
-        (halconfig.sample_rate <= 2 * config->sample_rate) &&
+        audio_is_linear_pcm(config->format) &&
+        audio_is_linear_pcm(halconfig.format) &&
+        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
         (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
         (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
         // FIXME describe the change proposed by HAL (save old values so we can log them here)
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index c096bdd..1a20fae 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -86,7 +86,13 @@
 #define ALOGVV(a...) do { } while(0)
 #endif
 
+// TODO: Move these macro/inlines to a header file.
 #define max(a, b) ((a) > (b) ? (a) : (b))
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
 
 namespace android {
 
@@ -5622,6 +5628,13 @@ reacquire_wakelock:
                     break;
                 }
 
+                // Don't allow framesOut to be larger than what is possible with resampling
+                // from framesIn.
+                // This isn't strictly necessary but helps limit buffer resizing in
+                // RecordBufferConverter.  TODO: remove when no longer needed.
+                framesOut = min(framesOut,
+                        destinationFramesPossible(
+                                framesIn, mSampleRate, activeTrack->mSampleRate));
                 // process frames from the RecordThread buffer provider to the RecordTrack buffer
                 framesOut = activeTrack->mRecordBufferConverter->convert(
                         activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);